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June 2014

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ffmpeg-devel.log.20140624
by burek 25 Jun '14

25 Jun '14

[01:08] <cone-622> ffmpeg.git 03Michael Niedermayer 07master:9326caa5faa1: avutil/lzo: allow selecting the compression function in the test code [01:08] <cone-622> ffmpeg.git 03Tobias Rapp 07master:f6e772f9b5cf: ffprobe: add color_range and color_space to stream output [01:19] <cone-622> ffmpeg.git 03Vittorio Giovara 07master:ab72eda15e98: pixfmt: mark the reserved values [01:19] <cone-622> ffmpeg.git 03Michael Niedermayer 07master:faa1d4b1340f: Merge commit 'ab72eda15e98197cf148abc08574206cfde0d9b0' [01:23] <BBB> jamrial: so & is it faster? (ssse3 emu edge) [01:42] <jamrial> didn't bench it, but it should considering it's one mov + pshufb vs two movs, an imul and like two punpk* in most cases [01:46] <cone-622> ffmpeg.git 03Vittorio Giovara 07master:10306e9c5fcc: jpeg2000: fix dereferencing invalid pointers during cleanup [01:46] <cone-622> ffmpeg.git 03Michael Niedermayer 07master:97511bc7cf44: Merge commit '10306e9c5fcc28bd9310a9b38f21540e9e1433e9' [02:05] <cone-622> ffmpeg.git 03Vittorio Giovara 07master:772d150a6e82: h264: error out from decode_nal_units() when AV_EF_EXPLODE is set [02:05] <cone-622> ffmpeg.git 03Michael Niedermayer 07master:82e4340f1ef0: Merge commit '772d150a6e82542c06b0c251e73dd299d98d1027' [02:05] <cone-622> ffmpeg.git 03Michael Niedermayer 07master:47048aa30b5c: avcodec/h264: do not leave ret random on minor failures, causing major failure [02:06] <jamrial> BBB: http://pastebin.com/AerP440D this win64 [02:18] <ac_slater> hey guys. #ffmpeg is a little dead and I don't think I've ever received any responses to any libav* questions. I'm just looking for some general workflow documentation. What are contexts and when to use them? I guess just basic things. `doc/examples` doesnt have any networking/streaming code so I'm not sure and of the url_* functions work and how URLContext is maintained. Any help is much appreciated [02:18] <ac_slater> and sorry for going offtopic [02:19] <ac_slater> s/not sure/not sure how/g [02:21] <cone-622> ffmpeg.git 03Michael Niedermayer 07master:eab2509f8ccf: avcodec/x86/h264_qpel_10bit: locally define pb_0 [02:23] <BBB> ac_slater: thats a very generic question, Im sure youre having a more specific problem? [02:24] <BBB> if not, just start doing stuff? and report if you have problems? [02:24] <BBB> network is typically done thoruhg opening a file with AVFormatContext btw [02:25] <ac_slater> BBB: I guess my question was meant to be vague - but you're right, without a specific misunderstanding, no one can help. I guess the size of libav* is intimidating. I'll post to the mailing list of #ffmpeg with specifics and leave you guys alone ;). [02:44] <Compnn> ac_slater : theres some docs on http://wiki.multimedia.cx [02:45] <Compnn> not sure how up to date they are [02:45] <Compnn> but yeah libav-user list is best if developing 3rd party thing [02:47] <ac_slater> Compnn: thanks mate. [03:17] <cone-622> ffmpeg.git 03Michael Niedermayer 07master:236aa4de2a10: avfilter/vf_hqx: avoid floats [03:17] <cone-622> ffmpeg.git 03Michael Niedermayer 07master:723550d3cadb: avfilter/vf_hqx: optimize table init [03:19] <michaelni> ubitux, just pushed some hqx float fix attempt, hope you didnt mind, but i hate leaving things broken [04:52] <cone-622> ffmpeg.git 03Georg Lippitsch 07master:e3b03da77299: EBU Tech 3285 - Supplement 3 - Peak Envelope Chunk encoder [04:52] <cone-622> ffmpeg.git 03Georg Lippitsch 07master:fd504f7c6fb8: Peak Envelope Chunk encoder: Indent [05:13] Action: Compnn trolls carl some [06:21] <ac_slater> alright I've been debating asking this. In the current decoding_encoding example, why isnt there a proper PTS set around line 444? Sending the output of `./decoding_encoding h264` to the `demuxing_decoding` shows NOPTS. Just curious mostly. [08:20] <ubitux> michaelni: no that's fine, i should have pushed the fix yesterday, thank you [08:29] <mraulet> kierank: commits ff7926d5092f9d4158108963e977e8c992322ba4 and ff7926d5092f9d4158108963e977e8c992322ba4 from https://github.com/OpenHEVC/FFmpeg/commits/ffmpeg_patch help you? [08:49] <kurosu> ah, seems to come and go [08:49] <kurosu> I guess he just reacts to the irc logs [08:49] <kurosu> mraulet: hi [08:50] <kurosu> mraulet: in case you are reading the irc logs as I suspect, are you still allocating full frame structures for the data? [08:51] <kurosu> except if you also use wpp, you really need only 2 lines of CTBs, and the compressed MV field for the next image [08:51] <kurosu> it should reduce memory use [08:52] <kurosu> but the biggest drawback will be the amount of painfully debugged changes [08:53] <plepere> good morning. :) [10:07] <BtbN> QSV on linux requires that you buy a 500$ Intel SDK... [10:07] <BtbN> And the encode api in vaapi is... not great. [10:45] <kierank> mraulet: yes :) [11:41] <mraulet> michaelni: ff7926d5092f9d4158108963e977e8c992322ba4 and 39c4d45c7788081c45c7fae51b7c5d0bcbaece9d from openhevc/ffmpeg will greatly reduce pps computation (useful for streaming application) [11:42] <kierank> Useful for watching world cup in 4k :) [11:43] <mraulet> you can use our ffmpeg flavor then :) [11:43] <mraulet> ^^ [11:44] <kierank> Yeah [11:48] <mraulet> bitrates for the world cup? [12:11] <kierank> mraulet: 36mbps [12:11] <kierank> www.bbc.co.uk/rd/blog/2014/06/bbc-r-d-ultra-high-definition-trials [12:19] <mraulet> 10bit? [12:25] <kierank> 8-bit [12:25] <kierank> mraulet: obe.tv/about-us/obe-blog/item/12-first-look-at-bbc [12:28] <mraulet> kierank: thanks [12:35] <kierank> av500: [12:35] <kierank> person: Vladimir Pantelic [12:35] <kierank> address: Archos Deutschland GmbH [12:35] <kierank> who's this guy [13:20] <kriegerod> is there a function in libavutil to read all the content of file descriptor to a memory buffer? if not, i'm going to make one [13:22] <kierank> fread? [13:22] <kriegerod> that's to read once, i have a need to read out to EOF [13:23] <thardin> fread the size of the buffer, check how much you actually got? [13:23] <kriegerod> libavdevice/lavfi.c dumps '-graph_file' argument with help of av_file_map(), which fails on /dev/stdin, which i want to use for that [13:24] <kriegerod> thardin: no guarantee to read up to EOF with one call [13:24] <J_Darnley> I asked about this recently... [13:24] <thardin> you want to read a filter frmo a pipe? [13:25] <kriegerod> from stdin [13:25] <BBB> glib has functions for that [13:25] <BBB> why do you specifically need that function in libavutil? [13:25] <BBB> (or java) [13:25] <thardin> and you can use -graph_file <(your_command_here) ? [13:25] <thardin> *can't [13:25] <J_Darnley> There's an av_* function, something like mmap [13:25] <kriegerod> thardin: yes [13:25] <thardin> schtrange [13:26] <kriegerod> J_Darnley: av_file_map(). fails on /dev/stdin by design, and it cannot fallback to usual reading [13:26] <kriegerod> so now i'm making a fallback mechanism in lavd/lavfi.c [13:26] <J_Darnley> oh, well [13:26] <kriegerod> willing either to not reinvent it if it is already there, or to move this code to lavu [13:27] <thardin> or wait, fread is designed to be buffered and all that. why would it not read until EOF if the element size is 1? [13:28] <thardin> (ignoring possible slowness, not sure if it's smart enough to not read byte-by-byte) [13:28] <J_Darnley> Other than that I think I was pointed to lut3d and/or curves that just uses fread on an entire file. [13:28] <kriegerod> thardin: even if this approach worked, what buffer size should we preallocate? 1 MB? delicate question [13:28] <thardin> well, then you just have the same problem again don't you? [13:29] <cone-96> ffmpeg.git 03gcocherel 07master:f7f1f4c7ce9c: avcodec/hevc_ps: remove min_cb_addr_zs [13:29] <cone-96> ffmpeg.git 03gcocherel 07master:ba70563d5549: hevc/pps: optimized size of min_tb_addr_zs reduce computation too (cherry picked from commit 39c4d45c7788081c45c7fae51b7c5d0bcbaece9d) [13:29] <michaelni> mraulet, kierank cherry picked the 2 comits [13:29] <kierank> thanks [13:30] <kriegerod> thardin: i have checked vf_curves and vf_lut3d, they don't do exactly what you've said [13:31] <thardin> ok [13:32] <thardin> well, someone's probably already written a "read fd into dynalloc'd buffer" function, as BBB hinted [13:34] <J_Darnley> I've got one but it won't work if you can't seek [13:34] <J_Darnley> Actually I think I stole it from x264 [13:34] <kriegerod> thardin: no, in lut3d i see consequential fread()s and parsing [13:34] <kriegerod> in vf_curves i haven't seen anything related to open/read [13:35] <kriegerod> J_Darnley: yes i have seen something checking buffer size using fseek+ftell [13:35] <kriegerod> so likely there's no such thing at the moment, proceeding with my code, will submit soon [13:36] <J_Darnley> vf_curves use av_file_map [13:36] <J_Darnley> (line 329 if you're interested) [13:37] <cone-96> ffmpeg.git 03Michael Niedermayer 07master:2e1e10a6bb7d: avformat/wavenc: use the bitexact flag from avformat instead of the one from avcodec [13:37] <J_Darnley> I look forward to seeing what you propose [13:47] <cone-96> ffmpeg.git 03Michael Niedermayer 07master:d34d3304a8b7: avformat/wavenc: do not hardcode array size in memset and other functions [13:59] <cone-96> ffmpeg.git 03Michael Niedermayer 07master:3bd0221bc1a0: avformat/wavenc: more specific error return for "Writing 16 bit peak for 8 bit audio" [13:59] <cone-96> ffmpeg.git 03Michael Niedermayer 07master:501158c682ce: avformat/wavenc: simplify malloc failure checking [15:31] <kriegerod> J_Darnley: i have posted my patches in maillist, see author Andrey Utkin [15:42] <J_Darnley> Your documentation about the function is good. [16:26] <J_Darnley> I wonder if that bug report is really because he hasn't used enough? [16:28] <J_Darnley> or maybe not. [16:49] <kierank> plepere: http://obe.tv/about-us/obe-blog/item/13-a-look-at-the-hevc-encoder-bbc-uhd-… [16:49] <kierank> it's not one of fionag's legendary tearaparts [16:49] <kierank> but it should deal with a lot of things [16:50] <plepere> I'm battling my avx2 function, but I'l read that ASAP. [18:41] <cone-96> ffmpeg.git 03Michael Niedermayer 07master:74bd039f8cc2: avformat/nutdec: improve probe speed by 30% [19:42] <michaelni> mateo`, thardin, any other mxf people around, is " Gaullier Nicola (3.8K) [FFmpeg-devel] [PATCH] set/force channelcount in MXF D-10" ok ? [19:44] <mateo`> michaelni: i'm having a quick look at it, i've faced this same bullshit by the past :) [19:58] <michaelni> mateo`, thx [20:00] <mateo`> haha funny, it comes from smartjog :) [20:04] <kierank> mateo`: is smartjog now arkena? [20:04] <kierank> "Cognacq-Jay Image, PSN, Qbrick and SmartJog announce today that they regroup all their activities under a single company and a new brand, Arkena. Arkena" [20:04] <kierank> apparently so [20:09] <mateo`> kierank: yes [20:25] <mateo`> michaelni: review done [20:36] <keynote2k> Hello, all. I'm Software Freedom Conservancy's General Counsel. In the process of investigating a GPL violation report on behalf of one of our member projects, we discovered a possible GPL violation relating to ffmpeg. I'd like to submit a violation report; where should I send it? [20:37] <keynote2k> e.g., does FFmpeg have a dedicated email address for receiving and processing GPL violation reports? [20:39] <j-b> lol [20:40] <j-b> "possible GPL violation relating to ffmpeg" [20:40] <j-b> You mean, the 10000s violation, right? [20:41] <keynote2k> sigh, that wouldn't surprise me. :) [20:42] <J_Darnley> I don't know about a dedicated email address but there is a category for it on trac. [20:42] <michaelni> mateo`, thx [20:43] <iive> we used to have a lawyer that was actively negotiation settlements. [20:43] <iive> unfortunately after the fork the whole thing went down the drain. [20:44] <keynote2k> J_Darnley: thx. Is there a particular tag that you recommend? [20:44] <keynote2k> iive: sorry to hear that. [20:44] <kierank> keynote2k: gpl violation from whom [20:45] <iive> also, the primary license of ffmpeg is lgpl. [20:45] <nevcairiel> well it really depends how its build, its a GPL violation if they use a GPL build [20:45] <J_Darnley> keynote2k: not off hand but I can go look [20:46] <keynote2k> kierank: I'd rather not say on IRC. Conservancy's policy is to file a report privately, and let the project determine whether to make the report public or not. [20:46] <iive> yes, it have to be checked what options are used for the build. and afaik linking to gpl library makes the whole thing gpl (e.g. x264) [20:46] <keynote2k> But, if there's no other way to file a report, we are willing to submit a report to a publicly-archived mailing list, if necessary [20:47] <iive> i think there was ffmpeg-legal maillist, not sure if it is still active. [20:47] <kierank> it's gone afaik [20:48] <iive> maybe michaelni could be of more help... [20:48] <keynote2k> iive, nevcairiel: our analysis is currently focused on the use of our member projects' code, so I don't know if our compliance engineer's analysis went to the level of determining whether FFmpeg's optional GPL'ed components were included in the build. [20:49] <J_Darnley> keynote2k: I meant using the "Type" category with "license violation" on trac but I should say that is public though. [20:49] <iive> i assume that ffmpeg libraries are used, not the whole ffmpeg. [20:49] <keynote2k> (which is another reason why I'd prefer to submit the report privately to the right channels, as opposed to making a public accusation of a license violation) [20:49] <iive> ffmpeg application. [20:50] <michaelni> ffmpeg-legal ML was deleted because it was unused for years [20:51] <michaelni> see " [FFmpeg-devel] [RFC] deleting ffmpeg-legal mailing list" [20:53] <keynote2k> well, if there isn't a dedicated email address, then it sounds as if ffmpeg-trac is the only viable option. I'll check back in with our compliance team to determine how to best proceed with submitting the report. [20:53] <keynote2k> Thank you for your responses. :) [21:41] <cone-96> ffmpeg.git 03Derek Buitenhuis 07master:c11043aca736: mjpegdec: Properly set the context colorspace info [21:52] <cone-96> ffmpeg.git 03Michael Niedermayer 07master:0ed110133f77: avfilter/vf_colormatrix: fix macro () [21:52] <cone-96> ffmpeg.git 03Michael Niedermayer 07master:f364101144f1: avfilter/vf_deshake: fix macro () [22:25] <cone-96> ffmpeg.git 03Derek Buitenhuis 07master:2deb614272e6: mjpegdec: Properly set the context colorspace info [22:25] <cone-96> ffmpeg.git 03Michael Niedermayer 07master:50daffead2de: Merge commit '2deb614272e6faad8802c5341971d08c7272f74d' [22:41] <cone-96> ffmpeg.git 03Marton Balint 07master:9dd97aa3d337: ffplay: eliminate pictq_prev_picture [22:41] <cone-96> ffmpeg.git 03Marton Balint 07master:ba800defa7c8: ffplay: pass simple integers to calculate_display_rect and set_default_window_size [22:41] <cone-96> ffmpeg.git 03Marton Balint 07master:1ca5c1784b1d: ffplay: calculate SDL audio buffer size based on sample rate [22:41] <cone-96> ffmpeg.git 03Marton Balint 07master:371f02388cb1: ffplay: decrease max audio callbacks per second [22:41] <cone-96> ffmpeg.git 03Marton Balint 07master:e281671e21d3: ffplay: decrease audio_diff_threshold [22:41] <cone-96> ffmpeg.git 03Michael Niedermayer 07master:0f59bba9cb95: Merge remote-tracking branch 'cus/stable' [23:30] <Timothy_Gu> plepere: why are there four extra blank lines at the end of x86/hevcdsp.h? [23:31] <Timothy_Gu> plus some more in your AVX2 patch [00:00] --- Wed Jun 25 2014

1 0

ffmpeg.log.20140624
by burek 25 Jun '14

25 Jun '14

[00:01] <ogrgkyle> okay thanks! [00:01] <ogrgkyle> I'll play around with that [00:01] <ogrgkyle> maybe i'll be back at some point with more questions [00:04] <ogrgkyle> So the Dev version doesn't contain ffmpeg.exe... [00:04] <ogrgkyle> How do I use this? [00:05] <c_14> Oh, the dev version is for linking programs you make against. If you want a .exe just go with the static or shared. static is probably the easiest if you just want to try facedetect on a video. [00:05] <ogrgkyle> okay... So static will probalby contain all the "opencv stuff"? [00:06] <c_14> I'm guessing he compiled it in, but try it and you'll find out. [00:06] <ogrgkyle> thanks [00:24] <elichai2> hey [00:24] <elichai2> i'm trying to use avconv [00:24] <elichai2> and i get this error: [00:24] <elichai2> encoder 'aac' is experimental and might produce bad results. [00:24] <elichai2> Add '-strict experimental' if you want to use it. [00:24] <elichai2> any idea how to fix? [00:24] <elichai2> (already tried adding that -strict) [00:24] <c_14> "Add '-strict experimental' if you want to use it." [00:25] <c_14> Also, avconv is part of libav, not ffmpeg. Either use ffmpeg or see #libav for support. [00:25] <elichai2> c_14: tried that [00:25] <c_14> ffmpeg -i file -c:v aac -strict experimental outfile [00:26] <elichai2> ffmpeg isn't deperated? [00:27] <c_14> libav is a fork of ffmpeg, that provides a binary called ffmpeg out of historical reasons and added that message for whatever reason even though it is way too misleading. [00:27] <c_14> If you just want to download a quick working ffmpeg. [00:27] <elichai2> ok.... [00:27] <elichai2> so what command i need to run with ffmpeg to convert 720p to 480p? [00:27] <elichai2> (it's mp4) [00:28] <c_14> ffmpeg -i file.mp4 -vf 'scale=-1:480' -c:v [videocodechere] -c:a copy out.mp4 [00:31] <c_14> See the filtering guide and the h264 guide if you want something decent to start out with: https://trac.ffmpeg.org/wiki/FilteringGuide https://trac.ffmpeg.org/wiki/Encode/H.264 [01:52] <ac_slater> hey all. I'm loosing my mind with ffmpeg's (libavformat) RTP stack. The idea of all of these contexts are fine, but I see little usage documentation. How does an RTPContext relate to a URLContext, etc? [02:16] <ac_slater> I'm actually a little shocked at how little examples there are ... ~4k lines of examples and ~1mill lines of code [02:18] <c_14> Write some, submit a patch. Additions are always welcome. [02:19] <ac_slater> c_14: I totally will. For the next ~8months I'll be working in almost nothing but ffmpeg. [02:19] <ac_slater> (libav*) [02:20] <c_14> The best documentation for the libav* libraries that I can give you is the doxygen. [02:20] <ac_slater> c_14: yea, I mean the doxygen is nice. But picking the lucky context/function out of the large pool is not very guided. [02:21] <ac_slater> Looking at the mux/demux/enc/dec code is useful too... but without setup or context [02:24] <c_14> The libav* libraries have a rather steep learning curve. [02:55] <ac_slater> c_14: it seems that way. I'm interesting in writing some *muxers that work with RTP - thus implementing our own payload types. [04:01] <ac_slater> so, regarding the decoding_encoding example ... the pts (h264) is set to the frame number. When playing the file in `mplayer`, I get complaints about PTS being wrong. What's the deal? [04:14] <needmorespeed1> I'm also facing a problem with pts, using av_read_frame and avcodec_decode_video2, the AVPacket pts equals dts. Changing the rate that these are called, pts and dts are adjusted. I am expecting that pts is a constant regardless of decoding rate. [04:18] <needmorespeed1> There is some info on pts/dts here: [04:18] <needmorespeed1> http://dranger.com/ffmpeg/tutorial05.html [04:18] <needmorespeed1> ac_slater: You find anything? [04:19] <ac_slater> needmorespeed1: I found some various things .. I also read that tutorial ... didnt help :( [04:19] <ac_slater> needmorespeed1: http://arashafiei.wordpress.com/2012/10/19/pts-for-h264-encoding/ ... this looks interesting ... except I have no AVStream [04:19] <needmorespeed1> However, ffmpeg reorders the packets so that the DTS of the packet being processed byavcodec_decode_video() will always be the same as the PTS of the frame it returns. But, another warning: we won't always get this information, either. [04:20] <ac_slater> needmorespeed1: oh nevermind, I do have an AVSTream [04:20] <ac_slater> interesting [04:20] <needmorespeed1> Not to worry, because there's another way to find out the PTS of a frame, and we can have our program reorder the packets by itself. We save the PTS of the first packet of a frame: this will be the PTS of the finished frame. So when the stream doesn't give us a DTS, we just use this saved PTS. We can figure out which packet is the first packet of a frame by letting avcodec_decode_video() tell us. How? Whenever a packet starts a frame, theavcode [04:21] <needmorespeed1> My file is not h264 though: Video: msmpeg4v2 (MP42 / 0x3234504D) [04:21] <needmorespeed1> I think ffmpeg handles it generically at this level? [04:25] <ac_slater> needmorespeed1: I'm a newbie with video stuff... I'm almost clueless regarding the importance of PTS/DTS and their calculations [04:26] <needmorespeed1> ac_slater: In arasha link there is no AVStream, did you get a correct PTS? [04:26] <ac_slater> needmorespeed1: no sadly [04:27] <needmorespeed1> Need PTS to render at the correct rate. I can fake it now by decoding at roughly the frame rate. But I will need to do better. [04:29] <needmorespeed1> It seems like a common problem on google. Noone in this channel stepped up yet. They are letting us tough it out. [04:29] <ac_slater> needmorespeed1: ;) [04:29] <ac_slater> it seems encoder/decoder and muxer specific... right? [04:31] <needmorespeed1> Maybe it can be retrieved from the context or codec using the fps there? [04:31] <needmorespeed1> Stream #0:0: Video: msmpeg4v2 (MP42 / 0x3234504D), yuv420p, 854x480, 1840 kb/s, 24 fps, 24 tbr, 24 tbn, 24 tbc [04:31] <needmorespeed1> Create my own PTS using fps? [04:33] <ac_slater> I tried ... pts = (1.0 / fps * 90 * current_frame); ... apparently the incorrect approach. It was monotonic though. [04:34] <ac_slater> I'm a newbie with this stuff so I honestly dont know why 90khz was chosen [04:34] <ac_slater> (apparently a requirement of h264 to some extent) [04:35] <ac_slater> I might just recompile ffmpeg.c with debug flags ... find a commandline run that works and set soem break points [04:36] <needmorespeed1> Just use the fps, not sure about the 90khz either. So, (1/25 fps) / (1/1000) = 40 timebase units. Increment by 40 each frame. [04:36] <needmorespeed1> http://stackoverflow.com/questions/13595288/understanding-pts-and-dts-in-vi… [04:37] <ac_slater> interesting [04:37] <needmorespeed1> Would be nicer if it was provided by ffmpeg as pts, that would be official [04:40] <ac_slater> needmorespeed1: I guess I'm confused. I set pts... ffmpeg is fine with it. Other players like mplayer claim "No PTS!" [04:41] <needmorespeed1> VLC? [04:41] <ac_slater> ffplay doesnt care either [04:41] <ac_slater> only mplayer says this ... but to some extent... there must be some incorrect settings [04:41] <needmorespeed1> try VLC [04:42] <needmorespeed1> I think VLC is more maintained than mplayer [04:42] <ac_slater> but my mplayer (actually mpv) use libav* for its backend [04:42] <ac_slater> yea I use the newer `mpv` [04:42] <ac_slater> I don't like vlc very much ... [04:43] <ac_slater> do you know of a way `ffplay` can dump its pts? [04:43] <ac_slater> (for the playing stream obviously) [04:43] <needmorespeed1> libav and ffmpeg diverged, probably some incompatibilities [04:43] <needmorespeed1> libav split from ffmpeg [04:44] <needmorespeed1> try VLC for a second opinion [04:44] <needmorespeed1> or another player [04:44] <ac_slater> I meant the ffmpeg libav* libraries ... wildcard meaning all [04:44] <Dark-knight> mkv to mp4 [04:45] <Dark-knight> i need the coder [04:45] <Dark-knight> code* [04:45] <Dark-knight> please and thank you [04:46] <ac_slater> Dark-knight: source code? or ffmpeg utility invocation? [04:46] <Dark-knight> command line to convert mkv to mp4 [04:46] <ac_slater> Dark-knight: literally the first result on google [04:47] <Dark-knight> nvm [06:28] <ac_slater> I asked this in the devel channel since I cant really fin the answer in the docs / examples ... but if I run `./decoding_encoding h264` then `./demuxing_decoding test.h264 test2` ... the second output reports NOPTS. What's the proper way to set PTS here? [06:33] <ParkerR> ac_slater, where are you getting these scripts? [06:34] <ac_slater> ParkerR: they are in `doc/examples` since 2.0 [06:34] <ParkerR> Ahh [09:10] <polysics> hello! I am now concatenating some silent video files, then adding audio with a separate command [09:11] <polysics> oops, nvm [09:11] <polysics> I was using m2v as output :) [09:12] <volmatrix> hello everyone, is it possible to encrypt a multicast udp live stream (with ffmpeg of course) ? [09:25] <satanist> volmatrix: https://ffmpeg.org/ffserver-all.html#crypto [09:31] <polysics> my H264 encoding looks really bad (and file size is not that great) [09:31] <polysics> currently just using -vcodec mpeg4 -acodec aac -strict -2 output/concat.mp4 [09:32] <polysics> I guess I need way more options :) [09:44] <satanist> you can use -b:v or -qscale:v to config the output quality [09:45] <satanist> and mpeg4 != h264 [09:45] <volmatrix> satanist: crypto works only with ffserver? I heard it's not maintained anymore [09:50] <satanist> i don't know but how about you test it [09:50] <satanist> if it don't work you can use srtp or a wrapper witch openssl s_server [09:52] <maksimkaa> hello, whenever I generate a new video with some drawn text on it, I keep seeing a cursor blinking on various parts of the screen. Can anybody advice about why is that and how to hide that cursor? [10:00] <volmatrix> satanist: I tried srtp with no success [10:30] <kriegerod> if i have all my inputs specified in -filter_complex_script file, how can i execute ffmpeg so that it doesn't complain about missing input file directives? [10:44] <kriegerod> and when i add "-f lavfi -i testsrc" as placeholder, i get "not connected to any destination" error. Is there currently a way to overcome this? [11:03] <SleepyPikachu> Hello everyone, I'm looking for someone who has experience using libavfilter. I want to set the sample_fmt of a buffer source to AV_SAMPLE_FMT_NONE but can't work out how to do so. [11:12] <saste> SleepyPikachu, what's the point of setting it to NONE? [11:15] <SleepyPikachu> saste: It's compulsory to set, must match the decoder and the decoder has NONE [11:15] <SleepyPikachu> saste: At least in my understanding. ;-) [11:23] <saste> SleepyPikachu, from my understanding the decoder sample_fmt should be set [11:24] <saste> you just need to make sure that decoder and buffer source sample format *match* [11:24] <saste> to set the sample format on the filter, you can initialize the arguments in the init function [11:24] <saste> or you can set the option on the filter context by using the av_opt API [11:25] <saste> did you have a look at the doc/examples examples? [11:27] <SleepyPikachu> saste: Thanks for taking the time to investigate, the decoder -> encoder has been working without setting the sample format. Now I'm trying to decoder -> filter -> encoder and as you say the decoder and filter must have matching formats. Could the decoder not need a sample format because the av concerned is the audio? [11:45] <saste> SleepyPikachu, when you configure the decoder (avcodec_open) the decoder should have the sample format set [11:45] <saste> i am assuming you have an *audio* decoder [11:46] <saste> the sample format set on the buffersrc sets the sample format *expected* by the source buffer [11:46] <saste> so yes you need to set that [11:47] <saste> i don't understand what you mean by "av concerned" [11:49] <SleepyPikachu> AV stands for AudioVisual, I was proposing that given the sample formats are things like floating planar and the decoder -> filter -> encoder is for audio it might not need a format. [11:50] <saste> SleepyPikachu, this is the workflow, you init the decoder and the encoder [11:50] <saste> then you set up the filterchain [11:50] <saste> you need to make sure that decoder <-> abuffersrc and abuffersink <-> encoder sample formats match [11:51] <SleepyPikachu> agreed [12:36] <maksimkaa> hello, whenever I generate a new video with some drawn text on it, I keep seeing a cursor blinking on various parts of the screen. Can anybody advice about why is that and how to hide that cursor? [12:36] <kriegerod> maksimkaa: no idea. Describe your ffmpeg cmdline and how you playback it [12:40] <maksimkaa> kriegerod: it is like this: ffmpeg -y -i input.mp4 -vf "drawtext=fontfile=/usr/share/fonts/truetype/ttf-dejavu/DejaVuSerif.ttf: text=\\\\'Hello world!\\':draw='eq(n,12)': x=200: y=200: fontsize=14: fontcolor=white: box=1: boxcolor=black -vcodec libx264 -preset fast -crf 18 output.mp4 [12:43] <kriegerod> maksimkaa: what's your ffmpeg version? [12:45] <kriegerod> maksimkaa: i suppose what you see may be just presentation artifact during playback [12:46] <maksimkaa> i am using latest ffmpeg [12:47] <maksimkaa> kriegerod: what i see is like a blinking cursor that is most of the time at the beginning of the text but sometimes it shows on other areas... [14:44] <maksimkaa> kriegerod: any thoughts or suggestions? [14:45] <slystone> Hey guys! I need your expertise. So I have to convert some video files (*.dv to *.webm). So I'm using ffmpeg. But I also have to add some images (beginning and end of the videos). [14:45] <slystone> Is there a way to merge the images without reencoding the whole video (and thus not losing quality)? [14:46] <slystone> Otherwise, would you recommend the version of ffmpeg in Debian testing (2.2.3)? [14:47] <sacarasc> If you do it at the same time as the dv ¶ webm conversion, you'd only have to encode once. [14:47] <sacarasc> To do that, you'd have to use the concat filter, I believe. [14:50] <slystone> Cheers sacarasc :) [16:09] <Dave77> where can I get a binary of ffmpeg? [16:11] <sacarasc> Which OS? [16:11] <Dave77> linux with ARM cpu [16:12] <Dave77> never had much luck with compiling myself.. never normally works [16:13] <sspiff> Dave77, what goes wrong when compiling? [16:14] <Dave77> just linux source compiling in general.. either compile script don't work.. some lib version , some link problem etc etc.. [16:15] <sspiff> I see [16:20] <Fjorgynn> troll [16:21] <sspiff> Dave77, well, I don't know of any static builds, is your ARM board running a distribution? [16:22] <Dave77> yes ubuntu.. will try compile src.. see how far i get [16:22] <sspiff> Dave77, why not use the ubuntu provided package? [16:24] <klaxa> sspiff: that would be avconv then presumably [16:24] <sspiff> klaxa, really? [16:26] <sspiff> doesn't ubuntu bundle ffmpeg proper? [16:26] <c_14> Nah, ubuntu bundles libav. At least last time I checked. [16:30] <sspiff> bah [17:51] <bd12> Hello, it is possible to ffmpeg to redirect output to a socket but to create a socket in that time! For example -f mpegts tcp://127.0.0.1:8000 , but i wanted to make ffmpeg to create that socket too if not exists [17:58] <cuba> bd12: listen=1 ? [18:00] <bd12> what is the complete command after -f [18:04] <cuba> ttp://www.ffmpeg.org/ffmpeg-protocols.html [18:04] <cuba> http://www.ffmpeg.org/ffmpeg-protocols.html [18:09] <bd12> I'm using this command [18:09] <bd12> ffmpeg -i "STRAEM_URL" -acodec copy -vcodec copy -f mpegts tcp://127.0.0.1:8000?listen=1 [18:10] <bd12> but when i connect to 8000 then ffmpeg stops [18:13] <bd12> It accepts one connection, it starts the stream but then when i close the connection [18:13] <bd12> ffmpeg stops [18:20] <cuba> you can use socat bd12 [18:20] <cuba> for every connection you start a new ffmpeg instance [18:21] <bd12> can't be done to one ffmpeg? I mean we can use netcat or something to handle th is [18:21] <bd12> ? [18:23] <ogrgkyle> Hi [18:23] <ogrgkyle> I downloaded Zeranoe FFmpeg dev build [18:23] <ogrgkyle> not build [18:24] <ogrgkyle> But I downloaded Zeranoe FFmpeg dev version [18:24] <ogrgkyle> I'm trying to get one of the example C scripts to compile [18:24] <ogrgkyle> But I get a bunch of "undefined reference to" errors in codeblocks [18:25] <cuba> you need to link against the lib ogrgkyle [18:25] <ogrgkyle> I've tried adding the lib folder to Search Directories > Linker [18:25] <cuba> add the .lib things in codeblocks [18:26] <cuba> you need to add like "avcodec.lib" etc [18:26] <ogrgkyle> And adding the .lib files to Linker Settings > Link Libraries [18:26] <cuba> i dont know codeblocks but you need to add every .lib on it own i guess [18:26] <ogrgkyle> Yeah, that's what I don't understand. I already added those [18:26] <cuba> the names [18:28] <ogrgkyle> Which IDE do you use? [18:30] <cuba> vs ogrgkyle [19:50] <fixedtype> hi. is it possible to use complex filtergraph with ffplay? i'd like to play two videos side by side to compare quality but for this i need overlay filter. [20:02] <Filarius> hello, I want stream frames generated from my software, I thought I can just save frames to jpeg file and use ffmpeg to stream. But ffmpeg crash if I trying to replace file. Is here any fix ? [20:18] <keynote2k> Hi, all. I'm with Software Freedom Conservancy. We were investigating a GPL violation report on behalf of one of our member projects. During our investigation of the report, we discovered that the violator is likely using ffmpeg as well. [20:18] <keynote2k> I'd like to report this possible violation to the proper channels. Where should I send the report? Please advise. [20:20] <klaxa> ffmpeg-devel list might be the right place [20:20] <klaxa> the mailing list that is [20:21] <sacarasc> There used to be a fun page on the website about projects doing that... [20:21] <sacarasc> Can't find it now, though. [20:25] <sfan5> fixedtype: https://lists.ffmpeg.org/pipermail/ffmpeg-user/2013-June/015662.html [20:33] <keynote2k> klaxa: is the ffmpeg-devel list a better place to go than the #ffmpeg-devel IRC channel? [20:33] <klaxa> maybe do both? [20:33] <keynote2k> sacarasc: I had heard there was a website and/or a dedicated email address; I couldn't find it. [20:33] <keynote2k> klaxa: worth a shot. [20:34] <keynote2k> Thanks for your responses. :) [20:34] <klaxa> if it's about licensing, emails seem more apropriate [20:34] <fixedtype> sfan5: this works with ffmpeg but not with ffplay. ffplay doesn't have -filter_complex option [20:35] <sfan5> you could try ffmpeg <options> -vcodec huffyuv -f matroska - | ffplay - [20:35] <sfan5> something like that might work [20:35] <sfan5> ^ fixedtype [20:39] <fixedtype> sfan5: yes, this works, although pipe seeking is not possible with pipe. [20:39] <fixedtype> http://ffmpeg.org/ffmpeg-filters.html#movie [20:39] <fixedtype> i think i've found the workaround [20:51] <Filarius> why ffmpeg do so many fps ? http://pastebin.com/gDQs6HZM [20:53] <c_14> Do you mean that fps counter on the bottom? [20:53] <c_14> That's your encoding fps, how many frames you're encoding per second. It has nothing to do with your playback fps. [20:54] <Filarius> oh, thanks [20:58] <Filarius> hm [20:59] <Filarius> I do something wrone, I save it as flv file and after few seconds I have about 2 hour movie [20:59] <c_14> How many input images do you have? [21:00] <Filarius> I need take 2 pictures per second and stream it like 20 fps video [21:00] <c_14> I'm pretty sure -loop 1 isn't doing what you think it's doing. [21:02] <Filarius> I want make live streaming of pictures what generated on the fly. Just I`m too dummy to add streaming inside of my application [21:03] <Filarius> so as test I make "1.jpg" and "2.jpg" and trying to stream it [21:03] <c_14> ffmpeg -framerate 1 -i %01d.jpg [output optinos] rtmp://localhost:1935/live [21:04] <c_14> Should give you 40 frames output video. [21:05] <Filarius> what about loop ? if I want stream only this two images [21:06] <c_14> afaik loop will only loop over a single image. [21:06] <Filarius> and replace it with another images (same name) [21:06] <Filarius> no, it good for sequence [21:07] <Filarius> at last it work, just I see big fps [21:07] <c_14> ye, like I said before. That should be fine. That's just the encoding fps. Or do you mean in the output file/stream? [21:09] <Filarius> if I save to flv file instead of stream I have file with hours of video, and it was generated in seconds [21:10] <c_14> Well, you're looping forever. And encoding two frames forever isn't all that difficult. [21:10] <c_14> Add something like -t 5:00 to the output options and it'll stop after 5min [21:10] <Filarius> but I need livestreaming... [21:11] <c_14> just for when you're outputting to the file... [21:13] <Filarius> so for streaming it must be okay ? I think I found strange behavior with rtmp stream, so it looks like same with file - it not show changes even of think about big buffers [21:14] <Filarius> if I replace files with another one [21:15] <c_14> I'm guessing that if you use -loop 1, ffmpeg just loads the files into memory and then never queries if they've changed. [21:17] <Filarius> is here trick to make live encoding/streaming to file ? [21:18] <Filarius> so ffmpeg will not do it such fast [21:20] <c_14> try adding -re as an input option? but I'm still not sure what you're trying to do. [21:24] <Filarius> ok, from start point - I want generate jpeg files from my application, and use ffmpeg to stream it on the fly, so it will be like pretty dummy way to live stream from own application. First I thought to save frames to jpeg file with same name [21:25] <c_14> just save them with increasing numbers ie 00000.jpg 000001.jpg etc, then ffmpeg -re -framerate 1 -f image2 %04d.jpg [output options] [21:25] <c_14> (note, I didn't actually count how many zeros I used) [21:27] <Filarius> but this way somehow I must know what last image was processed by ffmpeg [21:27] <c_14> Hmm, why? [21:28] <Filarius> what ffmpeg will do if no more image left ? [21:28] <c_14> stop [21:29] <Filarius> so if my software somewhy will not create new image right in time- stream will be ended, i do not want this [21:30] <Filarius> also too many images not so good, if I will want to make long time stream with good framerate [21:31] <ogrgkyle> Hello again [21:31] <Filarius> I had to work with folder with +50000 images on usual HHD, it`s not so nice [21:32] <ogrgkyle> I have solved the issue of the "undefined reference" errors, I think. [21:32] <Filarius> *HDD, sorry for misstypes [21:32] <ogrgkyle> Now there's one error left: ld.exe cannot find -lswscale.lib [21:32] <ogrgkyle> Do you know what this is about? [21:33] <c_14> Filarius: just delete the older pictures as you go along? [21:33] <Filarius> but what if I will delete images what was not processed ? [21:34] <c_14> ffmpeg should eat one frame per second, keep around 60 frames and you should be fine as long as your app generates frames at about 1 fps [21:35] <Filarius> I do not want to make timing be boss here [21:37] <Filarius> I know my situation pretty synthetic, sorry if I can look like annoying person [21:38] <Filarius> I want mike live stream what show (almost) real time video, 60 seconds delay will be too long [21:42] <c_14> you might be better of muxing the pictures into some sort of video codec like mjpeg and then feeding that to ffmpeg [21:43] <iamtakingiteasy> hi, which options would you recommend for white/black (even not grayscale) x264-encoded video of many small volatile text characters (ascii-art video) for real-time streaming to rtmp server? [21:44] <iamtakingiteasy> for some reason i am expriencing massive lags on some clients on -maxbitrate >100k and -crf<40 [21:44] <iamtakingiteasy> but it is nearly unwatchable on such low bitrates [21:45] <iamtakingiteasy> -maxrate, sorry [21:45] <Filarius> I just tested it with 1 image file in sequence, with replacing it with 3 else images, and it save to file pretty in right way, just too many frames per seconds... [21:45] <Filarius> it`s with loop [21:45] <c_14> Filarius: -re -framerate 1 should limit ffmpeg to processing only 1 frame per second [21:46] <c_14> iamtakingiteasy: you're probably going to need a relatively high bitrate because of the fast movement of the chars so it doesn't blur, what kind of lag are you experiencing? [21:47] <iamtakingiteasy> c_14: video plays fine on client (when buffer is filled, which takes about 5-10 seconds), plays for a while and then stops (sometimes only video stops while audio is playing fine), then vidoe jumps without plaing and stops again [21:48] <iamtakingiteasy> err [21:48] <iamtakingiteasy> the first thing when it plays lasts not longer than for 15 seconds [21:49] <c_14> what if you were to output to a file and then play that file, same problems? [21:49] <iamtakingiteasy> hmm, can't really say, i am encoding&streaming on headless system; i'll try to setup Xvnc now [21:50] <Filarius> Oh thanks c_14, can I kiss you? I`m so stupid. Now it works who I need, at last with file, will check it with stream now [21:59] <beastd> Filarius: Sorry, I am not sure what exactly you are trying to accomplish, but maybe image2pipe could help with your producer/consumer synchronization problem? See e.g. at the end of this FAQ for a simple example: http://www.ffmpeg.org/faq.html#How-do-I-encode-single-pictures-into-movies_… [22:00] <Filarius> I think pipes not available in OS Windows [22:01] <iamtakingiteasy> c_14: yes, the same problem persists when played from local file [22:01] <Filarius> just think about it like bad(easy) way to make rtmp streaming from own software [22:02] <Filarius> not production, just hobby and curious [22:02] <c_14> iamtakingiteasy: can you pastebin the command you're using to make the video and if possible a sample file with which the problem persists? If not a paste of the ffprobe output maybe? [22:06] <iamtakingiteasy> the command i am using is: [22:06] <iamtakingiteasy> ffmpeg -re -i 'VTS_01_1.VOB' -vf 'scale=iw/3:ih/4, aa=contrast=50:fontsize=10:fontname=liberation mono' -crf 30 -vcodec libx264 -pix_fmt yuv420p -preset ultrafast -ar 44100 -f flv ~/out.flv [22:07] <iamtakingiteasy> the sample file: http://static.eientei.org/data/out.flv [22:09] <iamtakingiteasy> ffprobe output: http://paste.eientei.org/show/225/ [22:10] <c_14> And you are experiencing the stuttering with this sample file? [22:11] <iamtakingiteasy> yes, on my clients which supposed to decode this as real-time stream [22:12] <c_14> The file plays just fine for me. [22:12] <c_14> not stuttering. [22:12] <iamtakingiteasy> hmm [22:26] <znf> [22:40] <beastd> Filarius: AFAIK pipes are available in Windows since somewhen in the 80s and still supported today. No guarantee about any behavioural differences that might matter in your use case though. [22:42] <fixedtype> when ffplay sends output via xv does it prescale 720x480 (DAR 4:3) movie to "square" pixels or is the scaling done by hardware? how can i tell from output? [22:43] <fixedtype> the output tells me: w:720 h:480 sar:0/1 -> w:720 h:480 sar:0/1 [22:44] <fixedtype> but i get correct aspect ratio on display. does it mean the scaling is done in hardware? [22:44] <fixedtype> also what does 0/1 value for sar mean? [22:45] <fixedtype> actual sar value is 8:9 [22:45] <sacarasc> Screen aspect ratio of 8 across for every 9 up. [22:50] <fixedtype> sacarasc: yes, i know what it means. i'm trying to investigate who's in charge of scaling correction [22:51] <fixedtype> i was referring to odd value for sar -- 0/1 [22:55] <fixedtype> i also confused xv (as in mplayer) for sdl in the previous message. ffplay is sdl based [23:04] <ogrgkyle> Hey folks [23:04] <ogrgkyle> Very basic question here [23:04] <ogrgkyle> I want to build the FFmpeg libraries [23:05] <c_14> https://trac.ffmpeg.org/wiki/CompilationGuide [23:05] <ogrgkyle> Is "compiling FFmpeg" the same as "building FFmpeg libraries"? [23:06] <sacarasc> They get compiled along with the ffmpeg executable. [23:06] <sacarasc> You can pass a switch to the configure to not compile that bit if you don't want it. [23:06] <ogrgkyle> Oh ok [23:06] <_aeris_> ohai everybody ! [23:07] <ogrgkyle> What is cross compiling? [23:07] <_aeris_> i try to concatenate 2 files in webm (vp8 576p + mono vorbis) [23:07] <_aeris_> but the result is very ugly « iron » sound [23:08] <_aeris_> concatenate 2 files with stereo : ok, 2 files on mono : nok :( [23:08] <c_14> ogrgkyle: compiling something on one platform for use on a different platform [23:11] <_aeris_> https://paste.imirhil.fr/?86e551d1e63566fa#X4ByEWhuchB7gh+GkYUec01Lg4kAB5pz… [23:13] <Nitrodist> anyone know why I get an sample_aspect_ratio of 0:1 (same for display_aspect_ratio) when I convert a gif to mp4 with these options? https://gist.github.com/Nitrodist/cadcf4c6949d903dc6b7 [23:14] <Nitrodist> the sampel_aspect_ratio is determined from ffprobe -show_streams [23:14] <c_14> _aeris_: And the command? [23:15] <_aeris_> c_14 > what command ? [23:15] <_aeris_> ffmpeg -y -f concat -i /tmp/concat20140624-29289-15dd8ul -c copy transition-libre-experience-ong.webm [23:15] <_aeris_> this command, in the pastebin ? [23:15] <ogrgkyle> Is there a way to download FFmpeg prebuilt with GCC 4.7.1? [23:19] <fixedtype> ogrgkyle: it's probably easier to build it yourself [23:20] <c_14> Nitrodist: can you paste the output of ffprobe ? [23:21] <c_14> _aeris_: I don't know. Might be a bug. Does it have the same problems if you reencode the audio? [23:21] <Nitrodist> c_14: https://gist.github.com/Nitrodist/a1cdbfaac1698e865f43 [23:23] <c_14> Nitrodist: try adding setsar=1 after the scale filter? [23:27] <Nitrodist> c_14: sorry, I'm not sure how to append that to the end of that [23:27] <c_14> *2,setsar=1' [23:27] <Nitrodist> ah, comma, cool [23:29] <Nitrodist> c_14: well, it's set now -- I'll see if it fixes my problem. Thanks a lot! [23:30] <debianuser> _aeris_: If instead of concatting 2 different files you specify the same file twice does the sound gets concatted correctly? [23:33] <_aeris_> debianuser > yep [23:44] <debianuser> _aeris_: Both of them? Have you checked both file1+file1 and file2+file2 sounds? [23:45] <_aeris_> one of the file is a generated one with « no » sound [23:45] <_aeris_> fix image + /dev/zero for sound [23:45] <_aeris_> difficult to say if good or not :P [23:49] <debianuser> _aeris_: Heh... It's just '-f concat' is very picky about what it wants to concat. I've hit this recently when tried to concat wav+ogg. I ended using concat filter, e.g.: ffmpeg -i 1.wav -i 2.ogg -i 3.wav -filter_complex concat=n=3:v=0:a=1 0.wav [23:50] <_aeris_> not able to reencode for me [23:50] <_aeris_> too long [23:50] <_aeris_> and bad quality [23:58] <debianuser> I just tried that with two 24kHz mono ogg/vorbis `ffmpeg -i file1.ogg -i file2.ogg -filter_complex concat=n=2:v=0:a=1 concat.ogg` worked fine for me... [23:59] <_aeris_> for 3min sound, perhaps [00:00] --- Wed Jun 25 2014

1 0

ffmpeg-devel.log.20140623
by burek 24 Jun '14

24 Jun '14

[00:00] <Timothy_Gu> debianuser: yep, it's the ogg's problem [00:01] <Timothy_Gu> It is actually 0.02 seconds long, not 3.43 [00:01] <Timothy_Gu> try ffmpeg -i 2.ogg -f null - [00:01] <Timothy_Gu> Also, there is a warning: [ogg @ 0xafbf3c0] Estimating duration from bitrate, this may be inaccurate [00:02] <Timothy_Gu> that explains the inaccuracy of the duration of 2.ogg [00:02] <Timothy_Gu> All in all, the output file from concat filter is completely fine [00:02] <debianuser> If it's 0.02 seconds long then how could I hear it? [00:02] <debianuser> ;) [00:04] <nevcairiel> obviously your ogg file is broken [00:06] <Timothy_Gu> Hey also nevcairiel: mind if you send a patch disabling all those stupid anonymous struct/union warnings for MSVC? [00:07] <nevcairiel> i'm not at home for the next two weeks, i'm not going to send anything [00:07] <Timothy_Gu> warning C4201: nonstandard extension used : nameless struct/union [00:07] <Timothy_Gu> oh, ok [00:07] <nevcairiel> also, some people would say that you better name them [00:08] <Timothy_Gu> in C11 anonymous unions are supported [00:08] <nevcairiel> but we use C99 [00:09] <Timothy_Gu> i know, but the flood of this kind of warnings really masks the warnings we should care about [00:11] <nevcairiel> flood? its maybe 30, and 20 of those are all collected in dxva2api.h (which is a C++ header, where this is technically allowed) [00:12] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:c69f7299473d: ffmpeg: fix reinitializing with threads while flushing [00:13] <michaelni> ubitux, your hqx filter tests are evil ;) [00:14] <debianuser> Timothy_Gu: It seems you're right. It's still a bug in ffmpeg, but in a different place: `ffmpeg -i 1.wav -i 2.ogg -i 3.wav -filter_complex concat=n=3:v=0:a=1 -ac 2 -ar 48000 0.wav` at the end says "time=00:00:23.83", but `ffprobe 0.wav` shows: "Duration: 00:00:25.88". So the output file is correct, but the output TEXT is not. [00:18] <Timothy_Gu> This is because ffmpeg calculates the output duration for file header in advance with the incorrect ogg duration [00:18] <cone-826> ffmpeg.git 03Diego Biurrun 07master:49470b940458: build: Add missing replaygain object file for CAF demuxer [00:18] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:2ca4802601c2: Merge commit '49470b9404583ab3aff6835f78a759349d331696' [00:20] <Timothy_Gu> michaelni: any idea why they fail on ARM? [00:21] <Timothy_Gu> and FreeBSD? [00:22] <michaelni> hqx seems using floats init, this might cause some failure [00:23] <Timothy_Gu> floats are evil [00:24] <michaelni> for bitexact regression tests yes [00:25] <michaelni> loosing the floats should be easy, ill look at it in an hour or 2 if noone else did it by then [00:25] Action: J_Darnley recognises those constants [00:28] <debianuser> Timothy_Gu: ffmpeg says "time=00:00:23.83" at the end, after it's finished encoding. And if that was calculated from incorrect ogg length, that would be 27.24 (10.92+3.43+12.89), not 23.83. Anyway, incorrect output text is not too bad bug for me, I just wasn't aware of it. [00:29] <Timothy_Gu> Oh, well [00:30] <Timothy_Gu> At least you now know concat is working :) [00:31] <cone-826> ffmpeg.git 03Diego Biurrun 07master:03391b5432a2: build: Add missing object files for Matroska demuxer [00:31] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:738272826f4b: Merge commit '03391b5432a2b8b8dbc3d2db722a54dfb68737b9' [00:37] <cone-826> ffmpeg.git 03Diego Biurrun 07master:85d805d5cccb: build: Add missing object files for webm muxer [00:37] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:9e8c92226fe1: Merge commit '85d805d5cccb2f57e7f2e2254d7a870f05926cfd' [00:38] <ubitux> michaelni: it seems they help finding bugs everywhere ;) [00:38] <ubitux> for the float i could have a look, but not now [00:39] <ubitux> probably this week [00:39] <ubitux> do you have mismatches because of the floats? [00:40] <michaelni> ubitux, i dont know, ive not reproduced any yet [00:41] <michaelni> i was thinking its easier to just get rid of the floats than to test if they do cause any of the remaining problems [00:44] <debianuser> Timothy_Gu: Yes, thank you for your help. And for the great program you all are creating here. [00:44] <debianuser> While I'm here, I also have a lot of wav files (from rift game resources), which are basically "adpcm_ima_wav", but stereo files have blocks interleaved, not nibbles, like in "adpcm_ima_qt". Are you interested in supporting such files? If yes, how can I help? [00:45] <cone-826> ffmpeg.git 03Diego Biurrun 07master:c54e118722cb: build: Have the eatqi decoder depend on the MPEG-1 decoder [00:45] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:9d91fae8a109: Merge commit 'c54e118722cbbdc04945538d1796d4472a1ff406' [00:51] <cone-826> ffmpeg.git 03Diego Biurrun 07master:fbf4d2197f35: build: Record explicit dependency of intrax8 on error resilience [00:51] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:7695a538ed24: Merge commit 'fbf4d2197f35bddccec00ad53185ee4ab81778a0' [00:53] <Timothy_Gu> debianuser: you should probably open a ticket on trac.ffmpeg.org . No guarantee on when we will implement it, but at least we will have a record and samples for that. [00:57] <cone-826> ffmpeg.git 03Diego Biurrun 07master:a6a675a79287: configure: Drop stray error resilience dependencies [00:58] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:ffc2c095ef7c: Merge commit 'a6a675a79287565c3c23455439b114676b63a96c' [01:01] <ubitux> michaelni: i'm removing the float, don't worry about it [01:02] <michaelni> ok [01:02] <michaelni> thx [01:03] <cone-826> ffmpeg.git 03Diego Biurrun 07master:8d695dc411c1: mpeg: Mark ff_mpeg_set_erpic() static [01:03] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:412015694c08: Merge commit '8d695dc411c1ae28477f107d6f7bf2db59af95df' [01:12] <cone-826> ffmpeg.git 03Diego Biurrun 07master:f83896abda1e: ape: Replace memset(0) by zero initialization [01:12] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:efec73043184: Merge commit 'f83896abda1e78ebbbda0f184b682b4fabadc682' [01:19] <cone-826> ffmpeg.git 03Diego Biurrun 07master:73953df71a28: Replace av_malloc() and memset(0) by av_mallocz() [01:19] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:32365667cceb: Merge commit '73953df71a28e94d8959facba7906cf5975ca4ef' [01:25] <kierank> For some reason ff_hevc_decode_nal_pps is slow on this file [01:26] <nevcairiel> Decoding the pps is slow? [01:27] <nevcairiel> Isn't that like a few bytes of flags? [01:27] <wm4> maybe the file consists entirely of repeated PPSs [01:28] <cone-826> ffmpeg.git 03Luca Barbato 07master:0ef256d51518: libx264: Correctly manage constant rate factor params [01:28] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:49f1b869b60f: Merge commit '0ef256d51518f0e483c38b9ee3b993cf0709d00e' [01:36] <cone-826> ffmpeg.git 03Diego Biurrun 07master:463a7cde563f: Differentiate assembler and assembly in documentation and comments [01:36] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:a125f87ac49b: Merge commit '463a7cde563fd805864c48a76dd1b03fc24671ed' [01:44] <kierank> https://www.dropbox.com/s/9bvezkhvrv8ne4k/BBC4k.002.ts [01:44] <kierank> nevcairiel: yeah it's weird [01:49] <ubitux> michaelni: http://pastie.org/pastes/9315169/text [01:49] <ubitux> a bit ugly but this seems to do the trick [01:50] <michaelni> ubitux, LGTM [01:50] <michaelni> ubitux, or you could divide by 1000 [01:50] <michaelni> and use prettier numbers [01:51] <michaelni> hmm no [01:51] <michaelni> i realize te array is a bit big so divide would possibly suck even for init [01:52] <ubitux> i'm not sure about the bithack because of signdness [01:52] <ubitux> and my brain is off :( [01:52] <ubitux> it's late, gtg [01:54] <michaelni> ubitux, sleep well in that case, dont worry about it [02:06] <kierank> smarter: any idea how to optimise a slow ff_hevc_decode_nal_pps [02:21] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:42e6fc147b5a: avcodec/apedec: optimize sign combination [02:21] <cone-826> ffmpeg.git 03James Almer 07master:c172683bf46a: x86/dsputil: remove redundant global motion compensation code [03:25] <cone-826> ffmpeg.git 03Justin Ruggles 07master:44127546b0a8: Check if an mp3 header is using a reserved sample rate. [03:25] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:2bcf34a0ef60: Merge commit '44127546b0a81dc9dd6190739a62d48f0044c6f3' [03:43] <cone-826> ffmpeg.git 03Justin Ruggles 07master:f2f2e7627f0c: Check mp3 header before calling avpriv_mpegaudio_decode_header(). [03:43] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:ca350378de06: Merge commit 'f2f2e7627f0c878d13275af5d166ec5932665e28' [04:02] <debianuser> Found another problem with concat filter. Is it expected that `ffmpeg -i mono8kHz.wav -i stereo48kHz.wav -ac 2 -ar 48000 -filter_complex concat=n=2:v=0:a=1 concat.wav` produces a file that sounds like 8kHz mono? Isn't concat filter supposed to select "common" sampling rate and channel layout? [04:06] <cone-826> ffmpeg.git 03Diego Biurrun 07master:7b9ef8d701c3: mpeg: Split error resilience bits off into a separate file [04:06] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:56afbe8dbc6e: Merge commit '7b9ef8d701c319c26f7d0664fe977e176764c74e' [04:06] <nevcairiel> debianuser: it probably just sticks to whatever came first [04:07] <nevcairiel> anyway whats "common"? one needs upsampling, one downsampling. For us its obvious that upsampling may be the better idea, but the filter may not know that :p [04:13] <michaelni> id say the filter should avoid loosing information by default and downsample would loose info, a patch that improves this would be welcome [04:20] <debianuser> Hm. So it ignores those "-ac 2 -ar 48000" options... Then I'll use a workaround: ffmpeg -i mono8kHz.wav -i stereo48kHz.wav -filter_complex aresample=48000:och=2,concat=n=2:v=0:a=1 concat.wav [04:22] <nevcairiel> those options are probably applied after the filter_complex before the final output [04:22] <nevcairiel> its not smart enough to combine filter_complex with any other options really [04:52] <Timothy_Gu> Yeah. I personally consider -ac -ar -s and other similar options half deprecated, especially when using filtering. [11:25] <plepere> what's the time function to test if a function is quicker ? [11:26] <plepere> I did an AVX version of a function, but I think it's slower than the SSE2 version [11:26] <J_Darnley> The {START,STOP}_TIMER macros? [11:26] <plepere> ah great, thanks [11:27] <J_Darnley> Just make sure the block/fucntion you time processes the same amount of data each time it is called. [11:27] <plepere> well I can just run the same videos [11:28] <J_Darnley> No [11:29] <J_Darnley> I mean the function needs to process the same number of samples everytime it is run. [11:29] <plepere> it's a function on a specific block size [11:30] <J_Darnley> If it processes 30 then 10 then 20 then 10 it will give a nonsense result [11:30] <J_Darnley> goog [11:30] <J_Darnley> good [11:38] <plepere> yeah, AVX is slower. :( [12:10] <kierank> plepere: 12:44 am " kierank https://www.dropbox.com/s/9bvezkhvrv8ne4k/BBC4k.002.ts [12:10] <kierank> Some bizzare benchmarking on this filw [12:11] <kierank> File [12:56] <plepere> kierank, you want me to run some benches on that ? [12:56] <kierank> plepere: you can but the weirdest bit is that a lot of time is spent on PPS parsing [12:56] <plepere> it's hevc ? [12:57] <kierank> yes [12:57] <kierank> plepere: see pm [12:58] <kierank> plepere: I've written a small patch to the slow part of the pps parsing but it doesn't make much difference yet [12:59] <kierank> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/hevc_ps.c;h=30e58… [12:59] <kierank> these two loops are sloow [13:03] <kierank> I tried changing the multiplies into shifts but nothing happened [13:04] <plepere> ok [13:33] <kierank> plepere: http://obe.tv/about-us/obe-blog/item/12-first-look-at-bbc [13:33] <kierank> wrote a small blog post about it [13:35] <kierank> also I thinks some of the functions need inlining [13:35] <plepere> the most significant missing optimisations in ffmpeg are the whole idct transforms [13:35] <kierank> yeah I mention that in my post [13:36] <plepere> if only we could use intrinsics in ffmpeg. [13:36] <iive> hum? [13:37] <plepere> we're even faster with intrinsics than with assembly. :/ [13:39] <iive> we have bad past experience with intrinsics. [13:40] <plepere> :( [13:40] <iive> and i don't understand the remark about missing whole idct tranforms. [13:40] <kierank> iive: we have no hevc idct asm [13:40] <plepere> we have the idct transforms in intrinsics and in C, but not in ASM [13:41] <plepere> in openhevc [13:42] <plepere> and for ffmpeg to be able to do real-time 4K, someone needs to translate the intrinsics to ASM [13:42] <iive> aha h265 [13:43] <iive> why intrinsics are not committed in ffmpeg in the meantime? [13:43] <plepere> we were told that intrinsics were more or less banned from ffmpeg [13:43] <kierank> they are [13:43] <J_Darnley> Why duplicate the effort? Why not use openhevc as a library in ffmpeg? [13:44] <kierank> J_Darnley: openhevc is a copy of libavcodec [13:44] <kierank> with intrinsics [13:44] <kierank> and other scalable features [13:44] <J_Darnley> oh [13:45] <plepere> the scalable is so good. We've got multi-view coming along too. :D [13:46] <plepere> but yeah, being able to add intrinsics as a temporary solution would really boost the performance [13:47] <plepere> at least enough to bring things to realtime [13:48] <iive> aren't BBB and ubitux working on the asm idct at the moment? [13:48] <plepere> vp9 [13:48] <plepere> and it's with a different way of doing it [13:50] <iive> well, imho, you should send RFC to the maillist asking if it is ok to commit the intrinsics, until somebody comes forward and rewrites them in ASM. [13:51] <plepere> I think things have already been negociated by smarter, michaelni and all. [13:52] <iive> what have been negotiated? [13:52] <plepere> well about the "no intrinsics in ffmpeg" [13:53] <cone-622> ffmpeg.git 03Diego Biurrun 07master:c67b449bebbe: dsputil: Split bswap*_buf() off into a separate context [13:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07master:35bb74900b95: Merge commit 'c67b449bebbe0b35c73b203683e77a0a649bc765' [13:56] <iive> afaik, intrinsics are avoid because they are kind of volatile. compilers might generate horribly inefficient code if some options are tweaked, and sometimes miscompile them. [13:57] <iive> e.g. ppc assembly is entirely written in intrinsics and it is full of hacks for one version of compiler or another. [13:57] <plepere> :/ [13:58] <BBB> iive: no [13:58] <BBB> iive: no intrinsics [13:59] <iive> that's why we prefer an assembly that is guaranteed to produce consistent results and doesn't need compiler hacks. [13:59] <iive> BBB: ok :) [14:00] <BBB> and Im happy to commit intermediate work for all hevc asm [14:00] <BBB> but no intrinsics [14:00] <iive> that's why I asked if you are working on it. [14:00] <kierank> plepere: it's probably the one thing both sides agree on [14:00] <kierank> out of all the hatred [14:01] <ubitux> don't yell it too loud, it could be a reason to disagree [14:01] <BBB> so true [14:01] Action: iive imagines all libav assembly written in intrinsics. [14:01] <BBB> iive: Im not working on hevc asm. honestly, if someone wants me to work on it, they probably have a commercial interest in it, and they should pay me marketrate; else, they are free to use vp9 [14:02] <plepere> why so much hate ? [14:02] <kierank> plepere: blog.pkh.me/p/13-the-ffmpeg-libav-situation.html [14:03] <ubitux> create mode 100644 vp9/encoder/vp9_speed_features.c [14:03] <ubitux> mmh [14:04] <cone-622> ffmpeg.git 03Dirk Ausserhaus 07master:e121ac634ba3: indeo45: use is_indeo4 context flag instead of checking codec ID [14:04] <cone-622> ffmpeg.git 03Michael Niedermayer 07master:b222e0773c78: Merge commit 'e121ac634ba324a318f4a97f978dcfb48da6b735' [14:13] <plepere> kierank, thanks for the read. :) [14:15] <kierank> plepere: any idea why nal_pps is so slow on ffmpeg but not in openhevc [14:15] <kierank> even though it's the same code [14:22] <plepere> the pps in openhevc was just optimized recently [14:22] <kierank> how do I do that [14:22] <kierank> whoops [14:32] <cone-622> ffmpeg.git 03Michael Niedermayer 07master:d6af26c55c1e: avutil/lzo: Fix integer overflow [14:34] <ubitux> libav, first on security [14:34] <ubitux> ;) [15:03] <cone-622> ffmpeg.git 03Luca Barbato 07master:ccda51b14c0f: lzo: Handle integer overflow [15:03] <cone-622> ffmpeg.git 03Michael Niedermayer 07master:cf2b7c01f81c: avutil/lzo: add asserts to be double sure against overflows [15:03] <cone-622> ffmpeg.git 03Michael Niedermayer 07master:957457a044f6: Merge commit 'ccda51b14c0fcae2fad73a24872dce75a7964996' [15:49] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.2:7d9c059a3525: avutil/lzo: Fix integer overflow [15:49] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.2:61796a8999f5: avutil/lzo: add asserts to be double sure against overflows [16:02] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.2:63e3a9781579: avcodec/aic: fix quantization table permutation [16:03] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.2:e173834af8b5: avformat/flvenc: Do not allow creating h263/mpeg4 in flv without unofficial format extensions being enabled. [16:03] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.2:dad0c9d686a3: avcodec/alsdec: Clear MPEG4AudioConfig so that no use of uninitialized memory is possible [16:03] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.2:252a0ccb80b4: cavsdec: fix qp fixed slice handling [16:03] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.2:e4cdde96b381: avformat/cavsvideodec: Fix probing when the file extension is avs [16:03] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.2:d07be523f5d8: avformat/mpc: attempt to allocate a packet that is not smaller than the data inside it [16:03] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.2:e0a03d1f9cb1: update for FFmpeg 2.2.4 [16:44] <cone-622> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n2.2.4': unknown revision or path not in the working tree. [16:44] <cone-622> Use '--' to separate paths from revisions [16:44] <cone-622> refs/tags/n2.2.4:HEAD: Merge commit 'ccda51b14c0fcae2fad73a24872dce75a7964996' [17:21] <kierank> mraulet: can you send that pps patch please =p [17:23] <J_Darnley> Damn! When explorer crashes, it crashes HARD! [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:946810d1da50: ffmpeg: dont call exit_program() from a signal hander [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:4201c7454551: avformat/mp3enc: use av_copy_packet() [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:2b5b21dbaca0: avcodec: Add padding after the remaining AVFrames [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:b8f6069bf398: swscale/swscale: fix integer overflow [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:998b5ac3a043: dox/scaler:fix bicubiclin typo [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:380587cc34c2: avcodec/x86/mpegvideoenc_template: fix integer overflow [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:61e5001b265b: avcodec/h264_mp4toannexb_bsf: prepend global headers before any in stream parameter sets [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:e4f9a5efa15a: avcodec/libx264: move where x264opts is applied down so it isnt overridden by avctx & defaults [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:4c9c76af8521: avfilter/vf_pullup: fix gray8 [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:c5085a030143: avcodec/h264: clear cur_pic structure instead of duplicating it in ff_h264_update_thread_context() [17:56] <cone-622> ffmpeg.git 03Lukasz Marek 07release/2.1:46b64d981884: lavu/opt: validate range before dereference [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:861732246ff2: avcodec/wma: use av_freep(), do not leave stale pointers in memory [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:9c83914a1fd6: avcodec/msrle: check return code for success before use [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:37f854f5b958: swresample/resample: Limit filter length [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:15871a90aa42: swresample/dither: use av_malloc_array() [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:9d8e2314592a: swresample/resample: use av_malloc_array() where appropriate [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:672919388651: swscale/x86/swscale_template: loose hardcoded dstw_offset [17:56] <cone-622> ffmpeg.git 03Anthoine Bourgeois 07release/2.1:26d72b3df0ce: avcodec/dirac_arith: Fix build with PIC and stack-check options [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:9c358c6e3b34: avutil/lzo: Fix integer overflow [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:c33770e74bfc: avutil/lzo: add asserts to be double sure against overflows [17:56] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:d2e3afbbd167: avcodec/x86/idct_sse2_xvid: fix non C99 inline function [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:522865543819: swscale/swscale: fix srcStride/srcSlice typo [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:861e453996f4: avcodec/mjpegen: Fix declared argument size [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:1d9125137039: avformat/mpegts: Remove redundant check [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:ec2ce83913ca: avcodec/diracdec: fix undefined behavior with shifts [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:af109ff12581: avcodec/g723_1: add assert to help static code analyzers [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:90a384cde163: avfilter/f_select: fix loss of precission in SAD calculation [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:4eb0b6c59051: avfilter/vf_deshake: fix loss of precission with odd resolutions [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:df208d6ab939: iavcodec/vc1dec: Fix missing {} [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:9cde200ba70f: avfilter/filtfmts: Support dynamically allocated in/outputs [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:ac182d6d8b50: avformat/mux: Check for and remove invalid packet durations [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:d89e5b20f2b9: avformat/h263dec: Fix h263 probe [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:2b45eab4c913: ffmpeg_filter: fix pointer to local outside scope [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:aaf6ddfa4d9d: avcodec/mjpegdec: Fix undefined shift [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:2a543e58ec52: avfilter/graphdump: Fix pointer to local outside scope [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:2572372f909a: avcodec/diracdec: move mc buffer allocation to per frame [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:6a51bd7360cb: avutil/cpu: force mmx on selection of higher x86 SIMD features [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:e3b6e23a0f64: avcodec/libvorbisenc: dont add the duration to AV_NOPTS_VALUE [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:76838fb171ab: avcodec/aac: fix () in IS_CODEBOOK_UNSIGNED macro [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:60ed43933402: avcodec/golomb-test: fix () in EXTEND() macro [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:a71b4209c26a: avcodec/h264: fix () in macros [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:ccbcb17960a3: avcodec/ivi_dsp: add some missing () to macros [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:f1feb7e388c4: avcodec/mlpdec: fix () in MSB_MASK() macro [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:3065919b6566: avcodec/mss34dsp: fix () in SOP* macros [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:66a9c50d42e0: avcodec/mss4: Fix () in MKVAL() macro [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:4d1346d46acc: avcodec/aic: fix quantization table permutation [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:d192ff0ece61: avformat/flvenc: Do not allow creating h263/mpeg4 in flv without unofficial format extensions being enabled. [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:c2c922460aa4: avcodec/alsdec: Clear MPEG4AudioConfig so that no use of uninitialized memory is possible [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:a45a66599143: avformat/cavsvideodec: Fix probing when the file extension is avs [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:0e3a1d60d393: cavsdec: fix qp fixed slice handling [17:57] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:9790ed63fcd2: avformat/mpc: attempt to allocate a packet that is not smaller than the data inside it [17:58] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/2.1:e7873dfccad5: update for FFmpeg 2.1.5 [18:12] <cone-622> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n2.1.5': unknown revision or path not in the working tree. [18:12] <cone-622> Use '--' to separate paths from revisions [18:12] <cone-622> refs/tags/n2.1.5:HEAD: Merge commit 'ccda51b14c0fcae2fad73a24872dce75a7964996' [18:52] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:1be06c853182: swscale/swscale: fix integer overflow [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:92ca3578bf40: avcodec/x86/mpegvideoenc_template: fix integer overflow [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:2af3a0416453: avcodec/h264_mp4toannexb_bsf: prepend global headers before any in stream parameter sets [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:e897eb108b51: avcodec/libx264: move where x264opts is applied down so it isnt overridden by avctx & defaults [18:53] <cone-622> ffmpeg.git 03Lukasz Marek 07release/1.2:8f7ebb88b816: lavu/opt: validate range before dereference [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:e5546d48717f: avcodec/wma: use av_freep(), do not leave stale pointers in memory [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:a61998e4f4e1: avcodec/msrle: check return code for success before use [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:ce959040135d: swresample/resample: Limit filter length [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:fe97363c38a9: swresample/dither: use av_malloc_array() [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:4065e557593b: swresample/resample: use av_malloc_array() where appropriate [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:4af56488c86e: swscale/x86/swscale_template: loose hardcoded dstw_offset [18:53] <cone-622> ffmpeg.git 03Anthoine Bourgeois 07release/1.2:90cf672d7a49: avcodec/dirac_arith: Fix build with PIC and stack-check options [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:98fe429ba555: avcodec/x86/idct_sse2_xvid: fix non C99 inline function [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:71104f7cb0fb: swscale/swscale: fix srcStride/srcSlice typo [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:7b38ea4003ed: avcodec/mjpegen: Fix declared argument size [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:424b00a30021: avformat/mpegts: Remove redundant check [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:90c7dfb907ee: avcodec/diracdec: fix undefined behavior with shifts [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:53c02d04e495: avcodec/g723_1: add assert to help static code analyzers [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:8d60d8bef228: avfilter/vf_deshake: fix loss of precission with odd resolutions [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:85b00d798fd1: iavcodec/vc1dec: Fix missing {} [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:753f0738cae9: avfilter/filtfmts: Support dynamically allocated in/outputs [18:53] <cone-622> ffmpeg.git 03Xi Wang 07release/1.2:9ade5b804ea1: lzo: fix overflow checking in copy_backptr() [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:eada99a854f8: avutil/lzo: Fix integer overflow [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:ecd30280f5ef: avutil/lzo: add asserts to be double sure against overflows [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:c75a915f66f2: avformat/mux: Check for and remove invalid packet durations [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:26d725344155: avformat/h263dec: Fix h263 probe [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:c7e9626b7b59: swresample: fix AV_CH_LAYOUT_STEREO_DOWNMIX input [18:53] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:3c385ea24cb6: ffmpeg_filter: fix pointer to local outside scope [18:54] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:0607c8af2bd1: avcodec/mjpegdec: Fix undefined shift [18:54] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:4ecd7182851f: avfilter/graphdump: Fix pointer to local outside scope [18:54] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:4b638ea42d31: avcodec/diracdec: move mc buffer allocation to per frame [18:54] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:2b8490b762d9: avutil/cpu: force mmx on selection of higher x86 SIMD features [18:54] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:5691d9a6c129: avcodec/libvorbisenc: dont add the duration to AV_NOPTS_VALUE [18:54] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:b2b4df8d51b3: avcodec/aac: fix () in IS_CODEBOOK_UNSIGNED macro [18:54] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:ea7fbd0b1699: avcodec/golomb-test: fix () in EXTEND() macro [18:54] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:8d1f38c4b1eb: avcodec/mlpdec: fix () in MSB_MASK() macro [18:54] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:5626e11bd553: avcodec/mss34dsp: fix () in SOP* macros [18:54] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:d13cf32099fa: avcodec/mss4: Fix () in MKVAL() macro [18:54] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:f2aaee49b241: avformat/flvenc: Do not allow creating h263/mpeg4 in flv without unofficial format extensions being enabled. [18:54] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:9d77a6293627: avcodec/alsdec: Clear MPEG4AudioConfig so that no use of uninitialized memory is possible [18:54] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:c2b72eac6cfe: avformat/mpc: attempt to allocate a packet that is not smaller than the data inside it [18:54] <cone-622> ffmpeg.git 03Michael Niedermayer 07release/1.2:cc8d20e345e2: Update for FFmpeg 1.2.7 [19:15] <cone-622> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n1.2.7': unknown revision or path not in the working tree. [19:15] <cone-622> Use '--' to separate paths from revisions [19:15] <cone-622> refs/tags/n1.2.7:HEAD: Merge commit 'ccda51b14c0fcae2fad73a24872dce75a7964996' [20:48] <cone-622> ffmpeg.git 03Janne Grunau 07master:f23d26a68641: h264: avoid using uninitialized memory in NEON chroma mc [20:48] <cone-622> ffmpeg.git 03Michael Niedermayer 07master:32cf26cc6a0d: Merge commit 'f23d26a6864128001b03876b0b92fffe131f2060' [21:39] <cone-622> ffmpeg.git 03Diego Biurrun 07master:fab9df63a315: dsputil: Split off global motion compensation bits into a separate context [21:39] <cone-622> ffmpeg.git 03Michael Niedermayer 07master:d7463c681363: Merge commit 'fab9df63a3156ffe1f9490aafaea41e03ef60ddf' [22:20] <J_Darnley> Does C++ automatically initialize variables to 0? [22:20] <nevcairiel> no [22:21] <nevcairiel> unless static [22:21] <nevcairiel> like C [22:21] <J_Darnley> Then I wonder what is supposed to be in this float matrix[16] [22:21] <nevcairiel> maybe its initialized at some point later? [22:25] <J_Darnley> Oh yes, my bad [22:25] <J_Darnley> a function call sets values in it. [22:25] Action: J_Darnley just wrote that! [22:28] <cone-622> ffmpeg.git 03James Almer 07master:476bd3c7e41f: x86/dsputil: move put_signed_pixels_clamped out of bswapdsp.asm [23:05] <jamrial> i don't understand that bug. is it a crappy compiler problem or what? [23:05] <nevcairiel> no idea [23:05] <nevcairiel> its too much CE in there to read it all [23:05] <jamrial> the 3dnow functions are not going to be used unless the cpu supports them. it sholdn't matter if the binary has them [23:06] <nevcairiel> thats what I said in the very first comments in that bug [23:06] <nevcairiel> but it keeps going and going and going [23:06] <jamrial> carl's latest patch in that bug is ugly and unnecesary [23:13] <iive> i'm more curious what sets these flags? [23:13] <nevcairiel> configure [23:14] <nevcairiel> apparently he just wants to make --disable-3dnow "work", as in exclude all 3dnow code [23:14] <nevcairiel> why, is beyond us [23:18] <iive> if I understand correctly, these flags are in the elf header [23:19] <iive> the compiler or the linker put them there. my question is... how do they know. [23:19] <nevcairiel> oh, those flags [23:19] <nevcairiel> probably the linker, and it scans the objects for instructions [23:19] <nevcairiel> which seems braindead [23:19] <nevcairiel> never heard of runtime selection [23:20] <iive> so, the linker does disassembly? [23:20] <nevcairiel> well either it sets them just willy-nilly, or it actually finds these instructions in the binary somewhere [23:21] <iive> also, he complains about .a file, that is just archive with .o files ... [23:22] <nevcairiel> so the assembler would then set it on the .o file it generates? [23:22] <nevcairiel> i guess its in a position to know, but still [23:22] <iive> this looks more plausable. [23:44] <iive> https://web.archive.org/web/20070210141458/http://blogs.sun.com/alblog/entr… [23:45] <iive> the above might be useful. [23:45] <nevcairiel> solaris is all kinds of crap, apparently [23:47] <iive> well, if `fbe` is the assembler, then adding "-nH" option should clear the hardware flags. [23:48] <nevcairiel> i figured we would be using some form of yasm or nasm as well on solaris, but what do i know [23:48] <iive> there is a lot of inline assembly especially for old code. [23:49] <iive> would you kindly put the above url in the track, so maybe more people could see it. [23:49] <nevcairiel> hm, if it uses this fbe for some task, might as well try to figure out how to pass the flags [23:50] <iive> the other solution is using mapfile [23:51] <cone-622> ffmpeg.git 03Michael Niedermayer 07master:49bf712a8929: libavcodec/wmalosslessdec: add missing emms [23:54] <iive> https://web.archive.org/web/20081204050526/http://developers.sun.com/solari… [23:54] <iive> that's almost the same info. [00:00] --- Tue Jun 24 2014

1 0

ffmpeg.log.20140623
by burek 24 Jun '14

24 Jun '14

[00:00] <iive> well, if it is just few lines , feel free to ask it right away. [00:10] <elec29a> should i use cygdrive/d/ or d:/ for prefix cross prefix ? [00:10] <elec29a> i guess there is the error [00:12] <elec29a> sorry it try both [00:20] <elec29a> noone here in teamview ? i couldnt pastebin all the possibilties [00:38] <elec29a> https://www.youtube.com/watch?v=MlSkiHgA51A just have a happy day (wrong channel maybe) [04:08] <cjoseph> I just created a ffserver setup that livestreams a webcam. The target client is the web browser. The latency even just on localhost is quite bad, though, ~3s [04:08] <cjoseph> I saw the StreamingGuide post on latency, but a lot of it seems to target clients [04:09] <cjoseph> I'm not sure how I could modify the cache configurations of a web client [04:09] <cjoseph> this is the server conf: http://tny.cz/3b2a4924 [04:09] <cjoseph> if you have any suggestions on config changes to reduce latency, or advice on how to specify the cache for a browser client, please let me know [04:49] <BtbN> cjoseph, i wouldn't recommend using ffserver. It's unmaintained since a long time, and likely to be removed from future ffmpeg versions. [05:19] <ac_slater> hey all, I have an MPEG2 video. I want to know what ffmpeg demuxer to use for it. Any clues? [05:19] <ac_slater> oh nevermind I found it [05:58] <ac_slater> hmm I guess not. What muxers/demuxers are associated with mpeg2 video ? [07:52] <shape> i'm trying to convert mkv to mp4 and I get this message after running ffmpeg -i 720.mkv -vcodec libx264 -acodec libvo_aacenc test.mp4 ...i get "Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height" [08:05] <diesel420> ffmpeg -i 720.mkv -vcodec copy -acodec copy output.mp4 [08:05] <shape> diesel420: if i copy it, my tv won't play it for some reason [08:06] <shape> diesel420: I only do this because my smartTV only wants mp4 and AAC [08:11] <shape> diesel420: The error is actually from the audio Stream #0:1 -> #0:1 (dca -> libvo_aacenc) [08:14] <shape> diesel420: it seems to be working if i put aac instead of libco_aacenc [08:14] <shape> libvo_aacenc* [08:39] <volmatrix> hello everyone, is it possible to encrypt a multicast udp live stream (with ffmpeg of course) ? [09:09] <decembersoul> Hello, i have a question about the x264. If i chose "-crf 24 -preset fast" or "-crf 24 -preset veryfast" will the quality be the same (account of crf 24) and the result will be differ only in filesize? Or will the quality with -preset veryfast be more poor the with -preset fast? [09:17] <relaxed> decembersoul: presets control compression [09:19] <decembersoul> so the compression will not have any effect for the quality? I don't case much about filesize. [10:18] <horus_> do anyone have compiled ffmpeg for ios(armv7,armv7s,i386) without mp3 decoding bug, that add distortion to decoded sound? [11:28] <iamtakingiteasy> hello, i have written a simple aalib video filter for ffmpeg. here is patch: http://paste.eientei.org/show/220/ and example of ascii-fied video: http://www.youtube.com/watch?v=Xd9dFuwVmD8 [11:29] <iamtakingiteasy> any hints how to proceed, what needs to be changed in order to upstream this patch? [11:32] <brontosaurusrex> iamtakingiteasy, afaik just post that to devel mail-list [11:34] <c_14> iamtakingiteasy: https://ffmpeg.org/developer.html#Submitting-patches-1 [11:40] <iamtakingiteasy> thanks [16:22] <termos> can I use ffmpeg filter graph to detect interlacing and if it's interlaced I run the yadif filter? [16:23] <termos> Without having to run two separate processes, one to detect and another to deinterlace [16:34] <c_14> I'm not sure that's possible. But you can try using yadif with deint=1 so it'll only deinterlace frames marked as interlaced. [16:41] <foonix> when encoding with cli, is there a way to set buffer size for output file or somehow force to flush in atleast 4MB chunks? [16:41] <c_14> -bufsize ? [16:42] <termos> okey I'll give it a try thanks! I will try to run it on some none-interlaced video and see what happens to the video and the cpu usage. [16:43] <c_14> You should still require a good amount of cpu, since in order to use a filter you'll need to reencode the video track. [16:46] <termos> that's, the filter is only running on the AVFrames that I get after decoding [16:47] <termos> that's true, ... [16:51] <foonix> c_14: i need that for writing the file to storage not for stream [16:55] <foonix> but will look more on AVFormatContext AVOptions, few things looks promising [17:09] <apoc> I just can't get this to work, I'm trying to record via x11grab/pulseaudio but the audio is always async (seems like it gets async over time): http://pastebin.com/hyhcYX3f [17:10] <apoc> the start timecodes look strange to me, (start: 0.004973 for audio, start: 1403535582.786433) isn't any audio syncronization effort by ffmpeg based on this? [17:11] <apoc> can I get the pulse audio input to use a unix timestamp as a timecode too? [17:12] <foonix> chunk_size,packetsize,bufsize doesnt help me, just to mention i write to NFSv4 share [17:38] <apoc> looks like an old known/ignored bug [17:38] <apoc> @ my x11grab async issue [19:20] <SrikanthJessu> Hi Team, [19:20] <SrikanthJessu> Unable to resolve this error... [19:20] <SrikanthJessu> >[libx264 @ 0x1639e30] broken ffmpeg default settings detected [19:20] <SrikanthJessu> >[libx264 @ 0x1639e30] use an encoding preset (e.g. -vpre medium) [19:20] <SrikanthJessu> >[libx264 @ 0x1639e30] preset usage: -vpre <speed> -vpre <profile> [19:20] <SrikanthJessu> >[libx264 @ 0x1639e30] speed presets are listed in x264 --help [19:20] <SrikanthJessu> >[libx264 @ 0x1639e30] profile is optional; x264 defaults to high [19:20] <SrikanthJessu> >Cannot open video encoder for stream #0 [19:20] <SrikanthJessu> >Error occurred: Operation not permitted [19:20] <SrikanthJessu> how to set -vpre option programmetically? [19:20] <SrikanthJessu> options tried.. [19:20] <SrikanthJessu> //libx264-ipod320.ffpreset [19:20] <SrikanthJessu> av_opt_set(enc_ctx->priv_data, "vprofile", "baseline", 0); [19:20] <SrikanthJessu> av_opt_set(enc_ctx->priv_data, "level", "30", 0); [19:20] <c_14> pastebin, please [19:20] <SrikanthJessu> Sorry, okay. [19:21] <sacarasc> SrikanthJessu: -vpre was removed a LONG time ago. [19:21] <sacarasc> Your version is ancient. [19:21] <sacarasc> Upgrade. [19:21] <SrikanthJessu> http://pastebin.com/mgiGi0mM [19:22] <SrikanthJessu> @sacarasc which one ffmpeg or x264? [19:22] <sacarasc> ffmpeg. [19:22] <SrikanthJessu> I compiled 2.2.3 [19:23] <sacarasc> The output you gave was not from 2.2.3... [19:31] <SrikanthJessu> @sacarasc I updated with my ffmpeg configuration (at bottom) http://pastebin.com/WnXGLpWC on 2.2.3. did I miss anything? [19:32] <sacarasc> If you don't need any of the other codecs, probably not. [19:32] <SrikanthJessu> I need h264 and mp3 encoders. [19:32] <SrikanthJessu> due to that I compiled x264 and mp3lame [19:33] <SrikanthJessu> Used android cross compilation. [19:36] <SrikanthJessu> @sacarasc added x264 configuration information http://pastebin.com/nLJqapu3 (at bottom) [19:45] <SrikanthJessu> what could be the possible reasons for "broken ffmpeg default settings detected"? [19:46] <sacarasc> An ancient ffmpeg. [19:46] <sacarasc> The defaults work now. [19:47] <SrikanthJessu> Okay @sacarasc, I will try recompiling ffmpeg once again. Thanks. [19:48] <sacarasc> You could look at the output of just `ffmpeg` and it will tell you the version number. [19:51] <SrikanthJessu> @sacarasc I used '--disable-programs' option in configure, so there is no `ffmpeg` executable available. I think so.. [23:36] <ogrgkyle> Hi! http://pastebin.com/uJksFVfS [23:36] <ogrgkyle> I have this OpenCV program which takes an image from the command line and outputs the detected faces' X,Y,width,height. [23:37] <ogrgkyle> I'm trying to modify it so that it will anaylze every frame of a video [23:37] <ogrgkyle> And I don't know the fastest, most efficient way to do that. Right now, I have to extract all the frames of a video and put them in a folder, and then analyze those one by one through a .bat file. [23:38] <ogrgkyle> Also, I read that finding an exact frame is slow. [23:39] <ogrgkyle> So I wonder if you can help me with this. Is there a way to feed the frames of a video through OpenCV in my simple script? Thanks. [23:42] <c_14> Maybe try looking at: https://code.dyne.org/frei0r/tree/src/filter/facedetect ? [23:42] <c_14> Not entirely sure how this is FFmpeg related though since you're not using the library. [23:43] <ogrgkyle> Yeah [23:43] <ogrgkyle> Well, I think I'll need to learn how to use the library [23:45] <ogrgkyle> So that code (in the link) is designed for video? [23:48] <c_14> ye, if you compile ffmpeg with --enable-frei0r (iirc) you can use them directly with libav* as a video filter. [23:50] <ogrgkyle> You mean if I enable frei0r, FFmpeg itself can do face detection? [23:51] <c_14> Assuming the filter does what its name implies, yes. [23:51] <c_14> I've never actually used it, YMMV. [23:51] <ogrgkyle> very interseting [23:51] <ogrgkyle> interesting [23:52] <ogrgkyle> So, if you've never used it, how did you find it? :) [23:52] <ogrgkyle> I mean, is this the only one... or are there others? [23:52] <c_14> I was browsing the FFmpeg configure options one lonely afternoon when I came across the --enable-frei0r option and decided to investigate. [23:53] <c_14> There are probably others, but I don't know how well they interface with FFmpeg or where you could find them. [23:53] <ogrgkyle> Very interesting indeed [23:56] <ogrgkyle> If I build ffmpeg with frei0r, does that include the opencv auxiliary library? http://www.dyne.org/software/frei0r/ [23:56] <ogrgkyle> That is, if I download the windows binary (with frei0r), does that include the opencv stuff? [23:58] <c_14> Do you mean the zeranoe binary? If so, I think it does if you get the dev version, but don't quote me on that. [00:00] --- Tue Jun 24 2014

1 0

ffmpeg-devel.log.20140622
by burek 23 Jun '14

23 Jun '14

[00:02] Action: wm4 hopes that libavscale doesn't stay vaporware [00:03] <J_Darnley> BBB: http://pastebin.com/CLmhJefE [00:10] <ubitux> wm4: it will probably have similar bugs, some more, it will be slower, and probably just a wrapper around sws at the beginning [00:10] <wm4> that's some optimism [00:11] <ubitux> it won't happen anytime soon anyway [00:11] <ubitux> unless they use the money of ffmtech [00:12] <ubitux> the avframe wrapper might happen soon or later though, but i believe it will stay in that state for quite a long time [00:12] <wm4> I have a patch to add that to swscale [00:12] <wm4> I posted it a year ago or so [00:18] <michaelni> wm4, whats the subject of the patch/mail ? [00:20] <wm4> [FFmpeg-devel] [PATCH] swscale: add API to convert AVFrames directly [00:37] <michaelni> wm4, the last thing in that thread is a review from stefano without a reply from you [00:43] <wm4> yes I think I decided not to add this, because libavscale was supposedly in the works [00:46] <wm4> and it was also kind of hacky how the reinitialization worked (sledge-hammer approach to make libswscale do what I want) [00:47] <michaelni> what problem was there with reinit ? [00:49] <wm4> I don't quite remember, but I had to uninit and reinit it in-place [00:50] <michaelni> I suggest we work on swscale and improve what needs to be improved [00:52] <wm4> well, as it stands even adding simple conversions seems to be hard and non-trivial [00:52] Action: wm4 remembers Daemon404 cursing for weeks when he wanted to add GBRP conversions [00:56] <michaelni> so improve its structure [00:57] <wm4> nobody really wants to refactor libswscale mess [00:57] <michaelni> is it easier when its called avscale = [00:57] <michaelni> ? [00:58] <wm4> libavscale supposedly starts from fresh, instead of forking libswscale or anything [00:58] <michaelni> also i do want to refactor it but i have too many things to do so its unlikely i will do it alone [01:31] <michaelni> ubitux, the valgrind hqx issue looks like a valgrind "bug" SIMD having some input bytes uninitialized and that infecting too many output bytes [02:06] <iive> are libav really working on avscale? [02:07] <wm4> "working" [02:12] <iive> wm4: ? [02:13] <wm4> they've been talking about it for a year or so [02:14] <iive> so just talk. [02:18] <wm4> well, let's call it plans [02:19] <nevcairiel> they dont have the man power to finish it any time soon anyway, as it is only like one guy is "working"/planning on it, isn't it [02:25] <iive> and that guy should maintain the existing libav :) [02:29] <wm4> nevcairiel: as long as there's literally nothing, nobody can help contributing anyway [02:30] <nevcairiel> he keeps talking about their weird "sprint" somewhere on a mountain when he wants to create something [02:33] <nevcairiel> it should be in a month or so, maybe we'll get something to bikeshed to death after \o/ [02:42] <wm4> lol [02:42] <j-b> good morning [02:43] <wm4> technically it's almost morning indeed [02:43] <nevcairiel> its nearly 8pm, whatchatalkingabout [02:49] <j-b> it's 3am almost [03:16] <cone-539> ffmpeg.git 03Michael Niedermayer 07master:a1a76b209b42: swscale/x86/input: prevent RGB32_TO_UV_FN from reading into the padding [03:16] <cone-539> ffmpeg.git 03Michael Niedermayer 07master:4abffbbc54a2: swscale/x86/input: prevent RGB32_TO_Y_FN from reading into the padding [03:16] <michaelni> ubitux, wm4 valgrind issue should be fixed [03:17] <michaelni> wm4, still not interrested in working together on improving swscale ? [03:18] <michaelni> i really think it would be better if everyone would work together [03:19] <wm4> I find the idea of starting fresh attractive, instead of forever-refactoring swscale, but I have no idea what will come of it [03:21] <michaelni> well, if you have an idea on how to "segment" the scaling which is better than how it is currently then you can change swscale toward this in small steps [03:21] <michaelni> i dont think it would be less work if you start from scratch [03:22] <michaelni> and OTOH if you have no idea on how to improve things, well .. [03:23] <iive> Daemon404 likes to complain about things he doesn't understand, blaming the author rather himself. [03:23] <wm4> if you look at one of the swscale sources (I think it was utils.c), you can see there's a BIG function which initializes EVERYTHING, and which even has self-modifying code, complete with platform specific code for unix and windows [03:23] <wm4> and it's for MMX [03:23] <wm4> and this last bit always cracks me up [03:23] <iive> and rewriting is good way to figure out how something works. [03:25] <wm4> well there's fmtconv (https://github.com/vapoursynth/fmtconv) which in some circles was hailed as possible libswscale replacement, but I'm not sure about its performance [03:25] <wm4> (performance as in speed and quality) [03:25] <wm4> it's written in C++ though [03:25] <iive> i'd like to see templates in C. [03:25] <wm4> I have no idea if it has better design [03:26] <michaelni> wm4, the self modifying code is just for the "fast biliear scaler" its quite seperate [03:26] <wm4> quite separate? but it's messed right into the shit [03:26] <wm4> technically it's probably not the worst part, sure [03:27] <iive> i'm kind of disappointed, I thought the SMC is used for everything. [03:28] <michaelni> wm4 you can comment it out and not use the FAST_BILINEAR scaler. it doesnt really affect any overall design [03:28] <wm4> iive: no it's just for MMX and a shitty scaler [03:28] <wm4> (apparently) [03:28] <wm4> michaelni: yes sure, but I don't know how you expect anyone not to run away with code like this [03:30] <iive> why do you call it shitty? [03:33] <michaelni> wm4, i cant really do anything about "dislike for the existing code" except to suggest to improve it [03:34] <wm4> can I send a patch to remove this code? [03:35] <michaelni> send? yes, but what advantage and disadvantage does it have to remove it ? [03:35] <iive> wm4: after you give benchmark proving that the remaining code is as fast as SMC and produces same output. [03:35] <michaelni> has it lost its speed advantage with sse & avx ? [03:36] <iive> michaelni: maybe you should add SSE and avx to it. and move it to separate file. [03:36] <iive> so people are not scared :O [03:37] <michaelni> a seperate file is certainly better than having it where it is now, yes [03:38] <wm4> <michaelni> a seperate file is certainly better than having it where it is now, yes <- you're using the word "better", so at least you agree there's some sort of problem [03:39] <michaelni> yes [03:39] <iive> ok. what are the other problems of swscale? [03:40] <wm4> iive: that apparently most conversions must go through yuv420 [03:40] <wm4> and general slowness and lack of quality [03:40] <wm4> (image quality) [03:41] <iive> you mean, rgb->rgb goes through yuv? [03:41] <michaelni> yuv420 isnt really true, theres a >8bpp internal planar YUV buffer [03:42] <michaelni> so it would be YUV 444 internally when its needed / the user wants [03:43] <michaelni> > 8 bit per sample that is not pixel ;) [03:48] <nevcairiel> You need to explicitly request it to preserve full chroma, which seems kinda unintuitive [03:49] <michaelni> yes and yes (unintuitive) [03:50] <iive> doesn't it use the least common denominator? [03:51] <nevcairiel> Its not that smart [03:51] <michaelni> making it more intuitive and use better defaults should not be very hard [03:53] <iive> maybe add two options, one to prefer speed over quality and another for the opposite. [03:53] <iive> why the most interesting chat happens when i have to leave... [03:53] <iive> n8 ppl [03:53] <michaelni> n8 iive [05:16] <cone-539> ffmpeg.git 03Michael Niedermayer 07master:d7efafd63a63: avfilter/vf_hqx: partly fix big endian fate test [05:16] <cone-539> ffmpeg.git 03Michael Niedermayer 07master:fd3c27375f96: fate/filter-video: fix hqx on big endian part 2 [05:35] <jamrial> seems that the hqx tests fail with threads > 2 [05:40] <jamrial> "make fate-filter-hqx THREADS=4" failure. "make fate-filter-hqx THREADS=4 THREAD_TYPE=slice" success [06:05] <cone-539> ffmpeg.git 03Michael Niedermayer 07master:954a38e9bf65: avfilter/vf_hqx: remove << 0 [06:05] <cone-539> ffmpeg.git 03Ronald S. Bultje 07master:0dae193d3ecf: swr: remove another forgotten division in DSP function. [09:18] <ubitux> oh thx michaelni [09:26] <ubitux> that threading issue seems related to the dimension changes again [09:33] <ubitux> it's not calculating one of the frame, for some reason [18:14] <cone-826> ffmpeg.git 03Diego Biurrun 07master:9a9e2f1c8aa4: dsputil: Split audio operations off into a separate context [18:14] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:99497b4683e5: Merge commit '9a9e2f1c8aa4539a261625145e5c1f46a8106ac2' [18:31] <wm4> how long until vf_drawtext becomes photoshop [18:35] <michaelni> thats easy, it will never, and actually i think thats sad .... a free software replacement for photoshop would be nice (gimp is nice yes but its user interface is not nearly as intuitive but then its long ago that i used either) [18:53] <J_Darnley> I think 2.8's one-window interface made it much better, on Windows at least. [18:54] <J_Darnley> ... and for what I do with it. [18:55] <wm4> J_Darnley: except they did stupid things with the close button [18:55] <wm4> now clicking the close button once closes a single document, not gimp (lolwut) [18:56] <J_Darnley> Oh [18:56] <wm4> I retried it, apparently they fixed it :D [18:57] <J_Darnley> I must have missed that "feature" [18:57] <J_Darnley> :) [19:31] <nevcairiel> that was probably a remnant from the multi-window system [19:32] <nevcairiel> since that was what happened there as well, and made somewhat sense then [20:11] <cone-826> ffmpeg.git 03Andrey Utkin 07master:3bb4d26a5df4: drawtext: drop unused draw_glyphs() arg "rgbcolor" [20:11] <cone-826> ffmpeg.git 03James Almer 07master:6ec3dc97fcd2: x86/audiodsp: move asm code out of dsputil [20:29] <debianuser> Hello, I'm trying to concat different (wav, ogg) audio files. But command `echo -e "file 1.wav\nfile 2.ogg\nfile 3.wav" | ffmpeg -f concat -i - -ac 2 -ar 48000 0.wav` gives incorrect output file and lots of errors: pastebin.com/KQQSCGM4 [20:29] <debianuser> Is it a bug? Or maybe I miss something obvious? [20:33] <J_Darnley> Apart from the fact that this is the wrong channel? [20:38] <J_Darnley> Well to start with you cannot cat those file together. [20:39] <wm4> hm, ffmpeg is still not in debian? [20:42] <debianuser> wm4: It's in deb-multimedia, but mine was built from sources. [20:43] <debianuser> J_Darnley: Sorry, I thought if that's a bug it must be reported to this channel. [20:44] <wm4> deb-multimedia is not official [20:45] <debianuser> J_Darnley: You mean it's impossible to concat audio files with ffmpeg at all, or am I using wrong command? [20:45] <wm4> debianuser: it's not a bug, I think the problem is that you can't mix codecs [20:50] <debianuser> Why? I mean it's not remuxing, it's recoding. And ffmpeg can decode any codec... Anyway, if I convert ogg->wav first and then try to concat 3 wav files I still get it wrong (47.35 seconds file while sources were 10.92+3.43+12.89 seconds). [20:51] <J_Darnley> What you are asking ffmpeg to is the same as if you had cat the files directly. [20:52] <J_Darnley> If you want to cat random files you will need to use the concat filter [20:54] <debianuser> Is it "almost" the same or "literally" the same? [20:56] <debianuser> No, it's not "literally" same. I just checked. When I `cat` wav files and try to recode result, only the first of 3 files is recoded. [21:02] <debianuser> So if it's not a concat-bug, it's some missing feature of -f concat. [21:02] <debianuser> J_Darnley: concat filter is harder for me to use, because I need this working for hundreds of files. For -f concat I could do `ls -1 dir/* | sed 's/^/file /' | ffmpeg -f concat -i - ...`. But it's much harder to construct a command line for concat filter. Anyway, I tried it too: `ffmpeg -i 1.wav -i 2.ogg -i 3.wav -filter_complex 'concat=n=3:v=0:a=1[a]' -map '[a]' -ac 2 -ar 48000 0.wav` gives a short file (10.92+3.43+12.89!=23.83) and many errors: past [21:06] <wm4> there's no such thing as switching codec mid-stream (on the same ffmpeg-level stream), so that can't be fixed AFAIK [21:06] <wm4> it needs to be implemented higher level [21:06] <wm4> but good luck touching ffmpeg.c [21:13] <nevcairiel> wasnt there some concept that allowed that somehow [21:16] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:31f77b46b2fa: avfilter/unsharp_opencl: fix macro () [21:16] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:9b33cdcab211: avfilter/vf_blend: fix macro () [21:16] <cone-826> ffmpeg.git 03Anshul Maheshwari 07master:ca2f59e12108: avcodec/dvbsubdec: support returning exact end times [21:16] <cone-826> ffmpeg.git 03Anshul Maheshwari 07master:7e6cf364537e: ffmpeg: fix transcoding dvbsub to dvbsub [21:29] <debianuser> But there're files/streams changing codec on the fly. And mkv chapters loading content from another file. Can't ffmpeg support any of them yet? [21:30] <nevcairiel> nope [21:30] <nevcairiel> and those mkv things kidna require to be all the same codec [21:32] <debianuser> Well, then, it's a feature request: support resetting/reinitializing codec mid-stream to support different codecs for -f concat. [21:33] <JEEB> that'd pretty much be what Libav wanted to do with concateration then, which would be fully on the side of the app, instead of putting it into demuxers etc [21:34] <wm4> isn't concatenation possible with lavfi? [21:35] <nevcairiel> cant you pipe several files in order, decoding to the same raw format first [21:35] <nevcairiel> doing this in the demuxer/decoder chain seems unlikely [21:35] <nevcairiel> more like an application feature if anything [21:36] <debianuser> That would be part of demuxer, i.e. only demuxer knows when stream of the first codec ended, and there goes another codec, isn't it? And -f concat looks like a demuxer for me, optionally calling other demuxers if needed... [21:37] <nevcairiel> just saying that its highly doubtful to ever be implemented in a demuxer [21:38] <nevcairiel> because in practice, it can't. the app controls the interaction between demuxing and decoding, which obviously alos needs to be updated when the format changes [21:38] <nevcairiel> so it would be much easier to just open the new file on the side of the app [21:39] <nevcairiel> in any case, best solution, just decode them all to the same raw format first [21:45] <debianuser> Hm... Does the app initialize the decoder manually? Or is there a way for demuxer signal to the decoder somehow to reinitialize itself to some new settings? [21:46] <nevcairiel> to new settings, sure, but not to an entire new decoder [21:46] <nevcairiel> its not designed for this, and the design is unlikely to change [21:48] <debianuser> Then concatting 3 wav files should work at least. But it does not. :( It gives me 47.35 seconds file while sources were 10.92+3.43+12.89 seconds. [21:49] <nevcairiel> same channels and sample rate? [21:50] <debianuser> No, different channels/rates, but same codec: wav [21:50] <debianuser> (pcm s16_le) [21:50] <nevcairiel> then thats your reason [21:51] <debianuser> So it can't even reinitialize the codec to new values? [21:52] <nevcairiel> i guess it could in theory do that, but in practice noone cares [21:59] <debianuser> If I have to manually decode them to the same format, then... Why ffmpeg (as application) can't do it automatically? I mean if I do the recoding anyway why can't ffmpeg decode them all to the same format before concatting? [22:00] <nevcairiel> like i said, because noone cares [22:00] <nevcairiel> send a patch [22:02] <wm4> AFAIK it's because ffmpeg.c is a big mess, and nobody wants to implement it the proper way [22:02] <wm4> so there are hacks like the concat demuxer [22:02] <wm4> which work "sometimes" [22:07] <debianuser> Well, I guess that would remain a feature request then. [22:09] <debianuser> How about -filter_comples concat, is it supposed to work for different codecs? This page says it does: http://trac.ffmpeg.org/wiki/How%20to%20concatenate%20%28join,%20merge%29%20… [22:09] <debianuser> And if it does, why command `ffmpeg -i 1.wav -i 2.ogg -i 3.wav -filter_complex 'concat=n=3:v=0:a=1[a]' -map '[a]' -ac 2 -ar 48000 0.wav` gives a short file (10.92+3.43+12.89!=23.83) and many errors: pastebin.com/nEYL9RLa [22:09] <debianuser> Is that some missing feature too? [22:10] <nevcairiel> you can probably fix that by mixing and resampling the audio before the concat filter [22:10] <nevcairiel> so the concat filter gets all audio in the same format [22:26] <debianuser> Hm... Like this? ([0:0]aresample=48000:och=2:osf=s16:first_pts=0[a0],[1:0]aresample=48000:och=2:osf=s16:first_pts=0[a1],[2:0]aresample=48000:och=2:osf=s16:first_pts=0[a2],[a0][a1][a2]concat=n=3:v=0:a=1[a]) http://pastebin.com/nA5JhXXZ Still 10.92+3.43+12.89 != 23.89 seconds [23:36] <Timothy_Gu> debianuser: it seems to me that ffmpeg is skipping the ogg track [23:37] <Timothy_Gu> 10.92+12.89 almost = 23.89 [23:47] <debianuser> Timothy_Gu: It's not, not completely. I can hear part of the track in the result file. (in case my ffmpeg build is broken) you can try it, these files are here: ge.tt/1Eg60nl1/v/1 [00:00] --- Mon Jun 23 2014

1 0

ffmpeg.log.20140622
by burek 23 Jun '14

23 Jun '14

[00:53] <RadSurfer> there is a way to silence an audio stream, (make volume 0, or audio track null?) how would I do that please? [00:54] <c_14> If you want to silence the entire audio stream, couldn't you just remove it? ie remux without it? If you want an audio stream, but want it to be silent, use the aevalsrc filter. [00:55] <RadSurfer> Some players require an audio stream be there; so I just want to silence it [00:56] <c_14> use -an and -af aevalsrc=0 [00:57] <c_14> hmm, wait. [00:57] <RadSurfer> ? [00:57] <c_14> looks like -an kills the implicit mapping in the -af filter [00:57] <RadSurfer> meanwhile, I'll edit the video that needs silencing... [00:59] <c_14> Right, -filter_complex 'aevalsrc=0[a]' -map '[a]' should work. [00:59] <c_14> You'll have to explicitly map the video/subtitle streams you want though. [00:59] <c_14> use something like -map 0:v if you want all the video streams [01:00] <RadSurfer> I want video UNtouched; just silence it [01:00] <c_14> add -c:v copy [01:00] <c_14> That'll copy the video stream. [01:06] <c_14> Oh, you'll want to add -shortest. [01:06] <c_14> Or else it'll just keep going. [01:06] <RadSurfer> what does that mean? [01:07] <c_14> aevasrc creates a source file with null audio of infinite length. So if you try muxing that with your video it'll go on forever. -shortest says that it should only create a source with the length of the shortest input stream. [01:09] <RadSurfer> you syntax is rejected by ffmpeg [01:11] <RadSurfer> http://pastie.org/9312691 [01:13] <c_14> Pretty sure you can't have a -vf and a filter_complex at the same time. use -filter complex 'aevalsrc=0[a];[0:v:0]yadif[v]' -map '[a]' -map '[v]' -c:v libx264 <file>.mkv [01:13] <RadSurfer> could have included error message... [01:18] <RadSurfer> what it is you are trying to do: ffmpeg REJECTS syntax [01:19] <RadSurfer> I'll try to paste whole thing this time, w/error [01:22] <RadSurfer> http://pastie.org/9312715 [01:22] <RadSurfer> Meanwhile, I'm stuck. [01:22] <RadSurfer> I can't post this with audio it has in it. :-) [01:23] <c_14> Ok, in that specific case I accidentally wrote -filter complex instead of -filter_complex [01:24] <RadSurfer> I didn't catch it: still learning syntax myself [01:24] <c_14> Which is why it thinks that everything in the quotes is an output file. [01:25] <RadSurfer> your syntax FAILS: 3 times now [01:26] <RadSurfer> http://pastie.org/9312722 [01:28] <c_14> The command I gave you works just fine on my machine. [01:29] <c_14> `ffmpeg -i in.mkv -aspect "16:9" -filter_complex 'aevalsrc=0[a];[0:v:0]yadif[v]' -map '[a]' -map '[v]' -c:v libx264 -shortest out.mkv' [01:31] <RadSurfer> why do I get errors, 3 [01:31] <RadSurfer> one last time, I'll just have to think of something else, or use avidemux later [01:32] <RadSurfer> SAME ERROR [01:32] <RadSurfer> [a] invalid [01:33] <RadSurfer> try that with an MP4 still work for you? [01:35] <c_14> yep, switched to mp4 as input and that worked, also worked with mp4 as both input and output [01:35] <RadSurfer> then why should it FAIL here [01:35] <c_14> Are you using a recent version of ffmpeg? [01:35] <RadSurfer> it was just downloaded 3 days ago! [01:37] <c_14> You're using the windows terminal, right? [01:37] <c_14> Are [] considered special characters? [01:37] <RadSurfer> Win8.1 [01:37] <RadSurfer> cmd.exe [01:37] <c_14> if not you could try removing the single quotes around [a] and [v] [01:37] <c_14> and maybe around the aevalsrc part [01:38] <c_14> Those quotes are only really there for shell level escaping, it might be that windows is mucking about with that [01:38] <c_14> start by removing the single quotes around [a] and see if the error message changes [01:38] <RadSurfer> anyone here willing to try that syntax in a winblows box? [01:41] <debianuser> Hello. Trying to join multiple audio (wav, ogg) files into a single ogg. Ended up with: ls -1 dir/* | sed 's/^/file /' | ffmpeg -f concat -re -i - -acodec vorbis -ac 2 -ar 48000 -strict experimental 0.ogg But I get a lot of "invalid dropping" messages (http://pastebin.com/Bmqp0zDp) Do I miss some important params? [01:42] <RadSurfer> crying out loud... [01:42] <RadSurfer> Syntax has been fixed. :-) [01:43] <RadSurfer> on a winblows box: it only wants " not ' delimiters [01:43] <c_14> I'll remember that for future reference. [01:44] <RadSurfer> Okie [01:45] <c_14> debianuser: Does the output audio sound ok/is as long as it should be? [01:45] <RadSurfer> Just hope it works now [01:51] <debianuser> c_14: It's shorter. On a small subset of 18 files the resulting 0.ogg is 00:01:04.06, while sum of initial files length is 71.46 (ls -1 dir/* | xargs -L 1 ffprobe 2>&1 | tr ':,' ' ' | awk '/Duration/{a+=$4}END{print a}') [01:53] <c_14> question, are you sure you need -re? [01:53] <debianuser> No, I don't. [01:53] <c_14> But that shouldn't be the problem. [01:53] <c_14> hmm [01:54] <debianuser> (I'm almost sure that I don't, but I tried to add it recently, just to check whether it changes anything, it does not) [01:55] <debianuser> I can share those 18 wav files (5MBytes) if it can help. [01:56] <c_14> Sure, I can test it on my system. See what happens. [01:58] <c_14> Those errors tend to hint that there is something wrong with the source files. At least with the timestamps. [01:59] <debianuser> http://ge.tt/1Eg60nl1 [02:00] <RadSurfer> this encoding has been going on a long time! [02:01] <RadSurfer> Appears that your syntax is really wrong after all. [02:01] <RadSurfer> this video is <6min long [02:01] <RadSurfer> it went WAY PAST that [02:01] <c_14> did you add -shortest? [02:02] <RadSurfer> whatever you had in there. I used your example [02:02] <RadSurfer> lets see if this even plays [02:02] <RadSurfer> its a mess [02:02] <RadSurfer> time index is all screwed up! [02:02] <RadSurfer> but at least it is silent [02:03] <RadSurfer> I just want SILENCE for crying out loud... lol [02:03] <RadSurfer> @least input file is undamaged [02:14] <debianuser> c_14: Maybe I'm using a wrong command and people usually join files differently? It's hard to get something wrong with timestamps in wav-files... [02:14] <c_14> Nah, command looks right. [02:17] <c_14> Ok, so what I just did was resample all the input files to 48kHz with a for loop and then concatted that. It still shows the DTS/PTS warnings but the output file is 01:11:49 long which corresponds to the correct length. [02:19] <c_14> Also, IIRC libvorbis is recommended over vorbis as an encoder. [02:19] <c_14> you won't need -strict experimental anyway [02:19] <RadSurfer> meanwhile: Should any one else have an idea how to SILENCE an MP4 audio track PLEASE let me know /msg welcome. [02:24] <RadSurfer> I wonder if avidemux can even mute audio tracks? [02:24] <RadSurfer> there's no obvious filter [02:25] <debianuser> RadSurfer: what about something like -af volume=0 ? [02:26] <RadSurfer> Can't hurt at this point [02:26] <RadSurfer> :-) [02:30] <RadSurfer> Here goes... [02:31] <RadSurfer> I'd say perfect! Just what I wanted; and straight-forward too. embarassingly so. [02:34] <RadSurfer> solved one problem, discovered another. But that works. [02:50] <debianuser> c_14: I just had this ffmpeg built without libvorbis. I also tried to encode into .wav, but still got those "invalid dropping" messages. So, is it a bug? Or maybe "concat" hates different codecs/rates? [02:58] <OnkelTem> Hi [02:59] <OnkelTem> I'm trying to use this guide https://wiki.ubuntu.com/ScreenCasts/ffmpeg to make a screencast, but it doesn't work [02:59] <OnkelTem> Unknown input format: 'x11grab' [02:59] <OnkelTem> This error [02:59] <OnkelTem> My command line is: ffmpeg -f x11grab -s 1280x760 -r 25 -i :0.0+3840+0 -f alsa -ac 1 -i pulse out.webm [03:04] <OnkelTem> How comes that x11grab became "unknown"? [03:04] <OnkelTem> ffmpeg version 2.1.3- http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2013 the FFmpeg developers [03:04] <OnkelTem> built on Jan 22 2014 19:50:14 with gcc 4.8 (Debian 4.8.2-14) [03:15] <klaxa> static builds don't support x11grab [03:17] <OnkelTem> wow [03:17] <OnkelTem> good news [03:17] <OnkelTem> I'm pretty sure, for a reason :) [05:15] <debianuser> Hello. I'm trying to join many (hundreds of) audio files into one, and get some errors whatever I try even for a small subset. [05:16] <debianuser> Command `echo -e "file 1.wav\nfile 2.ogg\nfile 3.wav" | ffmpeg -f concat -i - -ac 2 -ar 48000 0.wav` gives incorrect output file and lots of errors: pastebin.com/KQQSCGM4 [05:16] <debianuser> Command `ffmpeg -i 1.wav -i 2.ogg -i 3.wav -filter_complex 'concat=n=3:v=0:a=1[a]' -map '[a]' -ac 2 -ar 48000 0.wav` gives a short file (10.92+3.43+12.89!=23.83) and many errors: pastebin.com/nEYL9RLa [05:16] <debianuser> Those 3 test files: ge.tt/1Eg60nl1/v/1 What am I doing wrong? How to correctly join files without errors? [11:25] <_Crash_Laptop> Hi, just installed ffmpeg via "sudo apt-get installed ffmpeg" in Ubuntu 12.04 and when i try and run "ffmpeg" through terminal it cannot locate it "The program 'ffmpeg' is currently not installed. To run 'ffmpeg' please ask your administrator to install the package 'libav-tools'". I had removed the binary ffmpeg when i compiled it but i've ran the install three times and even removed it fully before re-installing with [11:25] <_Crash_Laptop> "remove --purge" - any ideas? [11:27] <sacarasc> Did you install the package it asked you to? [11:27] <_Crash_Laptop> yes [11:27] <sacarasc> (Also, Ubuntu doesn't have ffmpeg any more, it has libav which is a fork.) [11:28] <_Crash_Laptop> i know, but i need ffmpeg for serviio [11:28] <sacarasc> Grab one of them instead? [11:29] <sacarasc> Wait, did you compile yourself? Did you follow the Ubuntu compile guide? [11:29] <sacarasc> https://trac.ffmpeg.org/wiki/UbuntuCompilationGuide [11:29] <_Crash_Laptop> i compiled it before, but i removed it and tried via apt-get which doesn't seem to be placing the binary where it is supposed to [11:30] <sacarasc> In that case, ask in #ubuntu or #libav, because you're not installing ffmpeg, anyway. [11:31] <_Crash_Laptop> right [11:33] <_Crash_Laptop> well i am trying to install ffmpeg.. [11:33] <sacarasc> No, Ubuntu does NOT have ffmpeg any more. [11:34] <brontosaurusrex> _Crash_Laptop, either compile it or use the gusari static builds [12:13] <SimAV> Hi! I want to produce a livestream using ffmpeg/ffserver and wonder whether there is a way to specify a "fallback" feed that is used as input for the stream as long as the feed that will be live isn't feeded. [12:15] <maksimkaaa> Hi, i add text on some frames with draw option specifying the X,Y coordinates for my text, the text is printed successfully on the video but there is a blinking cursor that keeps appearing on random areas of the video..! Isn't there a way to remove that blinking cursor? [13:30] <oakwhiz> I'm trying to make a video out of an animated GIF and a FLAC file. The length of the FLAC is a multiple of the length of the GIF. How do I repeat the GIF as many times as is necessary to fill the length of the FLAC? [13:30] <oakwhiz> I'm thinking it might be -loop but I'm still reading through man pages [13:41] <iamtakingiteasy> hello, i am thinking of writing a ffmpeg video filter, which will convert incoming image into ascii-art using aalib, but i am not very sure from where to start. i've found a nice guide overviewing general aspects of writing filter for libavfilter component of ffmpeg: http://wiki.multimedia.cx/index.php?title=FFmpeg_filter_HOWTO ; but it is not covering it in great detail and missing complete [13:41] <iamtakingiteasy> example sources. is there any other guides, especially covering ffmpeg build-process integration and development-time simplifications of build routine? [13:43] <oakwhiz> iamtakingiteasy: you should look into libav/avconv, it seems to be some kind of fork of ffmpeg [13:44] <sacarasc> It has less features than ffmpeg. \o/ [13:44] <oakwhiz> Sometimes I come across files that don't work in ffmpeg but work in avconv, or vice versa [13:45] <oakwhiz> damn, -loop 1 just returns "Option loop not found." [13:45] <sacarasc> -loop_input maybe? [13:45] <sacarasc> And also -shortest [13:46] <oakwhiz> -loop_input 1 -shortest gives Unrecognized option 'loop_input'. [13:46] <sacarasc> What version are you using? [13:46] <oakwhiz> -loop 1 -shortest gives "Option shortest (finish encoding within shortest input) cannot be applied to input file [...] -- you are trying to apply an input option to an output file or vice versa. Move this option before the file it belongs to." [13:47] <oakwhiz> ffmpeg version 2.1.4 [13:47] <oakwhiz> built on Feb 25 2014 08:24:47 with gcc 4.8.2 (GCC) 20131212 (Red Hat 4.8.2-7) [13:48] <oakwhiz> You're supposed to put those arguments at the beginning before the inputs, right? e.g. -loop 1 -shortest -i whatever.gif [13:49] <sacarasc> Yeah. [13:49] <sacarasc> Well, -shortest is an output option, maybe. [13:50] <iamtakingiteasy> hmm [13:50] <oakwhiz> -loop 1 on the input and -shortest on the output gives "Option loop not found." [13:50] <oakwhiz> I might try another method of accomplishing this [13:50] <iamtakingiteasy> looks like there was a division in ffmpeg' developers which made separate project named avconv [13:51] <iamtakingiteasy> but where should i go? [13:51] <oakwhiz> Since I know that the two files share a least common multiple, I can repeat them until they match in length [13:52] <sacarasc> iamtakingiteasy: If you use ffmpeg, here. If you use libav/avconv, #libav. [13:54] <iamtakingiteasy> i am a bit confused about "libav" things, since ffmpeg seems to be using libavfilter as well. is that wrong? [13:55] <sacarasc> Libav is a project, libav* is the group of libraries both ffmpeg and libav create/use. [13:56] <iamtakingiteasy> aha, so it should be ffmpeg/libav agnostic if i want to write a libavfilter, which means it be available for both projects; but is there any separate channel dedicated to libav* libraries? [13:57] <iive> iamtakingiteasy: ffmpeg merges almost everything from libav, so it libav superset. [13:57] <iive> however afaik the people who designed and implemented avfilter are still ffmpeg developers, and ffmpeg have more filters. [13:58] <iive> this means that if you write the filter, it should be trivial to port it to the other project. [14:00] <iamtakingiteasy> okay, thanks for clarifying it [14:07] <oakwhiz> It turns out that I have no idea how to properly use imagemagick, so I just copied and pasted the same command line argument 2^5 times and now I have a 40MB GIF. [14:11] <oakwhiz> according to "identify -verbose test.gif | grep -i elapsed" it's within a few seconds of the correct length... close enough for me [14:12] <oakwhiz> probably should have made that "identify -verbose test.gif | grep -i elapsed | tail" [14:12] <iive> oakwhiz: what exactly are you doing? [14:12] <oakwhiz> I have an animated GIF and a sound [14:12] <oakwhiz> they are different lengths [14:12] <oakwhiz> I want them to loop [14:13] <oakwhiz> Their lengths share a least common multiple. [14:13] <iive> aha [14:13] <oakwhiz> I wanted to make it so that they match up exactly on that LCM. [14:14] <oakwhiz> Unfortunately there is no built in functionality for this, so I had to manually expand each input file [14:16] <oakwhiz> here goes nothing [14:17] <oakwhiz> Wow it actually worked [14:18] <oakwhiz> Actually scratch that... halfway through it gets stuck [14:18] <oakwhiz> :-( [14:22] <iive> yeh, loop seems to be gif output option and imd2dec input option. [14:30] <oakwhiz> One of my clips has a bit of a problem... the LCM of two adjacent prime numbers is usually a pretty big number. [14:31] <oakwhiz> video is 251 seconds long and audio is 193 seconds long... LCM is 48433 seconds [14:31] <oakwhiz> 13 hours... [14:32] <oakwhiz> lol [14:40] <Hello71> iamtakingiteasy: originally there was ffmpeg which had libraries called libav*, then some people split and called their fork libav to confuse everyone [14:41] <oakwhiz> now instead of having 1 broken video converter with a standard interface, we can divide our efforts amongst 2 broken video converters with different interfaces :-) [14:43] <oakwhiz> Joking of course, ffmpeg has saved my butt on numerous occasions when neither random windows programs nor VLC Player worked [14:54] <sacarasc> zack6849: ffmpeg -i blah -target dvd output.mpg [14:54] <sacarasc> Sorry. [14:54] <sacarasc> zybi1: ffmpeg -i blah -target dvd output.mpg [14:54] <sacarasc> Maybe -target dvd-pal, as you seem to be in Germany. [14:54] <zybi1> ntsc [14:55] <Fjorgynn> so [14:55] <Fjorgynn> never used dvd-pal [15:13] <zybi1> re [15:13] <Fjorgynn> re [15:23] <zybi1> how do i make ffmpeg combine audio (flac) with video (avi) to lossless mkv-video? [15:23] <sacarasc> ffmpeg -i blah.flac -i blah.avi -c copy output.mkv [15:28] <g0twig> Hello [15:29] <g0twig> I need to compress 250GB to something smaller, in a lossless way, any suggestion? It should be played back on a samsung smart tv [15:29] <sacarasc> What does your TV support? [15:29] <g0twig> sacarasc: second [15:29] <g0twig> sacarasc: http://www.samsungdforum.com/Guide/rel00010/index.html [15:31] <sacarasc> Try -c:v libx264 -q 0, but lossless video is usually VERY big. [15:31] <g0twig> sacarasc: but half the size, or? [15:31] <g0twig> sacarasc: you mean I can use h264 with lossless? [15:31] <sacarasc> Yes. [15:31] <g0twig> sacarasc: I compressed lossy h264 250 gb to 21 gb already [15:32] <g0twig> sacarasc: how big do you think it will become [15:33] <sacarasc> I don't know. [15:33] <zybi1> thanks sacarasc !! [15:33] <g0twig> sacarasc: Maybe its not even a compression? [15:33] <g0twig> Maybe it will just change the container :( [15:33] <sacarasc> It does compress, but you almost definitely want to re-encode. [15:34] <sacarasc> As iive is saying in #mplayer. Lossless means something rather specific in these quarters. [15:45] <zybi1> got an error: http://www.pasteall.org/52379 [15:46] <sacarasc> zybi1: You didn't use my command. :D [15:56] <zybi1> sacarasc: I tried now like you wrote: http://www.pasteall.org/52380 [15:56] <zybi1> still no luck! [16:05] <zybi1> I found this: http://askubuntu.com/questions/287461/using-avconv-when-remuxing-to-mkv-is-… [16:05] <zybi1> maybe it's for ffmpeg the same? [16:05] <zybi1> i have to first convert to mp4 and then to mkv? [16:08] <zybi1> this also doesn't work [16:08] <zybi1> http://www.pasteall.org/52381 [16:16] <zybi1> re [16:16] <zybi1> i dunno why, but what finally worked is$ ffmpeg -fflags +genpts -i audio.flac -fflags +genpts -i video.avi -c copy out-va.mkv [16:17] <zybi1> and how to output ac3 and mpeg for authoring a DVD later? is it possible in one command or do i have to do it separately? [16:18] <c_14> just create another output with different output options [16:19] <zybi1> is this a good howto: http://ffmpeg.gusari.org/viewtopic.php?f=25&t=1235 [16:19] <zybi1> ? [16:19] <luc4> Hello! Im trying to transcode an input from stdin with cat in.mp4 | ffmpeg -i - -acodec ac3 -vcodec h264 -preset veryslow -crf 18 out.mov. It seems this is not working. Is the command wrong? [16:19] <zybi1> okay: like ffmpeg -i in.mkv -o video.mpeg -o audio.ac3 [16:19] <zybi1> but is it DVD-compatible ntsc format then? [16:20] <zybi1> should i use somthing like -target ntsc-dvd [16:20] <c_14> I'd probably just use -target ntsc-dvd, it usually just works. [16:20] <c_14> ffmpeg doesn't have a -o option iirc [16:21] <sacarasc> luc4: MP4 has its index at the end of the file, and so catting it often won't work. [16:21] <luc4> sacarasc: oh so not possible. Ok, thanks! [16:21] <sacarasc> *usually has its [16:23] <luc4> sacarasc: any idea if it is possible to get some kind of progress bar when transcoding? [16:24] <sacarasc> Not sure you can without patching ffmpeg. [16:24] <luc4> sacarasc: ok, no problem then. [16:24] <luc4> sacarasc: thanks. [16:28] <zybi1> c_14: the resulting file by using -target ntsc-dvd has crappy quality, it's totally blurry [16:34] <c_14> I just tried it on a couple of my files and the output looks fine. [16:40] <zybi1> can't I tune the bitrate higher somehow or does this command use the highest possible bitrate for dvd automatically? [16:41] <zybi1> or some animation mode for fast movements and so on [16:51] <c_14> I'm pretty sure the dvd target automatically sets the highest bitrate it can. [16:52] <zybi1> ok [17:20] <Fjorgynn> so what's up with this -target pal-dvd? [17:20] <g0twig> video stabilization is included in ffmpeg, how can I use it on an existing video, and produce a new video with same codec/same container [17:27] <zybi1> what does this mean? while creating ntsc-dvd-compliant mpg file from mkv: [swscaler @ 0x245ac00] deprecated pixel format used, make sure you did set range correctly [17:31] <zybi1> Fjorgynn: what's your question? there is pal or ntsc dvd, that's what it means, you gotta decide wheater your dvd will have 29.97fps or 25fps [17:31] <Fjorgynn> zybi1: no I just saw someone earlier talking about dvd or pal-dvd [17:50] <user3> why can't i change the title with this, and how can i speed it up? ffmpeg -i in.mp3 -metadata title="abc" out.mp3 [17:56] <user3> it takes a long time to runs and has not effect on the title [17:56] <user3> no effect [18:01] <sfan5> user3: use "-c copy" to avoid re-encoding anything [18:01] <user3> trying it... [18:04] <user3> it definitely doesn't change the title much faster, any alternative? [18:05] <user3> i mean this still won't change the title [18:07] <user3> but whatever this does it does it much faster [18:09] <sacarasc> user3: Which OS are you on? [18:09] <user3> windows 7 [18:09] <sacarasc> Try using something like mp3tag. [18:09] <user3> is that included in the ffmpeg package? [18:10] <sacarasc> No, it's a different programme. [18:10] <user3> ok, googling it out... [18:12] <relaxed> user3: try ffmpeg -i in.mp3 -map_metadata -1 -metadata title="abc" out.mp3 [18:12] <user3> trying... [18:12] <relaxed> you'll probably have to add all the tags, though [18:13] <Fjorgynn> ligga [18:15] <user3> still doesn''t work but i can try it on linux too, I have to reboot... [18:15] <relaxed> user3: your first example works here [18:15] <relaxed> which version are you using? [18:16] <user3> maybe the problem is with my file, maybe i mess too much with it [18:16] <user3> messed [18:16] <user3> i'll try again, this time with a brandnew file... [18:21] <user3> i'd better try i on linux,bye folks [18:57] <user3> update! ffmpeg -i in.mp3 -metadata title="abc" out.mp3 does indeed change the title (both Windows 7 and Linux) when i check with "ffmpeg -i out.mp3" Initially i thought it didnt because i had only looked at what Windows Media Player shows i.e. the file name instead of the metadata title [19:20] <debianuser> _Crash_Laptop: You can get ffmpeg from some ppa, e.g. jon-severinsson's (apt-add-repository ppa:jon-severinsson/ffmpeg) or samrog131's (add-apt-repository ppa:samrog131/ppa). I've just googled them, never tested. :) [19:20] <debianuser> iamtakingiteasy: You can find some examples of libaa+ffmpeg in mplayer sources. I.e. mplayer actually decodes arbitrary file with ffmpeg and displays it as ascii art (mplayer -vo aa somefile.avi) It even supports colored-ascii-art (mplayer -vo caca somefile.avi) [19:31] <zybi1> a film festival is asking to send the video works in .mov, .mpg, .divx, .avi oder flash format - i don't know why they don't accept .mkv [19:32] <zybi1> which container would you prefer and how to encode from .mkv lossless [19:34] <relaxed> zybi1: try, ffmpeg -i input.mkv -map 0 -c copy out.mov [19:35] <sfan5> zybi1: there is no lossless re-encode, changing the container format is called re-muxing (IIRC) [19:36] <zybi1> relaxed: it says "file may be unplayable" see http://www.pasteall.org/52383 [19:38] <sfan5> seems like .mov does not support flac directly [19:38] <sfan5> s/directly// [19:38] <relaxed> zybi1: ffmpeg -i input -c:v copy -c:a pcm_s16le output.mov [19:39] <relaxed> still lossless [19:39] <sfan5> depending on the transmission medium uncompressed audio may be a bad idea [19:42] <relaxed> I doubt you want to step on the audio for a film festival entry. [19:44] <sfan5> Doesn't mov support ALAC? [19:44] <debianuser> zybi1: Don't know whether mov can contain flac, but, maybe, avi can... [19:45] <sfan5> yes, avi can contain flac [19:45] <sfan5> ..and mov supports ALAC [19:48] <relaxed> does recent Windows support alac out of the box? [19:49] <zybi1> final command line suggestions? [19:49] <zybi1> :) [19:50] <zybi1> good quality and good playability by all systems (osx, win, linux) [19:50] <zybi1> I guess would be nice [19:50] <debianuser> ffmpeg -i input -c copy output.avi [19:51] <relaxed> zybi1: I would say my command is the safest bet [19:52] <debianuser> Have anybody joined many different (wav, ogg) audio files into one with ffmpeg? I'm trying to, and get some errors whatever I try even for a small subset. Either lots of "invalid dropping", or "Non-monotonous DTS". And resulting wav file is either smaller than expected or sometimes larger than expected. [19:53] <sfan5> join = concat? [19:53] <zybi1> ok relaxed thanks! [19:53] <debianuser> sfan5: yes [19:54] <debianuser> Command `echo -e "file 1.wav\nfile 2.ogg\nfile 3.wav" | ffmpeg -f concat -i - -ac 2 -ar 48000 0.wav` gives incorrect output file and lots of errors: pastebin.com/KQQSCGM4 [19:54] <debianuser> Command `ffmpeg -i 1.wav -i 2.ogg -i 3.wav -filter_complex 'concat=n=3:v=0:a=1[a]' -map '[a]' -ac 2 -ar 48000 0.wav` gives a short file (10.92+3.43+12.89!=23.83) and many errors: pastebin.com/nEYL9RLa [19:54] <sfan5> did you try to convert all three files to wav first? [19:54] <debianuser> (I can share the files if needed) [19:55] <debianuser> Two of them are already wav files. :) [19:59] <Fjorgynn> 22.05 vs 44.1 khz? [20:00] <debianuser> sfan5: Just tried. I get the errors separately: "Non-monotonous DTS" while unpacking ogg into wav and lots of "invalid dropping" and other errors when joining them. [20:00] <sfan5> maybe your .ogg is broken [20:02] <debianuser> Well, maybe (then how can I "fix" it or ignore errors?). But wav files can't be broken, right? These 3 test files: ge.tt/1Eg60nl1/v/1 [20:07] <sfan5> debianuser: ffmpeg -i 2.ogg 2.wav && echo -e "file 1.wav\nfile 2.wav\nfile 3.wav" | ffmpeg -f concat -i - -ac 2 -ar 48000 0.wav seems to work [20:10] <debianuser> sfan5: gives me 47.35 seconds file. Seems too long for 10.92+3.43+12.89 sources :) [20:11] <sfan5> I guess you'll have to resort to using something like Audacity then; I don't know of a way to fix this [20:23] <debianuser> Maybe it's a bug... Ill try reporting it to #ffmpeg-devel [21:30] <cuba> anyone here have a working example for using x254 with libav? [21:30] <cuba> *has [21:31] <cuba> x264.h i meant [21:34] <JEEB> demux, decode a picture, convert pix_fmt if needed, feed decoded picture to x264, rince and repeat [21:35] <JEEB> would be pretty much a generic example of lavf/lavc (and possibly swscale) usage, and then pretty much an example of generic libx264 usage [21:35] <JEEB> there's also the libx264 wrapper in lavc that you can look at as well [21:46] <SimAV> Hi, I would like to produce a livestream using ffmpeg/ffserver [21:47] <SimAV> is there a way to specify a "fallback" feed that is used as input whenever the "live"-feed isn't sent yet? [21:48] <SimAV> I haven't found such an option in ffserver config yet [21:48] <SimAV> has anyone in here achieved such an setup or maybe found a workaround? [22:59] <dw2> hi so I want to convert a folder from 10 bit to 8 bit to play on my raspberry pi, is there a way to do a whole folder [22:59] <dw2> ffmpeg -i /home/dw/video/sao -vcodec libx264 -crf 20 -acodec copy /home/dw/video/sao8.mkv [23:00] <dw2> that's what I tried [23:01] <c_14> afaik not ffmpeg-internal. But something like `for file in /path/to/dir/*; do ffmpeg -i "$file" [options] "/path/to/outdir/$file"; done' should work just fine [23:02] <klaxa> pssst sao was bad [23:03] <dw2> yeah I know :p [23:03] <dw2> also converting some good stuff like inferno cop [23:03] <sfan5> you might also want to use -c copy and -map 0 to make sure attachments are copied too (pretty important if there are subs) [23:03] <dw2> thanks [23:03] <klaxa> might as well just hardsub when you are encoding for the pi anyway [23:03] <sfan5> ^ [23:04] <dw2> I'm not a media expert so idk how to do that [23:04] <sfan5> https://trac.ffmpeg.org/wiki/HowToBurnSubtitlesIntoVideo [23:06] <sfan5> would be ffmpeg -i $file -c copy -vcodec libx264 -vf subtitles=$file out/$file then [23:06] <klaxa> you will need to extract the font-files though [23:06] <klaxa> because ffmpeg will not find them in the embedded mkv [23:07] <c_14> IIRC, it actually does. [23:07] <c_14> At least for me. [23:07] <sfan5> ^ same IIRC [23:07] <klaxa> oh? it has been a while since i hardsubbed something [23:11] <sfan5> klaxa: just tested it again, ffmpeg does find the font files in the mkv [23:11] <klaxa> coolbeans [23:13] <dw2> so when I tried [23:13] <dw2> home/dw/videos/sao/*; do ffmpeg -i "$file" -c -map 0 -vcodec libx264 -crf 21 -acodec /home/dw/videos/$file [23:13] <dw2> it said unexpected token do [23:14] <c_14> -c -map is invalid, -c copy -map [23:14] <dw2> bash: syntax error near unexpected token `do' [23:14] <c_14> and you forgot the for file in [23:15] <sfan5> dw2: I'd suggest doing hardsub just to be sure [23:17] <dw2> what's for file in? [23:17] <c_14> It's the bash for loop syntax: `for file in /path/to/dir/*; do [command]; done [23:18] <dw2> gotcha [23:23] <dw2> after I hit enter all there is on that terminal is an > does that mean it's working? [23:24] <c_14> that means you're missing something. ie that thy syntax isn't finished [23:25] <dw2> is it missing something like an output command [23:25] <sfan5> you probably forgot the "done" [23:28] <dw2> for file in home/dw/videos/sao/*; do ffmpeg -i "$file" -c copy -map 0 -vcodec libx264 -crf 21 -acodec /home/dw/videos/$file done [23:28] <dw2> still dosn't work :( [23:28] <c_14> ;done [23:29] <c_14> the semicolon is important [23:29] <dw2> wow, I need to learn bash [23:43] <elec29a> hi hi could someone help me with ffmpeg on windows cygwin android ? [23:56] <elec29a> n00ne here for a helping hand ? [23:58] <iive> don't ask to ask, just ask your question and wait long enough for answer or the entropy death of the universe, whatever comes first. [23:59] <elec29a> ok thx i try to get it in r3questet format [00:00] --- Mon Jun 23 2014

1 0

ffmpeg-devel.log.20140621
by burek 22 Jun '14

22 Jun '14

[00:33] <J_Darnley> What was that joke RFC? IP over Carrier Pigeon? [00:33] <nevcairiel> IP over Avian Carriers [00:34] <nevcairiel> RFC 1149 [00:34] <J_Darnley> :) [01:29] <BBB> J_Darnley: well Im taking advantage of the convention that all 4byte operations zero the upper four bytes (i.e. auto zero extend) [01:31] <Daemon404> Timothy_Gu, fyi: in 109 of libx265.c [01:31] <Daemon404> ctx->params->frameNumThreads = avctx->thread_count; [01:31] <Timothy_Gu> yeah? [01:31] <Daemon404> something to point to rather than benchmarks [01:31] <Daemon404> is all [01:35] <Timothy_Gu> you can add a link to that in the ticket. im too lazy to do it myself :) [01:36] <J_Darnley> BBB: I didn't see anything else when I was looking and it compiled without complaint. [01:36] <J_Darnley> ... but I couldn't test it (patch didn't apply cleanly). [01:37] <J_Darnley> If you have a public repository somewhere, I will clone it tomorrow and try actually running it and do some debugging. [02:05] <BBB> J_Darnley: https://github.com/rbultje/ffmpeg/tree/swr [03:42] <jamrial> BBB: fixed the win64 crash [03:42] <BBB> \o/ [03:42] <jamrial> you were doing "mov ctxq, r0mp" before PUSH r3 [03:42] <jamrial> ctx is r3 [03:43] <BBB> ... [03:43] <BBB> *headbump* [03:43] <jamrial> :P [03:44] <BBB> ok pushed that to github then [03:44] <jamrial> anyway, try doing some benchs on your end. i got some really weird results on my linux x64 vm [03:44] <BBB> so speed is slower? [03:44] <BBB> Ill do a few tomorrow [03:44] <BBB> I havent done anything on that end, just random instruction sequences that work in terms of md5 [03:44] <jamrial> on my second try (win32 and linxu x64 on ubitux's box) it was slightly slower [03:44] <BBB> well slower isnt good, it should be faster :-p [03:45] <BBB> Ill have a look, thanks for testing anyway& youre just using a long mp3 -af & right? [03:45] <BBB> or is there a long test sequence I can use? [03:45] <BBB> like testsrc or so [03:45] <jamrial> a 22 minutes aac 44khz into 48khz [03:47] <jamrial> a very long Dream Theater song :P [03:47] <BBB> I see [03:47] <BBB> ok Il use something similar'ish [03:47] <BBB> bbl, test tomorrow [03:47] <jamrial> ok, later [04:21] <BBB> jamrial: did you test avx? or just win64? [04:21] <BBB> (or has anyone tested the avx?) [04:32] <cone-958> ffmpeg.git 03Michael Niedermayer 07master:1caedf629a06: avcodec/ituh263enc: fix advanced intra coding [04:42] <jamrial> i tested avx and it fate passes [11:41] <J_Darnley> BBB: I apologise that this sounds really stupid but, how do I test your resample re-write? [11:42] <J_Darnley> More than just using -ar and ffmpeg? [11:55] <cone-471> ffmpeg.git 03Diego Biurrun 07master:97578f5f3b27: build: Add missing object file for Matroska demuxer [13:00] <BBB> J_Darnley: I think you take a long music sample and you play it, and then resample (using e.g. ffmpeg -i audiofile -af aformat=fltp,aresample=some_value:internal_sample_fmt=fltp -f null -) [13:02] <BBB> then run that in a debugger or so to ensure it really hits that function, and then put START/STOP_TIMER around the function call in resample.c (*consumed = c->dsp.resample_common[fn_idx](c, dst, src, dst_size, update_ctx);, line 301) or time the whole ffmpeg run, before and after patch [13:02] <BBB> (the timing is obviously not in a debugger) [13:03] <J_Darnley> Ah planar, that might help [13:10] <J_Darnley> Well... I doesn't crash [13:11] <J_Darnley> gdb reaches the breakpoint I set in ff_resample_common_float_sse [13:18] <BBB> ok then youre fine [13:18] <BBB> so go out of debugger and time the function calls or the executable runtime :) [13:19] <J_Darnley> Yeah, its running [13:20] <J_Darnley> with your patch: 11479 decicycles in resample, 698905 runs, 349671 skips [13:20] <J_Darnley> (that's a lot of skips) [13:21] <ubitux> (there is no threading involved, right?) [13:21] <J_Darnley> Almost certainly, ffmpeg uses threads all over the place [13:22] Action: J_Darnley reconfigures [13:35] <J_Darnley> ugh, that's not any better [13:35] <J_Darnley> without patch: runtime 14.98, 10594 decicycles in resample, 698947 runs, 349629 skips [13:38] <J_Darnley> with patch: runtime 15.89, 11006 decicycles in resample, 698933 runs, 349643 skips [13:42] <J_Darnley> I'll be right in a short while. [13:42] <J_Darnley> um [13:42] <J_Darnley> I'll be back in a short while. [14:29] <BBB> ok new patch on github [14:29] <BBB> for me its a lot faster [14:29] <BBB> (I added some loop aligns) [14:53] <BBB> maybe llvm just sucks [14:53] <cone-471> ffmpeg.git 03Anshul Maheshwari 07master:36393434782b: ffmpeg: fix memleak and corruption of AVSubtitle with multiple outputs [15:04] <J_Darnley> FYI: you have left timer macros in it. [15:04] <BBB> in a top patch, to make it easier to reproduce my testing [15:05] <BBB> you can remove the patch if you dont like it :) [15:05] <BBB> I wonder if llvm is really that bad [15:05] <J_Darnley> No I don't mind, I just saw them when git said I had come conficts to merge [15:07] <J_Darnley> dammit git! [15:09] <J_Darnley> uh [15:09] <J_Darnley> is 520 decicycles right? [15:11] <J_Darnley> Maybe that's just your TIMER2 macro [15:13] <J_Darnley> With the usual STOP_TIMER() macro I get the same speed [15:19] <J_Darnley> without patch: runtime 13.29, 10466 decicycles [15:19] <J_Darnley> with patch: runtime 14.07, 11011 decicycles [15:38] <BBB> J_Darnley: you had massive numbers of skips, right? [15:38] <BBB> thats b/c the number of samples per run varies [15:38] <BBB> so I changed it to measure per-sample speed (thats my custom macro) [15:39] <BBB> then number of skips is minimal [15:39] <BBB> and yes per-sample cycle count is expected to be much smaller [15:41] <J_Darnley> that makes sense [15:42] <J_Darnley> I will re-run shortly [15:50] <BBB> yeah total runtime is also faster for me [15:50] <BBB> I wonder if its compiler or something else... [15:50] <BBB> can I see disassembly pooped out for the inline asm functions for you? [15:50] <BBB> and which compiler? [15:55] <ubitux> you can use asetnsamples to control the number of samples going into aresample [15:56] <ubitux> but i guess the internals split them again [15:56] <ubitux> swr internals* [16:07] <BBB> I get near-zero skips with this adaptation, versus about 50% skips before [16:07] <BBB> so I think this is ok [16:08] <BBB> 12.9sec -> 12.3sec total runtime (linear) [16:08] <BBB> or 10.1 -> 9.6 (non-linear) [16:08] <BBB> on 32bit, but I guess its the same for 64bit [16:09] <BBB> commandline: ./ffmpeg -i /tmp/x.wav -af aformat=fltp,aresample=48000:internal_sample_fmt=fltp[:linear_interp=1] -f null -nostats -v error - [16:09] <BBB> where the stuff [] is only for linear [16:09] <BBB> x.wav is a one-hour mp3 file transcoded to wav to decrease runtime spent in mp3 decoding [16:13] <J_Darnley> BBB I'll post those a little later I'm about to go out. [16:13] <BBB> ty [16:13] <J_Darnley> But I am using cygwin's gcc 4.8.2 or .3 [17:05] <cone-471> ffmpeg.git 03Michael Niedermayer 07master:a2de7b1bd504: avcodec/bitstream: document the double volatile [17:05] <cone-471> ffmpeg.git 03Michael Niedermayer 07master:5ab51f753583: avcodec/libtwolame: fix encoding lsf with defaults [18:55] <J_Darnley> BBB: I have what you asked for. [18:56] <J_Darnley> It is gcc 4.8.3 [18:57] <J_Darnley> and here is the output of objdump: http://pastebin.com/xBGpS3zZ [19:33] <cone-471> ffmpeg.git 03Clément BSsch 07master:ded3c9fd32af: avfilter: add hqx filter (hq2x, hq3x, hq4x) [19:39] <wm4> ubitux: :D [19:40] <ubitux> :) [19:40] <ubitux> i'm almost done writing about it for the curious [19:47] <Compnn> ubitux : nice job on hqx stuff :) [19:47] <Compnn> hows it look compared to lanczos ? :P [19:48] <Compnn> on real video not ... pixel art [19:48] <Compnn> i think hq2x is also used on video games , emulators etc ? [19:48] <Compnn> maybe its useful to those x264 guys [19:49] <ubitux> try it [19:49] <ubitux> but it will probably be a very ugly scale [19:50] <ubitux> Compnn: hqx should be compared to xbr or stuff like http://research.microsoft.com/en-us/um/people/kopf/pixelart/ [19:54] <j-b> how does it improve the old DivX blocks? [19:56] <ubitux> lol [19:57] <j-b> :D [20:00] <wm4> j-b: use libpostproc and vf_spp [20:01] <wm4> vf_fspp that is [20:01] <wm4> because supposedly that's the reason why these can't be deleted [20:01] <ubitux> 02:17:20 < michaelni> matrixbench_mpeg2.mpg -vf scale=320:240,format=monow,spp=6:63,hqx=4 [20:01] <ubitux> ;) [20:01] <j-b> wm4: libpostproc is too old and really not good [20:02] <wm4> ubitux: format=monow [20:02] <wm4> wat [20:02] <j-b> wm4: fspp? [20:02] <wm4> j-b: some old libmpcodec crap filter [20:02] <wm4> I think this one uses snow ;D [20:02] <ubitux> no, uspp is snow [20:03] <wm4> oh [20:03] <ubitux> fspp is just a "fast" version of spp [20:03] <ubitux> (fast but different) [20:03] <wm4> lol [20:03] <wm4> anf vf_spp which was ported is not enough? [20:03] <wm4> I don't get this [20:03] <j-b> wm4: but seriously, a lot of the 200x DivX/Xvid around look very very bad on our HD screens [20:03] <ubitux> well it's probably not fast enough ;) [20:04] <wm4> j-b: no doubt [20:04] <wm4> ubitux: having dozens of pp filters is not useful... [20:04] <ubitux> wm4: let's add another one to rule them all [20:04] <wm4> users will have no idea which one to use in the first place [20:04] <wm4> so pick a good one and delete the rest as useless [20:04] <ubitux> probably depends on source and settings [20:05] <j-b> I agree with my enemy wm4 on this :) [20:05] <wm4> enemy? :( [20:05] <j-b> wm4: lol :) [20:05] <j-b> One good pp filter would be nice [20:05] <ubitux> spp isn't good? [20:05] <wm4> ubitux: we don't know [20:05] <iive> spp uses 2d (i)dct, fspp uses a series of 1D (i)dct's [20:05] <wm4> I haven't really seen convincing results from them whenever I tried, either [20:06] <j-b> me neither [20:06] <j-b> gradfun and yadif are amazing piece of filter code, but for pp, it's disappointing for users [20:06] <iive> spp is excellent, but it is kind of slow. [20:06] <wm4> michaelni: you probably know most about these pp filters, your opinion? [20:07] <iive> gradfun and yadif and not postprocessing filters. [20:07] <wm4> j-b: I often see gradfun being ridiculed for being crap [20:07] <j-b> wm4: I see that too. But I disagree. It works wonders for anime and ocean-movies. [20:08] <j-b> it's crap if people apply it everywhere, like Complex Sharpen 2 :) [20:09] <j-b> iive: how slow? [20:09] <iive> j-b: it does about 64 2d fdct and idct for every block. [20:09] <iive> at the maximum (6) level. [20:11] <michaelni> spp with the right parameters can make old low bitrate files look much nicer then without [20:12] <iive> it is also an excellent non-temporal denoiser [20:12] <iive> btw, I wonder if maybe h264 8x8 dct would give slightly better results when h264 video is handled. (and speed wise) [20:15] <cone-471> ffmpeg.git 03Carl Eugen Hoyos 07master:29fc468d0a22: Do not show libzvbi in the configure output if it was not enabled. [20:15] <cone-471> ffmpeg.git 03Carl Eugen Hoyos 07master:e3fd263f0b73: Show duration for large asf files as written in the file header. [20:31] <ubitux> https://news.ycombinator.com/item?id=7925671 here you go [20:38] <cone-471> ffmpeg.git 03Michael Niedermayer 07master:dc5972f88601: avformat/flvdec: give live_flvdec a separate name [21:06] <Daemon404> [19:07] <@iive> gradfun and yadif and not postprocessing filters. <-- gradfun can be used just fine as a postprocessng filter [21:06] <Daemon404> during playback [21:06] <Daemon404> its used to be quite popular to do so [21:07] <Daemon404> unless youre using a strict definition of 'post process' so as not to be 'process post encoding' [21:07] <iive> yes i do [21:07] <iive> i mean post processing that hides encoding artifacts. like blocking and ringing. [21:08] <Daemon404> gradfun can hide banding [21:08] <Daemon404> thats what people use it for during playback [21:08] <iive> while banding could be considered encoding artifact... [21:08] <Daemon404> it de-facto *is* [21:08] <Daemon404> common artefact of dct [21:08] <Daemon404> on flat areas [21:09] <iive> still, gradfin is not replacement for postprocessing filter, it could be used in addition to one. [21:10] <Daemon404> it IS a postprocessing filter, by definition [21:10] <Daemon404> just because libpp is a deblocker, doesnt mean thats all pp is [21:10] <Daemon404> >the english language [21:12] <iive> every filter applied in video chain is post processing, because it is processing post decoding [21:12] <Daemon404> yes [21:12] <iive> however postprocessing have its own meaning, more specific than the english word. [21:12] <Daemon404> more specifcally, in this particular multimedia scene, it is applied to playback filters [21:13] <Daemon404> [20:12] <@iive> however postprocessing have its own meaning, more specific than the english word <-- yes and the one you refer to you are taking from a set of filters that smply used the word since it was convenient [21:13] <Daemon404> they were not named as such due to a definition postprocess [21:13] <Daemon404> you have cause and effect mixed up here [21:13] <iive> no, you ignore it, just to have something to argue about. [21:14] <Daemon404> what the fuck? [21:14] <iive> have a nice day. [21:14] <Daemon404> ...every time you speak, you somethign fuckign retarded [21:14] <Daemon404> for real [21:15] <Daemon404> youre the only person in ere i get the urge to stab in the face on a regular basis [21:15] <ubitux> haha [21:17] <iive> wasn't Daemon404 the one explaining what decimation means, and how we should be using it like avisynth and not use it for what decimation word really means? [21:18] <ubitux> that's indeed another topic where you fight with him [21:25] <J_Darnley> You mean reducing by one tenth? [21:25] Action: J_Darnley runs [21:25] <iive> yes. [21:25] <ubitux> "I asked the author, he apparently made the entire unrolled table by hand." [21:26] <ubitux> holy shit. [21:26] <J_Darnley> :O [21:37] <jamrial> [NULL @ 00000000004c4880] [IMGUTILS @ 000000000032b140] Picture size 27792x16128 [21:37] <jamrial> probably a bit too big source for hqx4 :p [21:39] <jamrial> sorry, full error was "Picture size 27792x16128 is invalid" [21:40] <ubitux> :) [21:42] <jamrial> source was http://lego.wikia.com/wiki/File:MegaManMix.png that, if you're curious [21:43] <jamrial> hqx2 and hqx3 work, but ffmpeg eats ~2gb of ram [21:47] <ubitux> aw. [21:47] <ubitux> strange, i'm not doing much with memory [21:49] <wm4> <jamrial> sorry, full error was "Picture size 27792x16128 is invalid" [21:49] <wm4> fixing this is hard [21:50] <wm4> it's a big mess :( [21:52] <nevcairiel> this one you could probably fix by simply taking bitdepth into account [21:52] <nevcairiel> and not assuming 8 bytes per pixel [21:53] <nevcairiel> but i guess the function doesnt know the bitdepth, and adding it is a API break [21:53] <anshul> Michlani, page_segment and end_segment can be called multiple times in a single call to dvbsub_decode. This will not cause save_subtitle_set() to be called multiple times. since that function is called optionaly according to compute edt value [21:53] <wm4> nevcairiel: yeah, that's a problem too [21:54] <jamrial> i posted that as a curiosity. doubt anyone will actually have a legit use for hqx4 with 6948×4032 images [21:57] <wm4> hm yes, but in general this is a problem [21:58] <wm4> high res images can easily reach these resolutions [21:59] <BBB> J_Darnley: ty! [22:00] <BBB> the main difference appears to be that it saves more stuff in intermediate registers than straight from-memory operations [22:00] <BBB> J_Darnley: so if thats true, then on a 32bit build, we should already be faster than the code that gcc-4.8.3 generates, only on 64bit is it faster, can you test that? [22:02] <michaelni> anshul, if tehre are 2 DVBSUB_PAGE_SEGMENT, dvbsub_parse_page_segment will be called twice [22:03] <michaelni> and it will call save_subtitle_set() twice if compute_edt == 1 [22:03] <michaelni> and if there are 2 DVBSUB_DISPLAY_SEGMENT, dvbsub_display_end_segment will be called twice [22:03] <michaelni> and it will call save_subtitle_set() twice if compute_edt == 0 [22:09] <anshul> Michaelni,In my last patch I moved ffswap in dvbsub_decode but for memleak I will have to add more code [22:10] <anshul> I do need one sample having such segments, without them its difficult for me. [22:12] <J_Darnley> BBB: probably. You want me to compare your patch vs. not on 32-bit cygwin, right? [22:12] <michaelni> anshul, i understand but i dont have such sample [22:12] <J_Darnley> that is basically 32 bit windows anyway [22:15] <anshul> I am comparing data size and ret values because if I dont do that way then parse_end_segment update datasize to 0 while actually it should be 1 [22:18] <anshul> To set datasize only in save_subtitle_set I have to put it in ctx or increase one parameter of some functions, before I was trying to solve with minimum changes. [22:25] <michaelni> theres also the "got_segment == 15 && sub" case that overrides data_size, i think this too can cause it to become wrong [22:25] <michaelni> before the patch this couldnt execute when the other case did i think [22:34] <cone-471> ffmpeg.git 03Clément BSsch 07master:4d8fc0e08828: avfilter/hqx: unroll the pattern calculation [22:34] <cone-471> ffmpeg.git 03Clément BSsch 07master:79198cb65a0b: avfilter/hqx: add some self promotion in doxy. [22:39] <BBB> J_Darnley: yes [22:41] <J_Darnley> 32-bit, cygwin x86, gcc 4.8.3 [22:41] <J_Darnley> without patch: 13.74 runtime, 535 decicycles/sample in float_resample, 1048484 runs, 92 skips [22:41] <J_Darnley> with patch: 14.07 runtime, 543 decicycles/sample in float_resample, 1048448 runs, 128 skips [22:42] <BBB> :( [22:42] <BBB> weird [22:42] <BBB> ok [22:42] <J_Darnley> I also pulled in the latest changes from github [22:42] <BBB> thanks, Ill look what it does [22:42] <BBB> its clearly not just register pre-loading [22:43] <J_Darnley> You want the objdump again? [23:09] <cone-471> ffmpeg.git 03Carl Eugen Hoyos 07release/1.2:6011b806dd5d: Show duration for large asf files as written in the file header. [23:09] <cone-471> ffmpeg.git 03Carl Eugen Hoyos 07release/2.1:f3802aa3250c: Show duration for large asf files as written in the file header. [23:09] <cone-471> ffmpeg.git 03Carl Eugen Hoyos 07release/2.2:52572ca1b339: Show duration for large asf files as written in the file header. [23:25] <cone-471> ffmpeg.git 03Michael Niedermayer 07master:04776cedec82: avfilter/avf_showspectrum: fix macro () [23:25] <cone-471> ffmpeg.git 03Michael Niedermayer 07master:84de3ed795d0: avfilter/deshake_opencl: fix macro () [23:28] <michaelni> ubitux, your fate valgrind client doesnt like your hqx filter http://fate.ffmpeg.org/report.cgi?time=20140621205440&slot=x86_64-archlinux… [23:28] <ubitux> mmh [23:29] <ubitux> that wouldn't be a problem in swscale right? [23:30] <michaelni> could be, didnt really look at it [23:31] <BBB> J_Darnley: hm& that might be useful yes [23:31] <BBB> J_Darnley: theres definitely a few differences, I have to test how relevant each of them is [23:32] <ubitux> michaelni: i wonder if it doesn't like the resolution change [23:34] <ubitux> michaelni: that seems related to it indeed [23:35] <ubitux> michaelni: and i can't reproduce with -cpuflags none [23:35] <ubitux> "valgrind ./ffmpeg_g -i ~/fate-samples/filter/pixelart%d.png -vf hqx=3 -f null -" vs "valgrind ./ffmpeg_g -cpuflags none -i ~/fate-samples/filter/pixelart%d.png -vf hqx=3 -f null -" [23:36] <ubitux> individual pixelart*png don't triggers it [23:36] <ubitux> so it's related to the resolution change [23:37] <michaelni> maybe its rescaling [23:37] <michaelni> from one res to another [23:39] <michaelni> also it might be a false positive (not that i have evidence just saying as there sure is code that overreads) [23:40] <michaelni> overread the odd size usig the padding that is [23:40] <ubitux> i have no idea how the reconfiguration of sws works and is actually triggered [23:41] <wm4> why would there be any scaling, other than hqx3 itself [23:42] <ubitux> rgba bgra maybe [23:42] <ubitux> (png outputs rgba, filter takes bgra) [23:43] <ubitux> it's strange that the issue is not reproducible with hq2x and hq4x though [23:55] <ubitux> it's fun that it's not reproducible with -cpuflags none, but with optimizations the issue happens in yuv2bgra32_full_X_c [00:00] --- Sun Jun 22 2014

1 0

ffmpeg.log.20140621
by burek 22 Jun '14

22 Jun '14

[00:00] <Serus> er, that's a typo [00:00] <Serus> that should be -vf [00:01] <c_14> You can't specify two -vf's on one command line, use one -vf and separate filters with commas [00:01] <c_14> Also, what you're doing won't create webvtt subtitles. [00:01] <c_14> It'll hardcode them. [00:01] <c_14> ffmpeg has a webvtt encoder in current git head. [00:02] <c_14> If that's what you're after. [00:05] <Serus> is there a binary available? [00:05] <Serus> (for windows) [00:05] <c_14> http://ffmpeg.zeranoe.com/builds/ [00:07] <c_14> Also, if the subtitles are already in the WebVTT format (which they might be judging from the .vtt ending) you could probably get away with another input option with -i daishuki.vtt and -c:s copy [00:08] <Serus> do I have to retime them? [00:08] <Serus> since they aren't showing up for me in MPC [00:09] <c_14> Depends, what's your current commandline? [00:10] <c_14> Also, if the subtitle file has the same subs (just in a different format) as what's in the mkv with git ffmpeg you can just use -ss (seek-time) -i input.mkv -t (duration) [codec options] out.webm [00:10] <c_14> That's what I usually do. [00:10] <Serus> I basically use the same command line as on the pastebin [00:11] <Serus> I just use -c:s webvtt now [00:11] <Serus> and not -sn [00:12] <c_14> Did you get rid of the -vf subtitles filter? [00:12] <Serus> yes [00:12] <c_14> That should work, does mpc have a vtt demuxer? [00:13] <Serus> honestly, I don't knwo [00:13] <Serus> know* [00:13] <Serus> but firefox claims the file is corrupt [00:15] <c_14> Can I pm you a link of a file I have with working vtt subtitles so you can test using that? It's 2.4MiB [00:15] <c_14> Firefox does claim that it's corrupt as well though. [00:15] <c_14> That way you'll know if mpc supports it. [00:15] <c_14> So I can eliminate that possibility. [00:16] <Serus> sure [00:18] <c_14> I can't seem to find any information anywhere on whether firefox is supposed to be able to support vtt subtitles. [00:20] <Serus> ah, no subtitles either with this one [00:20] <Serus> Can I burn them in via ffmpeg? [00:21] <c_14> ye, just use the subtitles filter and pass it the mkv as an argument [00:22] <Serus> and will it burn them in so my browser can work with it? [00:22] <c_14> Yep, they'll become part of the video stream. [00:22] <c_14> I can give you another link of when I did that. [00:23] <c_14> That one should work in firefox. [00:23] <c_14> Does in mine anyway. [00:24] <Serus> [Parsed_subtitles_1 @ 0000000002b7fe40] No filename provided! [00:24] <Serus> [AVFilterGraph @ 0000000002b79fc0] Error initializing filter 'subtitles' with args '' [00:24] <Serus> Error opening filters! [00:24] <c_14> That looks like you might have an escaping error somewhere. [00:24] <Serus> ok that video works fine [00:25] <Serus> I use "" to avoid escaping [00:25] <Serus> -vf scale=640:-1,subtitles="[Vivid-Taku] Gochuumon wa Usagi Desu ka - 05 [2C623D33].mkv" [00:25] <Serus> that should be correct right? [00:26] <c_14> looks ok [00:26] <c_14> try enclosing everything from scale to after the last " in single quotes. [00:26] <c_14> ie: -vf 'scale=..."' [00:26] <Serus> ah ok [00:27] <c_14> I usually need 2 layers of escaping with filterchains [00:27] <Serus> ok then I get a no such filter error [00:28] <c_14> no such filter subtitles or no such filter scale? [00:28] <Serus> No such filter: 'scale=640:-1' [00:28] <Serus> or should it be 'scale'=640:-1? [00:29] <c_14> it should be 'scale=640:-1,subtitles="file.mkv"' [00:29] <c_14> If that fails, switch the double quotes with the single quotes. [00:29] <c_14> ie: "scale=640:-1,subtitles='file.mkv'" [00:29] <Serus> ok [00:30] <Serus> Stream #0:2 -> #0:2 (ssa (native) -> webvtt (native)) [00:30] <Serus> That isn't correct right? [00:31] <c_14> looks fine, why? [00:31] <Serus> it's still transforming it into webvtt [00:31] <c_14> oh, [00:31] <c_14> add -sn [00:31] <Serus> instead of burning the subs into the video [00:31] <c_14> It's doing both [00:31] <Serus> ah ok [00:31] <Serus> looks to be working now [00:32] <Serus> it works! [00:32] <Serus> thanks bro [00:32] <c_14> no problem [00:42] <Serus> does the shared version include headers? [00:42] <Serus> or do I need the dev version for that? [00:43] <c_14> I'm guessing dev. [00:43] <Serus> oh god, I'm looking on that website [00:43] <Serus> and the amount of dependencies [00:44] <Serus> I imagine this stuff is a hell to build [00:44] <Serus> especially on windows! [00:44] <c_14> It depends largely on how many external libraries you decide to activate. [00:44] <Serus> ah true [00:45] <Serus> well, then again [00:45] <Serus> who uses libffmpeg in games? :P [13:53] <galex-713> How to know the duration of an audio file, but just that (without having to extract it, I just want to test in a shellscript if its long enough)? [20:37] <pixaal> greetings sirs! I'd like to know how to convert a video to an image sequence of uncompressed pngs or tgas [20:38] <pixaal> 'ffmpeg -i vid.avi frame%04d.png' works fine, but the pngs are compressed [20:42] <pixaal> suppose i could use imagemagick afterwards... [21:14] <cuba> hi, anyone using zeronoe ffmpeg dev/shared? [21:15] <cuba> i get an ordinal error when using the dll from the shared when compiling with the -dev zeronoe package [21:15] <cuba> both are the same version [23:41] <elec29a> hi hi could someone help me to compile ffmpeg on windows cygwin for android ? [23:41] <elec29a> i get arm-linux-androideabi-gcc.exe: error: missing argument to '-mcpu=' [23:42] <sfan5> are you using --enable-crosscompile ? [23:42] <elec29a> yes i am using script from http://roman10.net/src/build_android_r5b.txt [23:47] <sfan5> elec29a: try replacing line 3638 with "cpuflags=" (without quotes) [23:47] <sfan5> of configure [23:49] <elec29a> 3638 is original ? disabled fast_unaligned || enable fast_unaligned [23:51] <sfan5> hm.. [23:52] <sfan5> elec29a: the line before "elif enabled avr32; then" [23:52] <elec29a> yes [23:56] <sfan5> if that doesn't work ask other people, I'm off now~ [23:56] <elec29a> thx bb [00:00] --- Sun Jun 22 2014

1 0

ffmpeg-devel.log.20140620
by burek 21 Jun '14

21 Jun '14

[00:02] <cone-797> ffmpeg.git 03Vittorio Giovara 07master:440842c4eb1d: stereo3d: add missing include guards [00:02] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:b05d50898f5c: Merge commit '440842c4eb1d7709654ec97cd687663d11ef499c' [00:05] <J_Darnley> Oh my. I didn't realise doing a frequency transform would be this complicated. [00:06] <BBB> I love compilers [00:06] <BBB> ffmpeg_g[0x10082e87c]: movss %xmm0, -0x4(%rsp) [00:06] <BBB> ffmpeg_g[0x10082e882]: movss %xmm2, -0x8(%rsp) [00:06] <BBB> ffmpeg_g[0x10082e888]: movss -0x8(%rsp), %xmm0 [00:06] <BBB> ffmpeg_g[0x10082e88e]: movss -0x4(%rsp), %xmm1 [00:06] <BBB> ... [00:13] <ubitux> must check if the stack is readable and writable regularly [00:14] <nevcairiel> it does kinda flip the values around a bit [00:14] <nevcairiel> for whatever reason [00:14] <cone-797> ffmpeg.git 03Anton Khirnov 07release/2.2:6672f672d98c: avconv: make -shortest work with streamcopy [00:14] <cone-797> ffmpeg.git 03Michael Niedermayer 07release/2.2:220bbc44c2cf: Merge commit '6672f672d98cf4887e13321b6f67682cfd1273b4' into release/2.2 [00:14] Action: ubitux missed that [00:59] <cone-797> ffmpeg.git 03Anton Khirnov 07release/1.1:9455a023be9f: matroskaenc: do not write negative timestamps [00:59] <cone-797> ffmpeg.git 03Michael Niedermayer 07release/1.1:6af1793dd5a2: Merge commit '9455a023be9f3915ccf5511a0b8fdb5b8897b2b6' into release/1.1 [01:10] <cone-797> ffmpeg.git 03Anton Khirnov 07release/1.1:e0fcad77618a: avconv: make -shortest work with streamcopy [01:10] <cone-797> ffmpeg.git 03Michael Niedermayer 07release/1.1:45283653a30e: Merge commit 'e0fcad77618a0455ca9c2451ea0aa538597a08c0' into release/1.1 [01:21] <cone-797> ffmpeg.git 03Anton Khirnov 07release/1.1:ba8ab4e7ae01: avconv: do not send non-monotonous DTS to the muxers. [01:22] <cone-797> ffmpeg.git 03Michael Niedermayer 07release/1.1:62899d79a380: Merge commit 'ba8ab4e7ae016cf970b0f335a7933f2db53784bc' into release/1.1 [01:28] <BBB> now lets kill inline avx [01:31] <Daemon404> 30 [01:32] <cone-797> ffmpeg.git 03Sean McGovern 07release/1.1:4310ba273d3b: sgidec: fix an incorrect backport [01:32] <cone-797> ffmpeg.git 03Michael Niedermayer 07release/1.1:724deeb7a8dc: Merge commit '4310ba273d3bcb7f19a4e40b55b9e832568e56e5' into release/1.1 [01:36] <J_Darnley> BBB: is inline avx "more bad" than other asm? [01:37] <J_Darnley> inline asm I mean [01:38] <cone-797> ffmpeg.git 03Sean McGovern 07release/0.10:9c7321e2b898: sgidec: fix an incorrect backport [01:38] <cone-797> ffmpeg.git 03Michael Niedermayer 07release/0.10:c1ac5896ef9a: Merge commit '9c7321e2b8981ec867294309e9cf3833055df78f' into release/0.10 [01:54] <BBB> J_Darnley: yes! [02:04] <cone-797> ffmpeg.git 03Carl Eugen Hoyos 07release/1.2:3ac85d70f09d: Revert two backports to release/1.2 that broke fate. [02:41] <BBB> I guess I gotta find a host to test speed now [02:41] <BBB> or does someone else want to do that? [02:41] <BBB> jamrial: maybe you can test speed? [02:42] <jamrial> alright. although last time my results were very different than michaelni's [02:43] <jamrial> so one of us did something wrong [02:43] <jamrial> probably me [02:43] <jamrial> so i'd recommend you testing on your own as well to cross check resutls :P [02:50] <BBB> ok [02:50] <BBB> just make sure you check a long music sample and you do some per-sample correction or so [02:50] <BBB> otherwise frame length becomes the defining factor [02:50] <BBB> anyway [02:50] <BBB> just do what you feel is right [03:04] <jamrial> i'll add timer.h macros around the c->dsp.resample_{common,linear} calls (no need to bench swri_resample() as a whole since the call overhead didn't change with this patch) [03:04] <jamrial> how do i do per-sample correction, though? [03:18] <BBB> ignore for now [03:20] <BBB> reason is that if you resample a few frames of audio, the number of samples processed per resample call is not the same [03:20] <BBB> so cycle counts vary hugely [03:20] <BBB> but thats the case for before and after so i guess itsn not a major issue [03:20] <BBB> just make sure number of skips didnt change too much [03:22] <jamrial> ok [03:26] <BBB> ty btw :) [03:29] <Timothy_Gu> #3649 what the heck?! [03:29] <Timothy_Gu> ./version.sh: !: not found [03:57] <jamrial> i'm getting a sigsegv on win64 with "movss [dstq], xm0" (end of the horizonal sum). This is sse, not avx [03:57] <jamrial> let me check on a linux x64 vm [04:10] <BBB> linux64 works for me [04:11] <BBB> mac64, same [04:11] <BBB> win64 probably has bugs, I couldnt test that one :) [04:15] <jamrial> yeah, works on linux64 [04:19] <BBB> weird [04:20] <BBB> does the first loop cycle crash, or only after a couple? [04:47] <jamrial> after a couple .loop cycles [04:55] <jamrial> or several, rather [06:01] <cone-232> ffmpeg.git 03Timothy Gu 07master:cc0057a31c70: tests/fate.sh: report different status for different errors [06:10] <llogan> almost done encoding ~1000 hrs of video [06:11] <drv> nyan cat ultimate edition? [06:12] <llogan> a boring government contract [07:21] <llogan> AMD A10-5800K is slow [07:22] <llogan> just an idle server that i conscripted into encoding [07:53] <ubitux> should i wait for reviews on hqx? [08:03] <llogan> you already did [09:11] <ubitux> http://lav.io/2014/06/videogrep-automatic-supercuts-with-python/ cool usage of ffmpeg [09:12] <plepere> the only feedback for my avx2 patch is the SECTION_RODATA 32 ? [09:13] <ubitux> check if that's necessary for the text section as well maybe [09:14] <plepere> well things work for me without it. how do I force a crash if it's not there ? [09:15] <ubitux> no idea :) [09:15] <plepere> so I need SECTION .text 32 ? [09:17] <ubitux> there is a SECTION_TEXT in one of the x264 header [09:20] <plepere> I'll simply put this in the ML and if there's a problem, someone will say it [10:16] <zfox> with ffmpeg 2.0, converting a AAC audio to PCM produced will product a lot of noise, when resampling from AV_SAMPLE_FMT_FLTP to AV_SAMPLE_FMT_S16 . my code such as http://codepad.org/gsJU4Ydp [11:40] <zfox> when i use avresample_convert or swr_convert to convert AV_SAMPLE_FMT_FLTP to AV_SAMPLE_FMT_S16 of a 2 channels,44100 audio,it always has some noise. how to resolve this problem ? the code http://codepad.org/qoknuYAB [12:11] <anshul> is subtitle pts are in 1/1000000 timebase ? [12:23] <cone-296> ffmpeg.git 03Stefano Sabatini 07master:8c443767995d: doc/codecs: extend documentation for thread_type option [12:23] <cone-296> ffmpeg.git 03Stefano Sabatini 07master:d381109ce8ac: lavc/h264_slice: clarify informational message in case of disabled slice multi-threaded decoding [12:57] <J_Darnley> So, I've got a few questions about using libavcodec's avfft.h frequency transforms... [12:58] <J_Darnley> Are all 3 (fft, rdft, dct) suitable for use on audio? [12:59] <J_Darnley> Which one is fastest? I don't need a lot of quality when I put it into 8-bit samples. [13:01] <J_Darnley> How can I know the output range of values? [13:11] <BBB> plepere: alignment is random multiple of 16 [13:11] <BBB> plepere: so in one of 2 cases, its 32 [13:11] <BBB> plepere: so its not a surprise it doesnt crash for 1 of 2 people here [13:11] <BBB> (who have avx2 capable machines) [13:13] <plepere> I don't have an AVX2 capable machine. I'm using the intel SDE to emulate it [13:13] <BBB> well, same thing I guess [13:13] <plepere> maybe, yes [13:35] <plepere> ubitux, how do you do a !avx-enabled in preprocessor ? [13:37] <plepere> to do %if (%1 > 8 && !avx_enabled) || (%1 > 16) [13:47] <BBB> notcpuflag(avx) [13:48] <plepere> thanks BBB [13:48] <michaelni> J_Darnley, all the transforms are fft based so their spee should be similar [13:49] <plepere> I did an ugly if to assign [13:50] <plepere> BBB : but does notcpuflag(avx) return 1 if the CPU is avx capable but the function is in a INIT_XMM sse2 ? [13:50] <BBB> plepere: those are all preprocessor checks [13:51] <BBB> INIT_XMM sse2 means notcpuflag(avx) returns 1 and cpuflag(avx) returns 0 [13:51] <BBB> INIT_XMM avx is the inverse [13:51] <plepere> ok thanks [13:51] <BBB> they are not runtime checks, I mean, its a preprocessor after all [13:53] <plepere> ok thanks. just making sure. :) [13:54] <plepere> it could be at compile check. [13:57] <BBB> is someone interested in fixing my swr patch for win64? [13:57] <BBB> I dont have a win64 machine so cant test [14:01] <J_Darnley> BBB: I can look unless you also need avx [14:12] <Daemon404> ffs that bug hilights me [14:13] <BBB> J_Darnley: no, sse is fine [14:13] <BBB> Daemon404: happens to me all the time when people talk about big buck bunny [14:16] <Daemon404> lol [14:19] <kierank> lol [14:23] <ubitux> BBB: and beaglebone black? :) [14:41] <ubitux> hey saste [14:41] <ubitux> saste: interested in having a look to hqx before i push? [14:41] <saste> ubitux: yes [14:41] <ubitux> ok, i'll give you about 3 minutes [14:41] <ubitux> please hurry up [14:41] <saste> give me 24 hours, if I won't review it feel free to push [14:42] <ubitux> ok :) [14:42] <saste> i'm debugging a curious issue involving several udp sources [14:43] <ubitux> saste: you're back with us? :) [14:43] <ubitux> it's been a long time :) [14:43] <saste> ubitux, i hope so [14:43] <ubitux> cool [14:47] <cone-296> ffmpeg.git 03Timothy Gu 07master:6fdd19cbb26c: doc: remove RELEASE_NOTES [15:25] Action: Daemon404 loves it when thunderbird screws up and says he has 23k new mails in ffmpeg-devel [15:27] <ubitux> "oh btw, i forgot to tell you" [15:27] <ubitux> "here are thousands of mails" [15:34] <wm4> is there a non-shitty GUI mail client yet [15:34] <Daemon404> is there a nonshitty mail client full stop? [15:34] <Daemon404> ubitux, more like it reindexes half the folder on my drive [15:34] <Daemon404> because it screws up [15:35] <Daemon404> so it thinks theyre all unread [15:35] <nevcairiel> i've seen that, it claims all mails in a folder are new because its local thing got messed up [15:35] <Daemon404> yeah [15:35] <J_Darnley> The quality of thunderbird is assisted by gmail's IMAP server(s) [15:35] <Daemon404> lol [15:35] <Daemon404> its still the least bad client ive used (after configuring it to hell) [15:36] <nevcairiel> gmail maps how their interface works with tags to imap folders, thats just not perfect =p [15:36] <Daemon404> my main complain about gmail is its dedup [15:36] <J_Darnley> I'm not surprised people are always touting the next "email killer" [15:36] <Daemon404> i have to bcc myself on ML mails or they dont show up in the folder [15:36] <Daemon404> well it will, but only on its web interface [15:36] <Daemon404> not useful for a mail client [15:37] <Daemon404> there was one other client i liked, but they refuse to add a threaded view [15:37] <nevcairiel> luckily i only use the web interface, given up on actual mail clients a long time ago [15:37] <Daemon404> which is super shitty [15:37] <Daemon404> all threads look flat [15:37] <Daemon404> (copying Mail.app no doubt) [15:38] <J_Darnley> What I dislike most about the web interface is that you need to be logged into google to use it. [15:39] <nevcairiel> you need to be logged into the webmail you're using, shocker :D [15:39] <Daemon404> no he means all of google [15:39] <Daemon404> youtube, search, g+ [15:39] <Daemon404> just lets them track you even better [15:39] <nevcairiel> youtube is separate, fwiw [15:40] <Timothy_Gu> i dont think so [15:40] <Daemon404> it is not [15:40] <Daemon404> nto anymore [15:40] <Daemon404> they merged [15:40] <nevcairiel> i'm logged into gmail but not youtube [15:40] <nevcairiel> so yes. [15:40] <Daemon404> there was a whole shitstorm [15:40] <Daemon404> about g+ [15:40] <Daemon404> i see [15:40] <nevcairiel> they share accounts sure, but not sso [15:40] <Daemon404> right ok [15:40] <Daemon404> search shares [15:40] <Daemon404> as does g+ [15:40] <Daemon404> i know that for sure [15:41] <nevcairiel> everything officially on a google.com domain does [15:41] <Daemon404> i would just switch to my own mail server, but i have too much history using my gmail account [15:41] <Daemon404> in git [15:41] <J_Darnley> At least they still (or did last time I had to use it) the classic HTML only interface [15:41] <Daemon404> and stuff [15:42] <nevcairiel> running your own mail server properly is also a serious PITA [15:42] <Daemon404> it is [15:42] <Daemon404> except there's an easy solution now [15:42] <nevcairiel> i still maintain one, but i've been thinking of giving it up [15:42] <Daemon404> (zimbra) [15:42] <J_Darnley> I have been tempted to try Fastmail and finally leave the googleplex behind [15:43] <nevcairiel> zimbra looks like an entire social-crapload-solution [15:43] <Daemon404> or you could always set up your domain to use gmail [15:43] <nevcairiel> unless there is multiple things with that name [15:43] <Timothy_Gu> speaking of email, does anyone know how x264 changed all the "Jason Garrett-Glaser" to "Fiona Glaser" in the history and in the changes themselves, without changing hashes? [15:43] <Daemon404> nevcairiel, its basically a vm image [15:43] <Daemon404> with a web ui [15:43] <kierank> 2:34 PM <"Daemon404> is there a nonshitty mail client full stop? --> outlook [15:43] <nevcairiel> Timothy_Gu: they changed hashes [15:43] <J_Darnley> They did change hashes [15:43] <Daemon404> kierank, outlook has its own problems [15:43] <kierank> such as being too good perhaps [15:43] <Daemon404> i could not figure out how to get outlook to send non-flowed text emails [15:44] <Daemon404> or "running on not-windows" [15:44] <Daemon404> i run thunderbird on 3 oses [15:44] <kierank> outlook is the reason i use windows [15:44] <kierank> Daemon404: unified calendar and email [15:44] <kierank> nothing does it better [15:44] <Daemon404> i dont use the calende [15:44] <Daemon404> ri fucking hate calendar shit [15:44] <nevcairiel> i've always kinda liked outlook, but i still gave up on mail clients at some point [15:44] <Daemon404> i* [15:44] <kierank> 99% of people do [15:44] <kierank> gmail can't do BST properly [15:44] <Daemon404> kierank, linux desktop is the reason i use windows for day-to-day [15:45] <Daemon404> windows desktop is the reason i use linux desktop for dve work [15:45] <Daemon404> :V [15:45] <Timothy_Gu> really? i found a hash on an old website and it still works [15:45] <Daemon404> kierank, rhel fucked up BST too until i reconfigured by tz [15:45] <nevcairiel> Timothy_Gu: does it also show the new name? [15:45] <Daemon404> via tzconfig [15:45] <Daemon404> or w/e [15:45] <Timothy_Gu> yeah [15:45] <nevcairiel> did it silently redirect you? :p [15:45] <Timothy_Gu> also why does he change the name [15:46] <Timothy_Gu> maybe ;) [15:46] <Daemon404> kierank, UK customs (via parcel force) is satan [15:46] <Timothy_Gu> kierank: even my mom gave up on Outlook :) [15:46] <Daemon404> worst customs ive dealt with [15:47] <kierank> Daemon404: yeah parcelforce are shit [15:47] <kierank> i have a parcel stuck somewhere [15:47] <kierank> right now [15:47] <kierank> ups/dhl is better [15:47] <Daemon404> i have 2 parcels stuck [15:47] <Daemon404> :/ [15:47] <Daemon404> i would use ups or dhl or fedex [15:47] <Daemon404> but the seller only uses usps [15:47] <Daemon404> due to being in rural idaho [15:48] <Daemon404> and that ends up with parcel force [15:48] <Daemon404> rural idaho -> london = 3days [15:48] <Daemon404> london/customs -> me = inb4 3 weeks [15:48] <kierank> have you paid the monies? [15:48] <Daemon404> no [15:48] <Daemon404> they havent sent me a letter yet [15:48] <kierank> lol [15:49] <Daemon404> it sits i the customs backlog, then they have to physically post me a letter [15:49] <kierank> then they are deliberating how much it's cost [15:49] <Daemon404> with a new separate reference number [15:49] <Daemon404> which is not teh tracking number [15:49] <Daemon404> which i enter online, and pay [15:49] <Daemon404> why the shit cant i pay via tracking no?! [15:49] <kierank> you need a parcel reference number [15:49] <kierank> becaus eit's not the same as the tracking number [15:49] <Daemon404> but *WHY* [15:50] <Daemon404> wouldnt it make more sense to let me pay ahead of time via tracking no [15:50] <Daemon404> canada post always let me do that [15:51] <kierank> because you don't know what the customs fees are ahead of time [15:51] <kierank> ebay has a service now though that pays for you and charges you iirc [15:52] <Daemon404> kierank, so update teh db [15:52] <Daemon404> and support it via tracking number [15:52] <Daemon404> there is no reason it needs a new number [15:52] <Daemon404> it's a political problem i guess [15:54] <Daemon404> kierank, my main gripe is: £13.50 "processing fee" [15:54] <Daemon404> from parcel force [15:54] <Daemon404> in adition to VAT [15:54] <Daemon404> which is just bs [15:54] <kierank> yeah that's fucking annoying because i can't claim 13.50 back [15:54] <Daemon404> yeah [15:55] <kierank> the fedex service is slick [15:55] <kierank> it automatically gets your vat number [15:56] <Daemon404> fedex let me pay after delivery [15:56] <Daemon404> so there was no wait [15:57] <Daemon404> but at teh same time i had to phone them and yell at them [15:57] <Daemon404> because they charged me VAT on a package clearly labelled personal belongings [15:57] <Daemon404> when i moved [15:57] <Daemon404> in the end they refused to believe me [15:58] <kierank> yeah the equipment I get sent clearly states that it belongs to the sender [15:58] <kierank> but i still get charged fat [15:58] <kierank> vat [15:58] <nevcairiel> they are stupid like that [15:59] <nevcairiel> i did a bunch of technical reviews for a book a couple years back and got send like 3 copies after it was published, and they charged me full vat .. for free books! [16:00] <JEEB> yeah, here too gifts are taxed by the selling price [16:01] <ubitux> anyone has a pdf or sth of http://www.amazon.com/JPEG-Compression-Standard-Multimedia-Standards/dp/044… ? [16:01] <ubitux> the price is so indecent it makes me really mad [16:01] <JEEB> library genesis? [16:01] <ubitux> more than $170 for a book written in 1992, wth seriously [16:01] <nevcairiel> and thats already marked down from list price 229! [16:02] <Daemon404> ubitux, i can tell it was from 1992 -- look at that sexy cover art [16:02] <nevcairiel> ubitux: german amazon has a bunch of "used" copies of that for as low as ¬16 [16:03] <nevcairiel> guess US amazon does as well [16:03] <JEEBsv> ubitux: library genesis /seems/ to have that one [16:04] <JEEBsv> oh, not exactly that one [16:04] <Daemon404> http://www.openisbn.com/preview/0442012721/ [16:04] <Daemon404> whole book is readbale online here [16:04] <Daemon404> although as jpegs. [16:05] <ubitux> ah, thanks [16:07] <Daemon404> ubitux, entire think is on google books too [16:07] <Daemon404> which has a slightly less bad interface [16:08] <Daemon404> books.google.co.uk/books?isbn=0442012721 [16:08] <ubitux> it looks like a google book wrapper [16:08] <Daemon404> yea [16:08] <J_Darnley> BBB: I the first thing I see that might be a problem are that you probably need to sign or zero extend some of the integers you are loading [16:08] <Daemon404> ubitux, i dont think it was ever digitized [16:08] <Daemon404> as in OCRd [16:08] <Daemon404> why are you looking at a book about jpeg anyway [16:08] <ubitux> that's fine [16:09] <ubitux> i'm looking for a particular section [16:09] <Daemon404> anything interesting? [16:09] <ubitux> i don't know yet [16:09] <Daemon404> o ok [16:12] <ubitux> Daemon404: it's not actually complete [16:13] <ubitux> there are only a few pages [16:13] <Daemon404> damn [16:38] <cone-296> ffmpeg.git 03James Almer 07master:e93028790087: configure: remove superfluous check for AVX/FMA3 support in Yasm [16:59] <cone-296> ffmpeg.git 03Anton Khirnov 07master:874390e16342: lavc: add a convenience function for rescaling timestamps in a packet [16:59] <cone-296> ffmpeg.git 03Michael Niedermayer 07master:4a993335812f: Merge commit '874390e163427c1fe7682ab27924a7843780dbb3' [17:13] <cone-296> ffmpeg.git 03Anton Khirnov 07master:452860d7e056: Use av_packet_rescale_ts() to simplify code. [17:13] <cone-296> ffmpeg.git 03Michael Niedermayer 07master:afa7ab557a80: Merge commit '452860d7e056f9a894cac2fe52cdedeae4c38251' [17:58] <cone-296> ffmpeg.git 03Anton Khirnov 07master:778111592bf5: dvenc: initialize the profile only once, at init [17:59] <cone-296> ffmpeg.git 03Michael Niedermayer 07master:c2b1225c16f1: Merge commit '778111592bf5f38630858ee6dfcfd097cd6c6da9' [18:10] <cone-296> ffmpeg.git 03Anton Khirnov 07master:650dee63c8b1: dv: get rid of global non-const tables [18:10] <cone-296> ffmpeg.git 03Michael Niedermayer 07master:b0fdc76600a1: Merge commit '650dee63c8b1e6693c6cf5983f4a5ed3f571379f' [18:15] <cone-296> ffmpeg.git 03Anton Khirnov 07master:d349afb07bac: dv: cosmetics, reindent [18:15] <cone-296> ffmpeg.git 03Michael Niedermayer 07master:0a995311ceeb: Merge commit 'd349afb07bacccb62eb5369c38d6406d2909d792' [18:34] <jamrial> BBB: speed seems to be much worse here. timer.h macros show it's almost two times slower, and decoding time goes from 11 to 13 seconds using the same 22 minutes long song [19:23] <llogan> J_Darnley: i moved from my own mail server to fastmail and i've been happy with it so far [19:42] <kriegerod> if anybody is willing to help or consult on #3302 please let me know [20:16] <llogan> kriegerod: patches get more love if you send them to ffmpeg-devel (but i'm not sure if you did or not). [20:23] <cone-296> ffmpeg.git 03Moritz Barsnick 07master:10b8481a2634: fix several typos [21:52] <cone-296> ffmpeg.git 03Michael Niedermayer 07master:40cbad9ff7e0: avcodec/wavpackenc: fix () in macros [21:52] <cone-296> ffmpeg.git 03Michael Niedermayer 07master:33f83a215774: avcodec/x86/rv40dsp_init: fix () in macros [22:00] <cone-296> ffmpeg.git 03Michael Niedermayer 07master:329898aa45f5: avcodec/bitstream: try to workaround internal compiler bug in gcc 4.2 [22:11] <jamrial> BBB: nevermind, just tested win32 on that same machine (although native instead of a vm like the linux x64 above), and linux x64 on another machine and in both cases it's only slightly slower than inline [22:11] <jamrial> no idea wtf happened with the results above [23:17] <cone-296> ffmpeg.git 03Justin Ruggles 07master:9e500efdbe0d: Add av_image_check_sar() and use it to validate SAR [23:17] <cone-296> ffmpeg.git 03Michael Niedermayer 07master:0dceefc5fa81: Merge commit '9e500efdbe0deeff1602500ebc229a0a6b6bb1a2' [23:17] <cone-296> ffmpeg.git 03Michael Niedermayer 07master:fe43facd659f: avcodec/utils: check sar only when width/height are set [23:33] <cone-296> ffmpeg.git 03Vittorio Giovara 07master:ca1e36a8e4cd: h264: fix build when error resilience is disabled [23:33] <cone-296> ffmpeg.git 03Michael Niedermayer 07master:4667116c394b: Merge commit 'ca1e36a8e4cd416142487071dbca734567bdaddf' [23:46] <llogan> i'm off IRC for a month (but i'll have email access once a day or so). going to do some non-computer work. see you guys later. [23:49] <Compnn> non computer work? [23:50] <Compnn> lou is working on a farm? [23:50] <Compnn> green acres ! [00:00] --- Sat Jun 21 2014

1 0

ffmpeg.log.20140620
by burek 21 Jun '14

21 Jun '14

[02:08] <RadSurfer> trying to concat .mp4 files, why is it stopping after 1st one? [02:10] <llogan> RadSurfer: because you didn't show your ffmpeg command and the complete console ouptut [02:10] <RadSurfer> huh? [02:11] <RadSurfer> ffmpeg -i "concat:SDV_0223.MP4|SDV_0224.MP4|SDV_0225.MP4|SDV_0226.MP4" [02:11] <RadSurfer> why is that stopping after 1st one? [02:11] <sacarasc> IIRC, you have to use the concat filter with MP4s. [02:11] <RadSurfer> ? [02:12] <sacarasc> http://ffmpeg.org/ffmpeg-all.html#concat-3 [02:14] <RadSurfer> yet it works just fine with vob, etc. Grrr [02:16] <sacarasc> Because you can do cat *.vob output.vob and have it work, but not cat *.mp4 | player [02:17] <RadSurfer> and mkv's have also to be handled special? [02:18] <sacarasc> I think they work with the concat protocol. [02:19] <RadSurfer> Plan B... I have a video I made of Tall Ships sailing vessels coming to our town, overall its only 1:10:00 long, [02:19] <RadSurfer> but of course its over 4gb mp4 files. [02:19] <RadSurfer> trying to make it into a single movie [02:25] <RadSurfer> would be ironic to save out to .vob, if that were possible, and do it that way, lol [02:41] <llogan> RadSurfer: https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20%28join,%20merge%29%2… [02:44] <RadSurfer> meanwhile, I'm trying my own experiment :) [03:12] <RadSurfer> I managed to get my 1:12:00 movie from 3.94 raw .mp4 down to 1.082gb .mkv :-) seems to play well. [03:12] <RadSurfer> but that is still large. [03:16] <BtbN> 1GB for a 1h movie is quite small, at least if it's a good resolution. [03:16] <RadSurfer> 720p [03:16] <BtbN> you won't get it any smaller without massively degrading the quality [03:19] <RadSurfer> 980mb for Parts 1 -4, yet avidemux made it 1.082? that much overhead? [03:19] <BtbN> avidemux can output mkv? [03:20] <RadSurfer> Evidently it can... I used copy mode in this case though. [03:20] <BtbN> which format was it before? [03:20] <RadSurfer> 980mb of .mkv's copied into one... [03:20] <RadSurfer> I can try ffmpeg next :-) see how it compares [03:20] <RadSurfer> doesn't take long either [03:23] <RadSurfer> ok. running -i concat:mkv|mkv... [03:23] <RadSurfer> nope, it stops after 1st one... [03:23] <RadSurfer> can't just concat [03:23] <llogan> use the demuxer [03:23] <llogan> concat demuxer. not concat protocol. [03:23] <RadSurfer> ? [03:24] <RadSurfer> in any event, I did get what I wanted, & it will fit a DVD along w/other stuff. [03:25] <llogan> use two-pass encoding next time if you're trying to target a specific output file size [05:36] <skylto> I'm not solve for days [05:37] <relaxed> what are you trying to do? [05:37] <skylto> How to control the H264 frame rate? [05:39] <skylto> How to set the AVCodecContext parameters? [05:40] <skylto> Á÷ [08:27] <average> is there any free stream for the world cup.. somewhere [08:27] <average> ? [08:30] <llogan> ffplay -f lavfi -i worldcup=team=ghana [08:31] <sacarasc> Hehe. [08:56] <average> llogan: that is so funny [08:56] <average> llogan: you should try doing some stand-up comedy [09:00] <sacarasc> Try Googling for p2p streaming. [09:01] <average> sacarasc: any particular p2p stremaing software you'd recommend ? [09:02] <sacarasc> I don't use it at all. [09:02] <average> ah, that's sneaky [09:02] <sacarasc> I just heard about this kind of thing in other channels. [09:02] <average> I recommend things I never use all the time too! [09:07] <anshul> is there any invalid value of pts [09:08] <anshul> I want to use some value,so that i can take out meaning that its not set [09:11] <anshul> can someone point(mailing list) me disscussion that tell why AV_NOPTS_VALUE is safe value and in real time cant be achived [09:13] <anshul> if any intrested in looking what is it, #define AV_NOPTS_VALUE ((int64_t)UINT64_C(0x8000000000000000)) [10:14] <zfox> with ffmpeg 2.0, converting a AAC audio to PCM produced will product a lot of noise, when resampling from AV_SAMPLE_FMT_FLTP to AV_SAMPLE_FMT_S16 . my code such as http://codepad.org/gsJU4Ydp [10:19] <anshul> how does av_rescale_rnd works ? [10:19] <anshul> what a, b and c parameter means in this function [10:42] <metathink> Hi [10:45] <metathink> How can I set the profile High(a)4.1 please ? [10:45] <metathink> High(a)L4.1 [10:47] <sacarasc> -profile:v high -level:v 4.1 [10:47] <sacarasc> I think. [10:48] <metathink> Thanks :) [10:50] <miep1> I'm still trying to strip audio from a video file. I modified doc/examples/muxer.c so that it never copies any AVPackets with streams larger than 0 (assuming video is always 0 and audio is always higher). [10:50] <miep1> Code is here: http://codepad.org/tNduhPjQ [10:51] <miep1> But my file size reduction is very modest, for example an MP4 file goes from 6773494 (with audio stream) to 6273994 (without), a measly 500 Kb in reduction [10:53] <miep1> I have a feeling I'm doing something wrong, maybe my understanding of 'how to strip audio' isn't correct. My stripping takes place at line 579, where it filters out any packets not associated with stream 0 [10:53] <miep1> Is this the best way of doing it? [10:59] <Mavrik> miep1, did you actually check if stream 0 is audio in your code? [10:59] <Mavrik> or are you just blindly hoping that for EVERY input file the audio will ALWAYS magically be stream 0? :) [11:00] <miep1> Mavrik, blindly hoping, I'll replace that call with a check later. But I'm checking against a file for which I *know* video is stream #0, so for testing purposes I've hardcoded the stream id in. [11:00] <Mavrik> ah [11:00] <Mavrik> sorry, misunderstood [11:01] <Mavrik> yes, losing 50MB of 650MB file when stripping audio is totally normal [11:01] <Mavrik> after all, remember, a 10min mp3 is only like 5-6mb usually ;) [11:01] <Mavrik> (thought you were stripping video away) [11:02] <miep1> ah, nice, that is the test. I'll see what happens if I strip away the video stream, should end up with only audio then :) [11:04] <miep1> Success! So my code does work after all, it's just that my understanding of what would happen was off :) To be honest, I'm used to see audio being a much bigger chunk of data. A typical 650 mb video on my hard drive is over an hour of video, which translates to at least a 100-200mb audio chunk [11:05] <miep1> Hmm, maybe not :) [11:07] <miep1> So what is a good way to check if a stream is an audio stream? Should I look at the identity of the codec? Or is that stored somewhere as metadata? [11:10] <miep1> There is a s stream_identifier int on AVStream, but the comments indicate it is private [11:11] <Mavrik> miep1, AVStream->codec->codec_type == AVMEDIA_TYPE_VIDEO [11:11] <Mavrik> :) [11:13] <Mavrik> lookup AVCodecContext and AVStream on doxygen [11:15] <metathink> How can I set an encoding setting when I use libx264 ? The options we see with mediainfo for example. (Encoding settings : cabac=1 / ref=3 / deblock=1:0:0 / ...) [11:15] <JEEB> you shouldn't [11:15] <JEEB> use presets for speed vs compression [11:16] <JEEB> -preset name http://mewiki.project357.com/wiki/X264_Settings#preset [11:16] <JEEB> also -tune is also available, but you generally don't need that one [11:19] <metathink> I would like to follow a tutorial I found to create lightweight videos using x264 options. [11:20] <JEEB> if that tutorial is setting specific options it is either old or completely incorrect [11:20] <metathink> I already use preset for other things, and I whoul like to understand how to pass theses options to x264 [11:20] <JEEB> do you mean lightweight as in "uses less CPU power to decode"? [11:20] <metathink> Its not a tutorial for ffmpeg [11:21] <JEEB> I don't care what it's for, if it tells you to set specific x264 options it most probably is incorrect [11:21] <JEEB> or just old [11:21] <JEEB> and blindly following some idiocy never leads to good things [11:21] <JEEB> so as I ask you again, do you want to make it easier for a CPU to decode? [11:21] <JEEB> or what do you mean by "lightweight"? [11:21] <metathink> Just want to test myself to understand what theses thing does and how it works. [11:22] <JEEB> you can do that later after I tell you the correct way to do what you want to do [11:22] <JEEB> because otherwise you will end up left with old/incorrect information [11:22] <JEEB> and that is never, ever good [11:23] <metathink> For example, at the bottom of this page, there is some presets: http://dev.beandog.org/x264_preset_reference.html [11:23] <metathink> How can I change an option of this presets ? [11:23] <JEEB> first tell me what you want to do [11:23] <JEEB> then I tell you how to do that correctly [11:24] <metathink> I just want to understant how ffmpeg works and how to create custom preset to change theses options. I'm not trying to do someting special, just to test things and understand how it works. [11:25] <JEEB> then you don't need to poke at those settings [11:25] <JEEB> if you have a very specific use case, then maybe [11:25] <JEEB> but you are just telling me that you just want to poke at it [11:25] <metathink> Bu this IS what I want to test. [11:25] <JEEB> and that usually only leads to misunderstandings [11:26] <JEEB> and it's not "how ffmpeg works" but rather quite a bit "how libx264 works" since thankfully ffmpeg's libx264 defaults are pretty much libx264's defaults [11:26] <metathink> I already know how to use ffmpeg for a lot of thing, what I want to do now is understand how to change thses options. [11:26] <JEEB> the question is, why? [11:26] <JEEB> for a beginner it's really dangerous to start poking at these things [11:27] <metathink> Because i'm curious and it's fun to test things ? [11:27] <JEEB> at one point you had to poke at them, but no more. x264 even has an error if you try to use old ffmpeg defaults :P [11:27] <JEEB> well, what do you want to test? [11:27] <JEEB> how are you going to test that? [11:27] <metathink> What's wrong with you... [11:27] <JEEB> I am interested in helping you, but I'm not going to tell you things that you most probably are going to end up doing worse things with [11:27] <JEEB> so unless you tell me what exactly you want to change and for what reason [11:28] <JEEB> I might be taking you for dumber than you really are, but you really have to understand that 99%+ of all people don't need to set specific API settings, and they can only make things worse with them [11:28] <JEEB> by misunderstanding various results [11:29] <metathink> I'm not trying to archive something, i'm just trying to play with ffmpeg because it's fun. I guess I just have to read stuffs and source code by myself instead of asking questions here.. [11:29] <JEEB> I'm sorry, I just worry about you doing something wrong :( [11:30] <JEEB> some options have a specific option in ffmpeg [11:30] <JEEB> others can be set via -x264opts [11:30] <JEEB> you have to pretty much know the API name for that given setting though [11:30] <JEEB> but really, really -- just be careful to actually know what you're doing if you're comparing stuff [11:30] <JEEB> like [11:30] <JEEB> don't compare SSIM or PSNR values when you are not optimizing for that metric for example [11:32] <JEEB> http://x264dev.multimedia.cx/archives/458 and http://x264dev.multimedia.cx/archives/472 [11:32] <JEEB> are good reading regarding testing encoders [11:33] <metathink> Thanks a lot :) [11:33] <JEEB> and yes, there really are that many people who don't know what they're doing and still publish/release/decide on something [11:33] <JEEB> that's why one just has to be this vigilant when being asked [11:34] <JEEB> because the user might think he needs some information, but that information most probably is going to be misplaced and misused :< [11:34] <JEEB> esp. with libx264, where the thing actually has sane defaults, and the presets/tunings already handle most of the use cases [11:34] <JEEB> for example you can do -preset placebo -tune fastdecode [11:35] <JEEB> to get maximum compression, but then limit out things that can affect decoding speed [11:35] <JEEB> of course, nowadays fastdecode is really rare :) [11:35] <JEEB> it was more needed in the days of ye olde celeron 733MHz [11:37] <JEEB> and of course as I said, many x264 options are poked into ffmpeg's options [11:38] <JEEB> like -preset, tune, crf, maxrate/bufsize etc [11:38] <metathink> In the first place I was interested to understand how YIFY releases are so lightweight. And I figured out there where using x264 options that is not exactly the same as the defaults ffmpeg presets. [11:38] <JEEB> ugh [11:38] <JEEB> YIFY isn't even using x264 correctly :P [11:38] <JEEB> just for your information [11:38] <zfox> when i use avresample_convert or swr_convert to convert AV_SAMPLE_FMT_FLTP to AV_SAMPLE_FMT_S16 of a 2 channels,44100 audio,it always has some noise. how to resolve this problem ? the code http://codepad.org/qoknuYAB [11:38] <sacarasc> YIFY encodes are poo. [11:38] <metathink> ok x) [11:38] <JEEB> trying to plagiarize them does not lead to any good [11:39] <JEEB> if you want small encodes [11:39] <JEEB> you use either 1pass crf (if you have a specific crf value you want to use), or you use 2pass bit rate based encoding (if you have a specific file size/average bit rate over the whole clip you want to achieve) [11:40] <JEEB> and set preset as high as possible [11:40] <JEEB> that should get you started [11:40] <JEEB> after that you can start tweaking things in case you like blur more than pseudo-detail [11:42] <JEEB> zfox, are you making sure you are taking care of the planarity (as in, if it's planar or interleaved audio)? [12:19] <zfox> JEEB. it is planner, format is AV_SAMPLE_FMT_FLTP . [12:19] <JEEB> no, I mean what you output [12:19] <JEEB> if you have made sure the input is correct and correctly utilized [12:23] <zfox> output is interleaved. openSLES device of android [12:23] <JEEB> is the output /really/ interleaved? [12:24] <zfox> yes [12:24] <JEEB> have you made sure it is? [12:24] <JEEB> ok [12:24] <JEEB> then it's the input [12:24] <JEEB> you're dealing somehow wrong with it [12:30] <zfox> i have made two programs, one by avresample_convert and the other by swr_convert. i found the result is not the same. and all put to the openSLES to play ,it has noise all [12:30] <JEEB> yes, then you probably are dealing with either input or output wrong [12:31] <JEEB> I haven't looked at your code, but I recommend you look at some other examples of libavresample or libswresample usage [12:31] <JEEB> also look at what opensles expects from you [12:32] <metathink> JEEB: thanks for all the things you talked about, it helped me [12:33] <JEEB> metathink, no problem. But yeah, just compressing stuff is really not rocket science with libx264 :) [12:34] <JEEB> you just push the preset to the max as much as you can, and in case you are hitting the really low bit rates you might want to poke at some tuning that likes blur more than details with artifacts [12:35] <zfox> thank u ,JEEB. i will try [13:01] <volmatrix> hello everyone, is it possible to encrypt a multicast udp live stream (with ffmpeg of course) ? [14:12] <termos> I get video and audio out of sync when converting from 44.1khz to 48khz audio sample rate, it's about 1-2 seconds and seems constant. What could the reason for this be any ideas? [14:27] <termos> I think the problem is related to nb_samples, for the input it's 1024 but for the output after aresample filter it's shifting between 1114 and 1115 [14:28] <termos> okey using asetnsamples=n=1024 seems to fix the problem, audio is even in sync. [14:28] <termos> rubber duck debugging [15:17] <temhawk> how do you add shadows to text? i see the options "shadowcolor" and "shadowx/shadowy" but no option to actually enable it. shouldn't there be an option like "shadowblur"? [16:57] <SleepyPikachu> Hello, I'm using libavfilter. How should I free the memory allocated by a call to av_strdup? [19:11] <megaTherion> Hello [19:11] <megaTherion> If Im going to choose -vcodec mpeg4, what will it be? Is mpeg4 (not xvid) an built-in encoder in ffmpeg? [19:12] <sacarasc> Yes. It will encode using ffmpeg's own ASP encoder. [19:16] <megaTherion> sacarasc: and that seems to work out on windows media player, I just checked it - but I wasnt so sure ^^ [20:34] <iamtakingiteasy> hello, does ffmpeg has any bindings to aalib? [20:36] <llogan> iamtakingiteasy: it supports libcaca [20:36] <iamtakingiteasy> does libcaca supports non-colored output? [20:36] <llogan> i don't know [20:36] <iamtakingiteasy> :q [20:38] <llogan> Show the list of available dither colors and exit: ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors - [20:38] <llogan> https://ffmpeg.org/ffmpeg-devices.html#caca [20:38] <llogan> color - Set color to be used when rendering text. The accepted values are listed with -list_dither colors. [20:41] <iamtakingiteasy> aha, thanks [21:09] <iamtakingiteasy> hmm, but seems it is only output device, not an encoder [21:09] <iamtakingiteasy> isn't there a way to render all this charcters to video frames? [21:10] <iamtakingiteasy> maybe using some off-screen surface [21:12] <llogan> iamtakingiteasy: ffmpeg -f tty -i input output [21:17] <llogan> ffmpeg -h demuxer=tty [21:28] <iamtakingiteasy> yah. i think i'll go with xvfb :/ [21:28] <iamtakingiteasy> thanks anyway [22:28] <VooDooNOFX> Does anyone know of a good reason to use ffmbc for professional codecs (like dnxhd, prores, etc) instead of ffmpeg 1.x or maybe 2.x? Hasn't most of it been ported into ffmpeg now? [22:32] <iive> afair these were merged in ffmpeg first. [22:33] <iive> ffmbc serves a different purpose/niche, ffmpeg follows libav and breaks api/abi/options quite often. [22:34] <iive> i would also assume that ffmbc is better polished about (de)muxers. [22:35] <llogan> VooDooNOFX: ffmbc stuff is not usually ported to ffmpeg. the lack of a VCS doesn't help [22:36] <iive> amd ffmbc is GPL only, so merging things may also limit the default LGPL build. [22:36] <llogan> that may have been intentional [22:37] <iive> I think it is side effect. [22:41] <llogan> why it was forked in the first place i don't know [22:43] <iive> I think that baptiste had a number of customers and he needed some fixes and hacks, that were getting too much time to be reviewed, or even refused by some purists. [22:45] <iive> after the libav fork, ffmpeg started merging all the libav crap, so ffmbc decided that compatibility is more important. [23:10] <VooDooNOFX> Ok, thanks for the comments. My boss insists on ffmbc for some items, but i've been using ffmpeg since the 1.0 release, and client's Avids, baselights and other video products accept them just fine. [23:44] <ing_jk> How can i copy the source's framerate ? [23:44] <llogan> VooDooNOFX: good to know. if you notice something that ffmbc does that ffmpeg cant that would be good to know too or make a bug report [23:44] <llogan> ing_jk: the input is generally used for the output too [23:45] <VooDooNOFX> Yeah, i'll get better at that. We use ffmpeg pretty heavily here. [23:50] <Serus> hi [23:50] <Serus> I'm trying to add webvtt subtitles while reencoding a part of my video to webm [23:51] <Serus> However the subtitles do not show up, might this be because I'm using a font that's not existent outside the mkv? [23:53] <ing_jk> How can i encode multiple videos with keyframe alignment? [23:55] <Serus> here is the output and the command http://pastebin.com/FkP6crqS [23:59] <c_14> I'm pretty sure -vt isn't a valid option. [23:59] <c_14> Fails on my end anyway, isn't in the manual, and I've never heard of it. [00:00] --- Sat Jun 21 2014

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