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July 2014

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ffmpeg-devel.log.20140720
by burek 21 Jul '14

21 Jul '14

[00:15] <jamrial> speaking of http://coverage.ffmpeg.org/, wouldn't it be better if it's run with cpuflags=0? it doesn't cover .asm files and we've gotten rid of most inline asm by now [00:16] <jamrial> a lot of *dsp.c files outside the x86 folders are pretty red because the asm functions are used instead [00:21] <Compn> jamrial : --disable-optimizations ? [00:22] <Compn> probably just need more builds to be run, like fate i guess [00:30] <ubitux> jamrial: so you suggest makesopts="CPUFLAGS=none"? [00:30] <ubitux> i guess i can do that [00:30] <ubitux> i have --disable-optimizations in the extra_conf already [00:31] <ubitux> jamrial: added CPUFLAGS=none, you'll see the results at next run [00:31] <jamrial> ubitux: cool, thanks [00:36] <michaelni> maybe the code could be run twice for one coverage output, once with cpuflags=none and once normal [00:37] <ubitux> yeah, i thought about running it with threads as well [01:27] <michaelni> mraulet, d249e6828e8a84758010ec020a84dfcd156b585e breaks fate-hevc-conformance-INITQP_B_Sony_1 on alpha [01:28] <michaelni> http://fate.ffmpeg.org/report.cgi?time=20140719192958&slot=alpha-debian-qem… [01:33] <mraulet> all 10bits [01:38] <mraulet> I think I have th bug [01:52] <cone-472> ffmpeg.git 03db0 07master:9ce64ba11122: doc: update the documentation generator to match the new website [01:52] <mraulet> I see this as a potential fix https://github.com/OpenHEVC/FFmpeg/commit/0a8ce1cbdaff7cd026bcf95ee3787bc7b… [02:02] <cone-472> ffmpeg.git 03Clément BSsch 07master:28f31e78eb9c: doc: restore default.css [02:26] <BBB> sps->vps_id = get_bits(gb, 4); [02:26] <BBB> if (sps->vps_id >= MAX_VPS_COUNT) { [02:26] <BBB> libavcodec/hevc.h:#define MAX_VPS_COUNT 16 [02:26] <BBB> what does that do? [02:27] <mraulet> BBB nothing [02:30] <michaelni> mraulet, that patch breaks alignment i think but seems to fix alpha [02:31] <mraulet> I can put 16 for edges [02:31] <mraulet> instead of one [02:41] <michaelni> mraulet, i have a crash with s->sps being null in get_buffer_sao() with that patch [02:42] <michaelni> about alignment, dunno, i thought first it crashed because of aligment but it seems it doesnt, dunno if any simd functions access it and would benefit from align [02:43] <mraulet> at the moment none access to this buffer [02:43] <michaelni> then i guess + 1 is ok, but the correct sps need to be passed to this [02:45] <michaelni> can fix that myself i guess if you are tired/busy [02:47] <mraulet> ok it is almost 3am here :-) [02:56] <michaelni> done [02:56] <cone-472> ffmpeg.git 03Mickaël Raulet 07master:950a4f832661: hevc: fix offset for sao temporary frame [05:47] <Timothy_Gu> ubitux: huh? [05:47] <Timothy_Gu> I can probably fix the fate part [05:47] <cone-472> ffmpeg.git 03Mickaël Raulet 07master:729479a199a9: fate/hevc: update fate rext tests [05:49] <Timothy_Gu> michaelni, ubitux: Can you merge these two pull requests https://github.com/db0company/web/pulls?state=closed ? [05:50] <Timothy_Gu> and take a look at the makeinfo patches on ML and https://github.com/db0company/web/pulls [05:51] <Timothy_Gu> ubitux: for coverage you can take a look at it. [05:54] <michaelni> Timothy_Gu, ill leave it to ubitux, iam too tired atm, dont want to mess up and cherry pick of 940b76dc85547d3e97bd0c08dedba330397a7030 produces conflicts [05:55] <michaelni> ubitux, likely wil be awake tomorrow before me [09:59] <rcombs> wow, new website is super-sexy [10:00] <rcombs> though "Converting videos and audios has never been so easy." <-- feels like this would be better as "video and audio" [10:01] <kurosu> "The FFmpeg project is proud a brand new version" <- of ? [10:02] <rcombs> I see "The FFmpeg project is proud a brand new version of the website made by db0." [10:02] <kurosu> was made then ? [10:02] <kurosu> still confusing [10:02] <rcombs> which seems to be missing a "to launch" or "to announce" or something [10:03] <kurosu> yeah [10:04] <kurosu> I don't think it's very important but the top part before news doesn't fit in my screen height [10:06] <ubitux> oups, seems i was really tired yesterday [10:08] <ubitux> fixed [10:09] <rcombs> fixed which? [10:10] <ubitux> > to announce [10:11] <rcombs> huzzah [10:12] <rcombs> thoughts on "videos and audios"? "videos" seems natural, but I've never heard "audios" anywhere else [10:13] <kurosu> sounds? [10:14] <kurosu> your multimedia files? [10:14] <kurosu> or drop audios altogether [10:14] <rcombs> or just "video and audio" [10:17] <ubitux> i'll make the change as soon as you agree together :p [10:18] <rcombs> those in favor of "video and audio"? [10:18] Action: rcombs \o [10:18] <rcombs> those opposed? [10:19] <rcombs> motion passes; one for, none against, 115 abstain [10:19] <rcombs> (this is how democracy works, right?) [10:20] <ubitux> :) [10:20] <ubitux> ofc [10:21] <rcombs> (and #ffmpeg-devel is run via direct democracy, right?) [10:21] <rcombs> (yeah, let's go with that) [10:22] <kurosu> actually, you just displayed an oligarchy of 1 person :D [10:23] <rcombs> that'd be a dictatorship [10:23] <rcombs> which I may or may not be OK with [10:24] <rcombs> kurosu: would you like to cast an absentee ballot? [10:25] <kurosu> if it means I get a poney, why not [10:25] <kurosu> or bread and some games [10:25] <rcombs> you can have circuses, but no bread [10:26] <ubitux> i suggest a monarchy with king the queen, so please rcombs & kurosu, choose a change, the people of the ffmpeg are begging for you decision [10:26] <ubitux> s/the/and/ [10:26] <ubitux> your* decision [10:26] <rcombs> hear that, kurosu? This is a super-important bit of website copy [10:27] <rcombs> so important that it's the very first issue being brought before the newly-established monarchy [10:27] <rcombs> I'm sticking with my previous suggestion; what do you think, kurosu? [10:28] <ubitux> (dropping the 's'?) [10:28] <rcombs> (yeah) [10:28] <rcombs> (on both words) [10:28] <ubitux> the ss then [10:29] <rcombs> no, `-ss` has to stay or else we can't seek without filters [10:29] <ubitux> ah ok [10:29] <rcombs> :P [10:29] <ubitux> ffmpeg is really complex, i'm getting lost [10:32] <rcombs> (that was a joke) [10:34] <fionag> who's the king and queen? [10:34] <fionag> and if there's kings and queens we need princes and princesses. [10:34] <fionag> and knights, to fight the libav menace [10:45] <ubitux> fionag: yes we'll go at war after when we decided if we need to kill the 's' first [10:47] <rcombs> it seems we're just waiting on someone to second the motion [10:56] <fionag> 's' ? [10:58] <ubitux> fionag: yes the original discussion is how to reword the sentence on our new frontpage [10:59] <ubitux> like should we write "videos and audios", "video and audio", "videos", something else.. [12:27] <J_Darnley> Fucking designers! Apparently they no longer know the meaning of "required". [12:28] <J_Darnley> When I askes what javascript is required for on the download page, the answer should have been "to unhide the windows and mac links". [12:35] <Mavrik> o, new page <3 [12:53] <JEEB> J_Darnley, lol [12:54] <JEEB> yeah, for them "required" means "is it required for all the bells and whistles so that the site looks exactly like they designed" [12:54] <J_Darnley> Yeah... patch on its way. [13:20] <Mavrik> so basically, instead of using JS to hide icons, they used it to show them? :) [13:23] <J_Darnley> Something like that [14:09] <cone-390> ffmpeg.git 03Jan Gerber 07master:73b7a360d8a3: matroskaenc: Don't set language to empty string, use "und" [15:48] <michaelni> ubitux, db0 , on download.html theres a button with "git Git Snapshot", i guess the double git is a typo [15:49] <ubitux> one is an icon [15:49] <ubitux> but yeah i guess we could drop it [15:49] <michaelni> icon + alt="git" ? [15:52] <db0> ok [17:03] <Timothy_Gu> michaelni: yep. http://fortawesome.github.io/Font-Awesome/examples/ [17:09] <cone-390> ffmpeg.git 03db0company 07master:e2842cfe76c9: MAINTAINERS: Add db0 for the website [17:18] <Timothy_Gu> ubitux: any comments on my patches? [18:38] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/0.5:04fb6bb9155a: avcodec/parser: reset indexes on realloc failure [18:38] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/0.5:02ac859dfe1f: avcodec/jpeglsdec: check err value for ls_get_code_runterm() [18:38] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/0.5:43aae0045572: avcodec/vmnc: Check that rectangles are within the picture [18:38] <cone-390> ffmpeg.git 03Dale Curtis 07release/0.5:90c8fa52216b: matroska: Fix use after free [18:38] <cone-390> ffmpeg.git 03Xi Wang 07release/0.5:974c2ad87c34: lzo: fix overflow checking in copy_backptr() [18:38] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/0.5:24a0273cb86e: avutil/lzo: Fix integer overflow [18:39] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/0.5:a81f72e48231: Update for 0.5.14 [18:54] <cone-390> ffmpeg.git 03db0company 07fatal: ambiguous argument 'refs/tags/n0.5.14': unknown revision or path not in the working tree. [18:54] <cone-390> Use '--' to separate paths from revisions [18:54] <cone-390> refs/tags/n0.5.14:HEAD: MAINTAINERS: Add db0 for the website [19:00] <michaelni> Btw, about http://trac.ffmpeg.org/wiki/Downstreams, IMHO it should be sorted by ffmpeg release not by distro/app name [19:01] <michaelni> would be more usefull to see for us who still uses a given branch [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:0916d0f9d1b9: vorbisdec: Check bark_map_size. [19:17] <cone-390> ffmpeg.git 03Dale Curtis 07release/1.0:85b1ce977bd5: matroska: Fix use after free [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:588ae964f4bb: swscale/x86/swscale: fix missing xmm clobbers in yuv2yuvX_sse3() [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:1ec03662c0f2: avcodec/h263dec: Fix use of uninitialized memory from the bitstream buffer [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:5f7e6b9c37a4: avcodec/x86/mpegvideoenc_template: fix integer overflow [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:551f36955c6b: avcodec/h264_mp4toannexb_bsf: prepend global headers before any in stream parameter sets [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:12eb9d911121: avcodec/wma: use av_freep(), do not leave stale pointers in memory [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:81e5ce4c280e: swresample/resample: Limit filter length [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:b1ac8b1d3842: swresample/dither: use av_malloc_array() [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:d7573f88a9b9: swresample/resample: use av_malloc_array() where appropriate [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:f80af81e6e12: swscale/x86/swscale_template: loose hardcoded dstw_offset [19:17] <cone-390> ffmpeg.git 03Anthoine Bourgeois 07release/1.0:b5ec163e5a47: avcodec/dirac_arith: Fix build with PIC and stack-check options [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:0443fe264068: avcodec/x86/idct_sse2_xvid: fix non C99 inline function [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:f3230ba067af: avformat/mpegts: Remove redundant check [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:00476d924902: avcodec/diracdec: fix undefined behavior with shifts [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:43d24fc34cb8: avcodec/g723_1: add assert to help static code analyzers [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:8ef02e4e89a2: avfilter/vf_deshake: fix loss of precission with odd resolutions [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:ee0db8a51f8c: avfilter/filtfmts: Support dynamically allocated in/outputs [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:acba4d1e55e0: avformat/h263dec: Fix h263 probe [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:6b0c1ed11608: swresample: fix AV_CH_LAYOUT_STEREO_DOWNMIX input [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:2277706b3581: avcodec/mjpegdec: Fix undefined shift [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:033b0a4e7fc8: avfilter/graphdump: Fix pointer to local outside scope [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:1930efe23650: avutil/cpu: force mmx on selection of higher x86 SIMD features [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:a116f16e5dc7: avcodec/libvorbisenc: dont add the duration to AV_NOPTS_VALUE [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:8935e2390074: avformat/flvenc: Do not allow creating h263/mpeg4 in flv without unofficial format extensions being enabled. [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:f48d7abf40f3: avcodec/alsdec: Clear MPEG4AudioConfig so that no use of uninitialized memory is possible [19:17] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:8f443e4f32c9: avformat/mpc: attempt to allocate a packet that is not smaller than the data inside it [19:18] <cone-390> ffmpeg.git 03Xi Wang 07release/1.0:ff712a262d31: lzo: fix overflow checking in copy_backptr() [19:18] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:7b5c706494a7: avutil/lzo: Fix integer overflow [19:18] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:3a2c4453f04e: avformat/utils: do not wait for packets from discarded streams for genpts [19:19] <ubitux> Timothy_Gu_: which one? [19:19] <ubitux> (sorry) [19:20] <Timothy_Gu_> all of them ([PATCH 1/3] Use makeinfo to generate html doc for the new website et al.) [19:21] <michaelni> mraulet, smarter, is "Christophe Gisq (3.2K) [FFmpeg-devel] [PATCH] hevc: wait proper position for tmvp" and "Christophe Gisq (5.2K) [FFmpeg-devel] [PATCH 2] hevc: use different thread wait/report for tmvp" ok to be applied ? [19:21] <mraulet> I look into it [19:24] <cone-390> ffmpeg.git 03Michael Niedermayer 07release/1.0:f2a8c429dc60: update for 1.0.10 [19:48] <ubitux> Timothy_Gu_: this will require a different setup on the server [19:49] <cone-390> ffmpeg.git 03db0company 07fatal: ambiguous argument 'refs/tags/n1.0.10': unknown revision or path not in the working tree. [19:49] <cone-390> Use '--' to separate paths from revisions [19:49] <cone-390> refs/tags/n1.0.10:HEAD: MAINTAINERS: Add db0 for the website [19:52] <ubitux> Timothy_Gu_: doc/style.min.css is already in [19:52] <ubitux> for make info i dunno [19:53] <ubitux> (i don't know perl) [19:54] <ubitux> ah i was looking at the wrong patchset [19:55] <ubitux> Timothy_Gu_: no way for patch 3/3 [19:55] <ubitux> keeping in sync that way sucks [20:00] <kurosu> mraulet, michaelni I think it needs more careful analysis [20:00] <kurosu> I don't know if it breaks smething when wpp is used on a frame and not on another [20:01] <kurosu> and 1% is good but depends a lot on how the system behaves [20:01] <kurosu> I mean, the "use different thread" etc [20:02] <kurosu> mraulet, if you have a system with more than 2 cores and could test in either openhevc or ffmpeg... [20:08] <kurosu> BBB / michaelni, also thinking of a new threading api based on reporting x/y [20:09] <kurosu> as vp9/hevc can use structures that are up to 64 pixels wide and up, this might help [20:09] <kurosu> so far, I failed to make it work on hevc [20:10] <kurosu> https://github.com/kurosu/ffmpeg/commit/577561fcf48965dfdf74fc2c12b7675f37d… [20:10] <kurosu> not intended for real review, but just something that may get you interested more than just talk [20:35] <Timothy_Gu> ubitux: I know, and I tried to avoid it. But makeinfo sucks, it automatica [20:35] <Timothy_Gu> automatically inserts markup. [20:36] <Timothy_Gu> There's no way to work around it without doing this ugly hack. [20:36] <ubitux> open a ticket to them and tell them we can't migrate to makeinfo because of this [20:37] <ubitux> this is a maintainance regression, there is no way we're going to maintain 2 version with different split granularity [20:51] <cone-390> ffmpeg.git 03Nidhi Makhijani 07master:9f99a5f1d078: mpegencconetxt: Move rv10-specific orig_width/orig_height where they belong [20:51] <cone-390> ffmpeg.git 03Michael Niedermayer 07master:02e3e3ea51ff: Merge commit '9f99a5f1d078721a30a76aec27c58805b7b87e58' [21:01] <BBB> kurosu: vp9 is lazy as hell, it just does y only [21:01] <BBB> kurosu: I believe h264 does that also? [21:01] <kurosu> BBB: does the api allow that at all ? [21:02] <kurosu> it didn't seem to me [21:02] <BBB> sure [21:02] <BBB> ff_thread_report_progress(frame, y); [21:02] <BBB> and await_progress then adjusts y based on loopfilter and mc filter size [21:03] <BBB> its super-easy [21:03] <kurosu> errr, I mean waiting on x too, or am I missing something ? [21:04] <Timothy_Gu_> ubitux: I found a way to fix it! [21:04] <ubitux> good :) [21:08] <BBB> kurosu: sure, ff_thread_await_progress(frame, y * n_columns + x); [21:08] <BBB> kurosu: e.g. vp8 does that [21:10] <kurosu> weird, I thought about doing this and at some point "not gonna work" [21:10] <kurosu> I'll think about it then [21:11] <BBB> it depends on both halfs of how its used [21:11] <BBB> if you do loopfilter per-row: not gonna work [21:11] <kurosu> it's done per super structure so should work [21:11] <BBB> if your frame handling is strictly per-block: may work [21:11] <BBB> right [21:12] <BBB> vp9 does loopfilter per-row so wont work as-is [21:12] <BBB> I had a proof-of-concept patch to change it, but it wasnt faster [21:12] <mraulet> kurosu it will be complex to manage for tiles [21:12] <kurosu> yeah, it might but the more frequent checks/updates have a cost too [21:12] <BBB> nah tiles is easy, vp9 has that issue too [21:14] <kurosu> mraulet, wpp/slices/tiles may all report different levels of completion at for different y - does it actually work currently unless you force strict raster scan processing ? [21:15] <mraulet> but using y and x to report progress might become complex with tiles [21:15] <kurosu> does it work now ? [21:16] <mraulet> now? means? [21:16] <mraulet> with the current head [21:16] <kurosu> does frame-mt work with threaded decoding of tiles [21:16] <mraulet> yes [21:16] <kurosu> there was support for wpp, but it didn't seem obviously the case for tiles, from the code [21:16] <mraulet> we are reporting at the end of a row [21:17] <mraulet> ah you mean having for in parallel [21:17] <kurosu> yeah [21:17] <mraulet> you cannot have both slice and frame with ffmpeg upstream [21:18] <mraulet> but it is available in openhevc [21:18] <kurosu> ok [21:18] <kurosu> you told me you had wpp+frame-mt too [21:19] <mraulet> the counterparts it is hard to know the number of threads to give for wpp or frame [21:19] <BBB> wpp cant change frame-to-frame can it? [21:19] <kurosu> what's the level of signalling of wpp already? sps ? [21:19] <mraulet> sps [21:19] <kurosu> because you could report x,y report only if wpp is deactivated at the sps level [21:20] <kurosu> but same for slices and tiles, though [21:20] <kurosu> anyway, these are all rethorical if x/y report actually doesn't bring benefit [21:20] <mraulet> x265 has wpp activated inside :-) [21:21] <kurosu> both wpp and frame-mt ? [21:21] <mraulet> if I am not wrong [21:21] <kurosu> well wpp is the most obvious option for many implementations [21:21] <mraulet> we also have tiles and frame-mt but it does not give benefit [21:22] <mraulet> because of loop_filter across_tiles [21:22] <kurosu> that exists also for slices [21:23] <mraulet> slice level parallelism is not active [21:23] <mraulet> we use slice level for tiles and wpp [21:24] <mraulet> but not for slice itself [21:24] <mraulet> right now at least [21:24] <mraulet> michaelni: what is the issue with the arm platform? [21:26] <michaelni> issue ? arm ? [21:27] <iive> bytestream aligned access? [21:27] <mraulet> http://fate.ffmpeg.org/?query=cc:gcc%204.4.3%20%28Ubuntu%204.4.3-4ubuntu5.1… [21:27] <cone-390> ffmpeg.git 03Michael Niedermayer 07master:9a8a18a2e9c3: doc/issue_tracker: Correct sub-domain for mailing list [21:28] <kurosu> potentially the intreadwrite patch as it relates to mv pred [21:29] <mraulet> yeah but there is a new option in the command line of the configure that seems to fix hevc fate [21:29] <mraulet> disable-asm [21:31] <Timothy_Gu_> ubitux: https://gist.github.com/TimothyGu/d9df4cecf8297fb75168 [21:31] <Timothy_Gu_> is it ok? [21:31] <iive> could "libavcodec/arm/mlpdsp_armv5te.S:626: Warning: conditional outside an IT block for Thumb." be releated? [21:32] <iive> all i see are warnings. [21:32] <kurosu> iive: no the crashes are in hevc code [21:32] <mraulet> but fixed with disabled-asm [21:32] <mraulet> the only assembly code for now in hevc is the cabac [21:32] <kurosu> might be a pointer alignment issue [21:33] <kurosu> mraulet, no other dsp used like block copy or the like ? [21:34] <mraulet> no [21:34] <kurosu> anyway, having a backtrace should clear things up [21:34] <mraulet> emu_edge may be [21:34] <iive> is there way to automate backtrace ? [21:35] <kurosu> mraulet, was your patch around sao that used a one-pixel border applied ? [21:35] <kurosu> if yes, that might be it [21:35] <ubitux> Timothy_Gu_: ok, cool :) [21:35] <mraulet> now it is back to green [21:35] <mraulet> http://fate.ffmpeg.org/?query=cc:gcc%204.4.3%20%28Ubuntu%204.4.3-4ubuntu5.1… [21:36] <mraulet> without any disable-asm option [21:36] <mraulet> I am lost [21:36] <mraulet> :-) [21:37] <iive> well, it might use ununitilized data, thus giving random errors. or overheating or faulty hardware. (or emulation). [21:37] <michaelni> well ive no clue either, ill try to build it with similar options and see if i can get it to crash under gdb [21:38] <michaelni> would be nice though if we could extend fate so it always provides back traces [21:41] <iive> maybe `gdb -x runbacktrace.cmd --args "$*" ` [21:43] <iive> runbacktrace.cmd contains "run" and "bt" on separate lines. [21:47] <iive> add "quit" as third line too. [21:48] <michaelni> ideal would be some kind of --enable-gdb that does all [21:48] <michaelni> the needed stuff [21:48] <michaelni> for fate clients [21:49] <iive> i've seen some project link to gdb in order to provide backtrace at segfalut. not sure how it is done. [21:53] <iive> ok, instead of quit -batch option does the trick. [21:53] <iive> michaelni: you can execute tests with added `gdb -batch -x runbt.cmd --args "$*" ` where $* is the current command [22:00] <michaelni> iive doesnt work [22:00] <michaelni> --target_exec='gdb -batch -x runbt.cmd --args' leads to fate failures due to gdb output being interpreted as app output [22:01] <michaelni> like "+[Thread debugging using libthread_db enabled]" [22:03] <iive> oh.. [22:03] <iive> -batch-silent might help [22:03] <iive> i just hope it won't suppress the backtrace [22:03] <iive> it does... [22:04] <iive> and it doesn't seem to suppress everything. [22:07] <kurosu> BBB: neither h264 nor vp8 seems to wait based on the x position [22:07] <BBB> h264 doesn't [22:07] <BBB> I believe vp8 does? [22:07] <BBB> maybe I misremember [22:07] <kurosu> example: ff_thread_await_progress(ref, (3 + y_off + block_h + subpel_idx[2][my]) [22:08] <kurosu> so no [22:08] <BBB> oh it doesn't [22:08] <BBB> how interesting [22:08] <kurosu> if you think about it, you can just do x*row+x [22:08] <BBB> ohwell, sorry [22:08] <kurosu> err y*row+x [22:08] <iive> michaelni: -batch -q seems to suppress most output while showing the backtrace. [22:08] <BBB> y*n_cols+x [22:09] <kurosu> you need to do something like (y&(block_size-1))*row + x*block_size [22:09] <kurosu> *can't just do [22:09] <BBB> right, y/x is clearly in unit block size [22:09] <BBB> which for h264/vp8 would be 16x16, and for vp9/hevc is 64x64 [22:09] <BBB> although I guess it can be less for hevc with variable max_ctb_size [22:10] <BBB> gotta go [22:10] <kurosu> bye [22:10] <kurosu> for now can't make that to work with hevc [22:10] <michaelni> iive, still fails with diffs like "+[Inferior 1 (process 26165) exited normally]" [22:10] <kurosu> as soon as I add progress on an incomplete row, it fails [22:11] <iive> guh,, ERROR "OK" [22:11] <iive> is every text output treated as error? [22:16] <iive> maybe when normal run exits, it can't execute 'bt' thus returning error. [22:25] <iive> michaelni: does -return-child-result help? [22:31] <michaelni> iive why dont you try it? would be faster sure than if i do and post the result and yu interpret, and no that also fails the same way, fate uses the output of some tests and any change causes failure [22:32] <michaelni> stdout output must not chnage [22:32] <iive> i don't have fate here... and that's a problem. [22:32] <iive> i mean, it would take a bit to set it up. [22:33] <michaelni> you dont need to setup a real fate client [22:33] <michaelni> make fate fails too [22:34] <michaelni> you just need --samples=<some dir here> and make fate-rsync to get the samples [22:34] <michaelni> thats unless you already have them somewhere then you cna use that [22:41] <iive> so far I can't find a way to suppress gdb stderr output... [22:41] <iive> so I'm out of ideas. [22:44] <Timothy_Gu> how about *san toolchains or libsegfault? [22:55] <michaelni> none of that is available on that arm box [22:56] <michaelni> also at least valgrind does not work on it [22:56] <michaelni> before someone suggests it [23:00] <iive> that's to be expected, valgrind implements a whole virtual machine/cpu [23:01] <iive> so it is not trivial to port it to other arch [23:04] <michaelni> valgrind supposedly supports arm [23:08] <michaelni> also if someone has a arm box with working valgrind, please setup a fate client, independant of these issues now it would be very usefull [23:24] <Timothy_Gu_> db0, ubitux: updated patch posted. BTW this also fixes the double title as seen here: http://ffmpeg.org/ffmpeg.html [00:00] --- Mon Jul 21 2014

1 0

ffmpeg.log.20140720
by burek 21 Jul '14

21 Jul '14

[01:15] <ac_slate1> hey all. I'm building ffmpeg statically into a small project. I know ill only be using mpeg2 videos in an mpeg2 container. I've tried to compile with the basic demuxers and decoders (mpeg2video, mpeg1video, mpegvideo) and the file protocol, but I get 'Invalid data when processing input'... I cant tell what else I'd need for mpeg2 decoding/demuxing. Thanks! [01:17] <ac_slate1> c_14: yup, http://paste.debian.net/plain/110591 [01:19] <ac_slate1> and on my distro's ffmpeg install, here is what the ffprobe output looks like http://paste.debian.net/plain/110592 [01:20] <ac_slate1> c_14: I'm not actually using the ffmpeg command line applications, but libav*. The ffprobe error matches my error from avformat_open_input [01:24] <ac_slate1> c_14: oh ... doing `mediainfo` on it shows it's an mpeg2ps .. I havent enabled those [01:28] <c_14> maybe also enable the mpegtsraw demuxer [01:54] <ac_slater> c_14: I forgot the parser for mpegvideo [01:55] <c_14> So that was it. [02:52] <jrm> Hi. I'm trying to do a conversion (first time). I'm just trying to get the audio from a video, but I seem to be missing a codec. Here's the output: http://slexy.org/view/s2HzyEpwXx and here's the output of ffmpeg -codecs: http://slexy.org/view/s20y01eA6x. Do you have any tips? [02:53] <c_14> You don't have an mp3 encoder. [02:53] <c_14> Rebuild with --enable-libmp3lame or use a different output codec. [02:56] <jrm> c_14: Thanks. I'm not partial to mp3. Any suggestions, so that it will play the audio on my g/f's android phone. Open is, of course, better. :-) [02:57] <jrm> open format that is [03:00] <klaxa> try vorbis [03:00] <iive> jrm: it is best if you don't recode the audio, try stream copy ( -c copy -vn ) [03:00] <iive> -vn is to disable the video. [03:00] <iive> andoid should be able to play aac [03:00] <klaxa> android should play aac just fine, yeha [03:00] <klaxa> *yeah [03:01] <c_14> ^What they said. [03:02] <jrm> cool, thanks all :) [04:13] <onyx> hey guys, I want to record a rtmp live stream with ffmpeg, is there a way to record 10 mins of the live stream? [04:13] <c_14> ffmpeg stream -t 10:00 file [04:13] <c_14> -codec copy [04:14] <onyx> thanks [04:15] <onyx> so to make sure it will go something like this? "ffmpeg -i rtsp://serverip:port -acodec copy -vcodec copy -t 10:00 c:/abc.mp4" [04:16] <sacarasc> You might have to do C:\, but yeah in theory. [04:16] <onyx> ok, I'll give that a try! [04:17] <onyx> question, does it use a lot of resources to record? [04:18] <sacarasc> Not really. [04:19] <onyx> how about transcoding that .mp4 after, lets say convert it to .flv [04:20] <sacarasc> Depending on codecs you want to use, it could use a lot or not much at all. [04:21] <onyx> I will just want that file to be web optimized [04:22] <sacarasc> FLV is for flash players, not really for the web. [04:23] <onyx> mp4 are more standar right? [04:23] <sacarasc> If you want HTML 5, you would have to do in like 3 formats to support all browsers that support it. [04:24] <onyx> ok, how would I go about doing that? [05:24] <ac_slater> how the heck do I generate the doxygen? [05:25] <ac_slater> (when compiling obviously) [08:10] <CthUlhUzzz> Hi guys! Can I save just the audio track from remote video file, without downloading it completely? [08:11] <sacarasc> Probably not. [08:23] <CthUlhUzzz> sacarasc: I see, that HTTP support partial download using header 'Content-Range'. And perhaps MP4 container support it too. [08:25] <sacarasc> When you mux things, you don't put the video at the start and the audio at the back, you put them in at the same time. [08:34] <CthUlhUzzz> sacarasc: But I can download the audio constantly jumping from place to place? =D [08:34] <sacarasc> [07:11:28] <sacarasc> Probably not. [08:35] <CthUlhUzzz> sacarasc: F**k((( [11:01] <feliwir> Mavrik? [11:02] <feliwir> if i just want the rgba picture i can just return rgba_picture in ur code? (https://gist.github.com/izacus/26a624df7b5536191c84) [11:50] <optix2> I don't really get the doc of the blend filter http://ffmpeg.org/ffmpeg-filters.html#blend - is it possible to change transparency/opacity of each layer or what exactly do c0_mode etc do? [12:18] <vladtv> help me, how encoding 25i to 50p in vlc ore ffmpeg with deinterlace yadif2x? [12:25] <Mavrik> feliwir, I didn't really show you that code for you to copy-paste. [12:26] <Mavrik> but to see how conversion is done for whatever you're doing. [12:26] <feliwir> ye i know i just wanted to know if that rgba_picture is valid for use [12:26] <feliwir> since i don't want to convert to yuv i won't use it anyways :P [12:27] <feliwir> i just want a ordinary rgba_buffer that i can use for my gltexture [12:28] <Mavrik> yeah, that's the rgba picture [12:28] <Mavrik> you can even do the conversion directly into your buffer and just skip the AVPicture allocation alltogether [12:28] <Mavrik> the loop that goes over all rects for each of colors is the important part [12:29] <feliwir> i used to use the data directly from the rectangle [12:29] <feliwir> but that resulted in weird code [12:29] <feliwir> *weird texture [12:30] <Mavrik> yeah [12:31] <Mavrik> because in the rectangle you have only indexes of the color [12:31] <Mavrik> and the colors are stored in data[1] [12:31] <Mavrik> so for each pixel you need to do a lookup of color in data[1] array :) [12:31] <Mavrik> that's what RGBA_IN macro does [12:32] <Mavrik> optix2, it's mathemaical, you set just how you want the output generated from two images [12:33] <Mavrik> optix2, try with all_mode=normal,all_opacity=0.8 for start [12:38] <optix2> Mavrik: I'm actually trying to set transparency of a subitle (ssa format) and thought I could do it with blend [13:53] <Fjorgynn> ffmpeg -i windows.wav windows.flac [14:47] <BlackDream> See This: http://pastebin.com/PPKW2NbZ [14:47] <BlackDream> the stream is working perfectly, though ffmpeg sometimes starts restreaming [14:47] <BlackDream> some other times doesnt even start [14:48] <BlackDream> if i remove flags -global_header -fflags +genpts etc etc doesnt make any change [14:58] <Fjorgynn> :) [16:44] <karika200> hi [16:44] <karika200> is there any way to stream from an ip camera a h264 video in mp4 container? [16:45] <karika200> I get alway the next error: [mp4 @ 0x2651d70]muxer does not support non seekable output [16:47] <karika200> http://pastebin.com/bHcF1zav [16:50] <c_14> Try adding AVOptionVideo movflags frag_keyframe [16:58] <karika200> I added it, but didn't helped [16:58] <karika200> This is the ffserver's output: http://pastebin.com/2frZviNS [16:58] <karika200> I need to publish 2-3 IP cam's output to a website with html5 and as fallback solution with flash [16:58] <karika200> so I could use webm also.. but then i need to transcode the video [16:58] <karika200> from h264 to vp8 [16:58] <karika200> so, if any other container format is available in the mainstream browsers, that can be good for me [16:58] <karika200> but I don't want to transcode the video from h264 :\ [17:04] <c_14> Hmm, it doesn't look like it'll work with mp4. You'll need a different format. [17:07] <karika200> Then I have to transcode my videos.. :/ [17:07] <karika200> thx [17:07] <karika200> Is it an ffmpeg specific thing or it's impossible to use mp4 for streaming? [17:08] <c_14> The latter, although there are some specific hacks that can be used in certain situations. In this case the fault lies with mp4 though. [17:09] <karika200> I see [17:09] <karika200> ty for the help [19:44] <kaotiko> hi [20:05] <onyx> hey guys how do I make it to add the time of the recording to the file name? -acodec copy -vcodec copy -t 10:00 c:/abc.mp4 [20:05] <onyx> I want it to create a file that be abc-2.mp4, abc-3.mp4, and so on... [20:06] <c_14> You might be able to do that with the segment muxer depending on what exactly you want. [20:11] <onyx> something like this? -acodec copy -vcodec copy -t 10:00 c:/abc-hh:mm:ss[.mp4] [20:15] <c_14> On windows something like -t 10:00 c:/abc-%time%.mp4 should work. [20:15] <c_14> Can't verify because I don't have a windows though. [20:18] <onyx> Im running on linux [20:20] <c_14> Then what's with the c:/, anyway, just use c:/abc-`date +%H:%M:%S`.mp4 [20:23] <onyx> ok, ill try that [20:33] <onyx> that worked! [20:52] <onyx> thank you c_14 [20:52] <onyx> im out [22:52] <sebgonz> Hey guys, I'm really really new to ffmpeg, but I have a question. I'm trying to reduce the size of a file. I'm using ffmpeg -i test.mp4 -c:v libx264 -preset -crf 22 output test1.mp4 [22:52] <sebgonz> however I keep getting the error: "Unable to find a suitable output format for '22' [22:52] <sebgonz> Is that something to do with the crf? [22:53] <c_14> You need to specify a preset. [22:53] <c_14> -preset something -crf 22 [22:53] <sebgonz> do I have to use a preset? [22:53] <c_14> No, but you specified the option... [22:54] <sebgonz> Yeah, I was following something else. :) Thanks I'll give that a shot. [22:59] <sebgonz> Do I have to specify both video and audio streams? I keep getting "Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height. [22:59] <sebgonz> 0:1 is the audio [23:00] <JEEB> the error is probably somewhere before that, so pastebin your whole command and output [23:01] <sebgonz> http://pastebin.com/rT1kK9FZ [23:01] <sebgonz> Duh... sorry guys. [23:03] <Mavrik> uh [23:03] <Mavrik> *tries to wrap head around that output* [23:03] <JEEB> it's 5.1 and vo-aacenc doesn't habla that [23:04] <JEEB> which is why [libvo_aacenc @ 051d0b00] Unable to set encoding parameters [23:04] <JEEB> -c:a aac -strict experimental should fix that [23:06] <sebgonz> Looks like it's working. [23:06] <sebgonz> Thanks guys! Now to figure out how to process a whole folder. :) [23:12] <sfan5> JEEB: wouldn't it be better to try libfdk_aac instead of the experimental aac encoder? [23:13] <c_14> sfan5: It doesn't look like he has libfdk_aac enabled. [23:14] <sfan5> how about recompiling? [23:14] <sfan5> how "experimental" is the aac encoder anyway? [23:14] <c_14> He could do that, but the internal encoder isn't that bad if you give it a high enough bitrate. [23:14] <sfan5> doesn't it work fine? [23:16] <sebgonz> I wonder how much faster than handbrake this is... [23:16] <c_14> I've never seen it break anything, but I don't use AAC that much. [23:17] <c_14> sebgonz: I'm pretty sure handbrake uses x264 as well so as long as that's the bottleneck it should be around the same speed (assuming similar settings). [23:18] <sebgonz> Thanks guys. I really appreciate it when I can come into IRC, get help, and not feel like a total tool. [23:28] <phingers> Hey gang, I'm trying to get started with ffmpeg, and just sequence 3 images, into a 10 second movie clip, how do I set the length each image should be displayed? I'm using ffmpeg -f image2 -i foo%03d.png -r 3 -s 640x360 output.mp4 [23:40] <phingers> okay I made it a little further after RTFM - ffmpeg -r 1/5 -i foo%03d.png -r 30 -idct h264 -pix_fmt yuv420p output.mp4 - it is still not h264 streaming though any ideas? [23:44] <phingers> I added in an audio track, and it does stream the sound, but the image is all black [00:00] --- Mon Jul 21 2014

1 0

ffmpeg-devel.log.20140719
by burek 20 Jul '14

20 Jul '14

[00:23] <cone-444> ffmpeg.git 03Mickaël Raulet 07master:fdb20db64259: hevc/rext: put a warning log message instead of an error log message [00:23] <cone-444> ffmpeg.git 03Mickaël Raulet 07master:e1e3ec9b02ca: hevc: fix transquant_bypass [04:05] <cone-371> ffmpeg.git 03Timothy Gu 07master:69c7aad49468: oss_audio: use a macro to simplify ioctl() error checking [04:32] <cone-371> ffmpeg.git 03Diego Biurrun 07master:e0a2e60c0a6c: dct-test: Reuse enum idct_permutation_type instead of duplicating it [04:32] <cone-371> ffmpeg.git 03Michael Niedermayer 07master:a86d527c8b37: Merge commit 'e0a2e60c0a6cbcceef08e94af5081d2aa8e6a52f' [05:38] <cone-371> ffmpeg.git 03Michael Niedermayer 07master:76899be11355: swscale/swscale_internal: add needed version.h [05:38] <cone-371> ffmpeg.git 03Michael Niedermayer 07master:e9f7c7aef96d: sws: Move fast bilinear C code into seperate file [05:38] <cone-371> ffmpeg.git 03Michael Niedermayer 07master:6532a1a8286e: sws/x86: split mmxext fast bilinear scaler out [05:45] <jamrial> michaelni: isn't the above change going to break msvc? [05:45] <michaelni> why ? [05:45] <jamrial> you removed HAVE_<opt>_INLINE guards when you moved the functions to the new file, and added it as part of OBJS instead of MMX-OBJS in Makefile [05:47] <jamrial> you need to either put the guards back, or make the file be part of MMX-OBJS [05:47] <jamrial> assuming it works and i'm not missing anything, the latter is probably cleaner [05:54] <cone-371> ffmpeg.git 03Michael Niedermayer 07master:d5ee3580d84a: sws: move inline asm hscale to MMX_OBJS [05:54] <michaelni> jamrial, fixed i hope [06:30] <Timothy_Gu> Yay! Customized ffmpeg doc with texi2any/makeinfo with db0's new website stype is working! [06:30] <Timothy_Gu> *style [06:30] <Timothy_Gu> I'll submit a pull request when I have time (tomorrow or next week) [07:37] <jamrial> michaelni: it didn't fix. it's tries to build the file and of course fails at it [07:38] <jamrial> so just re-add the guards [07:39] <jamrial> also we should probably look into fixing arch.mak so MMX-OBJS files are not compiled when inline asm is disabled [10:37] <berzerka> how can i check against the libavcodec version macros? [10:38] <wm4> berzerka: include libavcodec/version.h and use LIBAVCODEC_VERSION_INT and AV_VERSION_INT [10:38] <berzerka> does something like #if(LIBAVCODEC_VERSION < 55280000) work? [10:39] <berzerka> wm4: what is the difference between the two? [10:39] <wm4> LIBAVCODEC_VERSION is a string [10:39] <berzerka> ok [10:39] <wm4> you need LIBAVCODEC_VERSION_INT [10:39] <wm4> and build the version number with AV_VERSION_INT [10:40] <berzerka> ok thank you [10:40] <berzerka> ah i see it AV_VERSION_INT(a, b, c) (a<<16 | b<<8 | c) [10:40] <berzerka> got it [10:40] <wm4> meh that macro is in bad style [10:40] <wm4> doesn't put the parameters into ( ) [10:41] <berzerka> but basically that's how it works, i will just use the macro provided. [12:55] <mustafa_muhammad> Hi, I used static builds from http://ffmpeg.gusari.org/static/ and from http://johnvansickle.com/ffmpeg/ with exactly the same parameters and got very differnt results [12:56] <mustafa_muhammad> The file size of the result of ffmpeg.static.64bit.2014-07-15 from http://ffmpeg.gusari.org/static/ is normal, 27MB. [12:58] <mustafa_muhammad> But the file size of the result of ffmpeg-git-20140718-64bit-static from http://johnvansickle.com/ffmpeg/ is 201MB [12:58] <mustafa_muhammad> I used vp9 codec with no audio, and I think there is a big problem in this build [13:03] <nevcairiel> You should ask the people doing those builds, we have no way to check every build around the web [13:06] <mustafa_muhammad> relaxed: Hi, I think this is your build? [13:06] <mustafa_muhammad> nevcairiel: I asked here because these two static builds are on the download page of ffmpeg [13:32] <cone-472> ffmpeg.git 03Michael Niedermayer 07master:54cba3f53efd: swscale/x86/hscale_fast_bilinear_simd: add inline asm guards [13:45] <cone-472> ffmpeg.git 03Nidhi Makhijani 07master:d6e1d37100af: oss_audio: Split muxer and demuxer [13:45] <cone-472> ffmpeg.git 03Michael Niedermayer 07master:80acedae3ed5: Merge commit 'd6e1d37100af568211f28ec0bcf7958a3a2a299e' [13:55] <cone-472> ffmpeg.git 03Diego Biurrun 07master:85cabb8d002f: fdct: Move x86-specific declarations to a header in the x86 directory [13:55] <cone-472> ffmpeg.git 03Michael Niedermayer 07master:6da96a9fc9cf: Merge commit '85cabb8d002f2cd100ced5cc17d87bfc9460d314' [14:07] <ubitux> mustafa_muhammad: probably debug symbols [14:09] <wm4> ubitux: he complains about the size of the _produced_ file, I think [14:10] <cone-472> ffmpeg.git 03Diego Biurrun 07master:5dcc201505f7: simple_idct: Move x86-specific declarations to a header in the x86 directory [14:10] <cone-472> ffmpeg.git 03Michael Niedermayer 07master:776647360db3: Merge commit '5dcc201505f71b1e73e9eef12ce89d4eed252ad0' [14:10] <wm4> i.e. a vp9 file encoded with the builds [14:12] <ubitux> ah mmh sorry [14:12] <ubitux> well, not linking with the same libvpx then :p [14:12] <ubitux> maybe. [14:12] <wm4> maybe [14:16] <cone-472> ffmpeg.git 03Diego Biurrun 07master:1a583c0c6024: fdct: Move ppc-specific declarations to a header in the ppc directory [14:16] <cone-472> ffmpeg.git 03Michael Niedermayer 07master:7bdbd2653fdd: Merge commit '1a583c0c60240adb8fa6620c6df33f1a0a0fe5d9' [14:23] <mustafa_muhammad> ubitux: wm4: Yes, I am talking about the resulting files. [14:30] <cone-472> ffmpeg.git 03Christophe Gisquet 07master:5e55c7e1bcb7: hevc: report more precise progress [14:43] <BBB> mustafa_muhammad: probably want to ask that in #vp8 (or #vp9), since that seems like a libvpx issue [14:43] <BBB> mustafa_muhammad: but just in general, unless you tell us how you attempted to encode the files (what settings, commandline arguments, &), theres not much we can help with [14:44] <BBB> it sounds like encoder defaults were changed [14:44] <BBB> mustafa_muhammad: also, putting the two files up for upload may help analyze, or telling us something about the source material, number of frames, reported bitrate, etc. [14:50] <mustafa_muhammad> BBB: This is the command and its output http://pastie.org/9405065 [15:04] <cone-472> ffmpeg.git 03Christophe Gisquet 07master:8da1defe6986: libavutil: document side effects of macros [15:15] <ubitux> "The unaligned variants [of AV_[RW][BLN](16|24|32|48|64)] are AV_[RW][BLN][8-64]" [15:15] Action: ubitux doesn't get something [15:16] <wm4> just say that the ones with A suffix require aligned pointers [15:16] <wm4> and AV_COPY etc. look so weird, should they be used at all [15:17] <ubitux> ok, so it's missing a '*U' [15:21] <BBB> ubitux: AV_WN32A vs. AV_WN32 [15:21] <BBB> so missing A, not U [15:21] <BBB> and then oddly, AV_COPY32 is actually aligned again [15:21] <BBB> Im not sure why [15:23] <kurosu> well I was not pleased with the documentation either, but mostly because it didn't sound as much clearcut as phrased [15:38] <cone-472> ffmpeg.git 03Christophe Gisquet 07master:7a4a5515b0ce: hevc: use intreadwrite [15:42] <wm4> also I wonder what's the point of this (hevc: use intreadwrite) patch is [15:43] <wm4> apparently no speed advantage, and looks less readable and probably is more UB-prone [15:43] <wm4> well, I don't know the hevc decoder, so I probably better shut up [16:14] <rmklp> is there already an avutil function that converts AVCodecContext.flags or flags2 into something human-readable? [16:24] <wm4> rmklp: I don't think so... if there is, it'd be a function that converts an avopt value to a string, but I don't see such a thing in libavutil/opt.h [16:26] <rmklp> ok, thanks. Ill look around some more but since I do not know any place in the CL tools where something like this is output, I think there is a fair chance it just isnt there. [17:19] <cone-472> ffmpeg.git 03Michael Niedermayer 07master:ba80b8d29b2a: avcodec/hevc: check nb_cpb [17:46] <cone-472> ffmpeg.git 03Michael Niedermayer 07master:d13a731fc149: avcodec/hevc_ps: Check abs_delta_rps [19:05] <kierank> how the hell can malloc be segfaulting [19:08] <wm4> on random memory corruption? [19:35] <michaelni> valgrind should be able to identify the cause [19:50] <kierank> is libavfilter meant to depened on libavcodec still? [19:51] <nevcairiel> It used to for a while unconditionally, but I thought that was fixed [19:52] <ubitux> it's optional [19:52] <ubitux> afaik [19:52] <nevcairiel> Unless you use one of the filters that use avcodec directly [19:52] <kierank> I still get errors about avcodec_find_best_pix_fmt_of_2 [19:52] <kierank> which is used in avfiltergraph.c [19:53] <nevcairiel> I thought we had that in avutil now with a new name [19:53] <nevcairiel> Maybe the patch rots on the ML [19:53] <kierank> I thought too [19:55] <kierank> we do it seems [19:55] <kierank> but why I have it different in my repo I dunno [20:11] <ubitux> Timothy_Gu: we're currently looking at setting it up on the server, but we've some trouble with the lessc insanity :p [20:11] <ubitux> Timothy_Gu: we'll also likely squash all the commit so we can destroy the upload of generated dist in the history [20:13] <ubitux> btw, should i look at integrating the new css to http://coverage.ffmpeg.org/ ? [20:13] <ubitux> what about fate? [20:26] <kierank> turns out I was using av_free instead of avcodec_free_context() [20:39] <Daemon404> woo... ffmpeg built with clang-cl [20:42] <cone-472> ffmpeg.git 03Michael Niedermayer 07master:d9ddbaa9245c: avfilter/avfilter: use av_malloc(z)_array() [20:43] <cone-472> ffmpeg.git 03Michael Niedermayer 07master:a7bb22c5aaba: avformat/seek: use av_malloc_array() [21:47] <cone-472> ffmpeg.git 03Michael Niedermayer 07master:cb169126c5ee: avformat/dvenc: Implement 32khz & 44.1khz for 25/50fps [22:09] <kierank> the parameter for 44.1 audio are very complicated for ntsc [22:10] <michaelni> :( [22:10] <ubitux> Timothy_Gu: ping for the doc [22:41] <rcombs> so, a note: we (Plex, Inc.) are looking into working with Intel to move forwards on a potential solution for getting QuickSync hardware encoding and decoding available in redistributable versions of ffmpeg [22:46] <wm4> lol hardware encoding? [22:46] <Mavrik> rcombs, hmm, isn't QuickSync available mostly only on Windows? [22:47] <rcombs> wm4: it's fast and CPU-cheap! [22:48] <rcombs> Mavrik: the libs are also available on Linux, and OS X has some sort of a facility for it (it's used for AirPlay mirroring from Macs), but I'm not sure of the details there [22:48] <wm4> I think lu_zero did something with quicksync [22:48] <wm4> but not sure what [22:49] <Mavrik> hmm, the Linux part is news to me. [22:49] <rcombs> there's a fork that had support, but because the libs are currently considered nonfree (accurately), it's not redistributable [22:49] <rcombs> https://trac.ffmpeg.org/ticket/2591 see ticket [22:51] <rcombs> we'll be working with Intel to see if either a GPL-compatible library can be released, or if their lawyers can determine whether or not the Media SDK can fall under the GPL System Library exception [22:52] <rcombs> no promises, but we'd really like to see this implemented and available for redistribution, so I'll notify in here and comment on that ticket if something useful comes up [22:52] <ubitux> can anyone tell me what's the difference between the source snapshot and the git snapshot? [22:52] <ubitux> (on our download page) [22:53] <rcombs> source snapshot is just the working tree; git snapshot includes git history [22:53] <ubitux> ah, simply, ok [22:54] <michaelni> rcombs, ok, thanks, more FLOSS is always good [22:54] <rcombs> (this is explained on the page) [22:54] <rcombs> to be painfully honest, I think it's more likely that they'd try to take the System Library exception route, based on previous conversations, but we'll see [23:29] <cone-472> ffmpeg.git 03Michael Niedermayer 07master:d84abf35c0f3: swscale/utils: remove unused define [23:29] <cone-472> ffmpeg.git 03Michael Niedermayer 07master:b53bdae11f1e: swscale/utils: fix rgb -> fullrange yuv [00:00] --- Sun Jul 20 2014

1 0

ffmpeg.log.20140719
by burek 20 Jul '14

20 Jul '14

[00:14] <Einewton> Anyone there? [00:37] <Hello71> no [01:41] <Freakshow> is there anyway to grab from individual pids from mpts over unicast to my instance of ffmpeg? [01:41] <Freakshow> the socket is locked (as I would expect) and everything else fails on connection attempt [08:13] <optix2> I'm trying to burn in subtitles, is the AlphaLevel setting in ssa format not supported? I get solid black irrespective of the value (CLI: http://pastebin.com/izq4xBjJ and part of ut.ssa: http://pastebin.com/qgAzgWcs ) [08:26] <t4nk680> Hello, anyone available to help me with an issue? [08:28] <t4nk680> Just need some help with Persistent Environment Variables section from here: https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu#PersistentEnvironmentV… [08:29] <sacarasc> What about it? [08:32] <t4nk680> I can get man ffmpeg to work [08:32] <t4nk680> but running ffmpeg still returns command not found [08:33] <sacarasc> Does ~/bin/ffmpeg work? [08:33] <t4nk680> yes [08:33] <t4nk680> When I do . ~/.profile I get -bash: MANPATH_MAP: command not found [08:33] <t4nk680> I believe that is where the problem is. [08:34] <t4nk680> I am configuring on a Debian wheezy box. [08:34] <t4nk680> Any idea? [08:36] <sacarasc> I'd do `echo PATH=$HOME/bin:$PATH >> ~/.bashrc` and then run do `source ~/.bashrc` [08:38] <t4nk680> neither one returned anything. [08:39] <sacarasc> What does `ffmpeg` do now? [08:39] <t4nk680> You are a genius sir [08:39] <t4nk680> It worked! [08:39] <sacarasc> Experience doesn't make a genius. [08:40] <t4nk680> heh, well thank you very much. I've been scratching my head for hours on this one. [08:41] <t4nk680> Care to explain what happened? [08:41] <t4nk680> Curious [08:42] <sacarasc> I think it didn't add anything to the PATH, which is why ffmpeg wasn't found. [08:42] Action: sacarasc mostly shrugs. [08:44] <t4nk680> heh alright. [08:44] <t4nk680> Thanks again [08:50] <t4nk680> mmm however, when I run ffmpeg via shell_exec in PHP it is still returning command not found [08:59] <sfan5> t4nk680: php uses a differere $PATH [09:00] <sfan5> I suggest using a wrapper script that sets the path before actually calling ffmpeg [09:00] <t4nk680> heh, I actually just changed it to /root/bin/ffmpeg and no I get permission denied *sigh* [09:00] <t4nk680> So it seems the web server use does not have rights to execute anyways [09:01] <t4nk680> user* [09:01] <sfan5> I'd move it to a more generic locatin like /opt/ffmpeg/ [09:03] <t4nk680> Yeah I'm going to do some more researching tomorrow. [09:04] <t4nk680> It's late and time to sleep lol [10:00] <feliwir> hello, where can i see a few example on how to decode & display subtitles with ffmpeg? [10:17] <mustafa_muhammad> relaxed: Hi, I am using static builds from http://johnvansickle.com/ffmpeg/ to encode vp9 and opus, but the resulting file size is huge. [10:19] <mustafa_muhammad> relaxed: then I separated the audio and video and the file sizes are small [10:20] <mustafa_muhammad> http://pastie.org/9404644 [10:21] <sfan5> 50k is a little low even for opus [10:21] <sfan5> I'd suggest using at least 96k [10:22] <sfan5> did you try tweaking the -crf value? [10:22] <mustafa_muhammad> the resulting file is 206M, the audio stream is 2M and the video is 46.7, why the file is 206M? [10:23] <sfan5> can you paste the console putput too? [10:23] <mustafa_muhammad> Yes, I am running well with another build of ffmpeg, but here the stream file size is larger than video+audio [10:25] <sfan5> this is likely just a problem of outdated ffmpeg then [10:26] <mustafa_muhammad> I am using git from today or yesterday [10:27] <mustafa_muhammad> Used this ffmpeg-2.2.4-64bit-static.tar.bz2 and this ffmpeg-git-20140718-64bit-static.tar.bz2 [10:27] <mustafa_muhammad> from john's site [10:28] <mustafa_muhammad> git from here http://ffmpeg.gusari.org/static/64bit/ works fine but with vorbis (no opus) [10:28] <mustafa_muhammad> and libav from http://johnvansickle.com/libav/ works fine [10:29] <sfan5> the second is libav not ffmpeg [10:30] <mustafa_muhammad> I know, just saying. [10:31] <mustafa_muhammad> so video+audio=48.7 but the resulting file size is 206 [10:31] <ubitux> feliwir: what do you mean exactly? [10:32] <ubitux> using the libraries? [10:33] <ubitux> for the text subtitles you can look at http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavfilter/vf_subtitles.c;h… [10:35] <mustafa_muhammad> sfan5: I want a very small size so I went 50k audio, it is 240p low quality video. I changed the crf value to my liking. Do you still want the console output? [10:35] <sfan5> no [10:38] <feliwir> ubitux: well i find it hard to find any resources on avsubtitles at all [10:39] <ubitux> yes, the current AVSubtitles api is actually here from the beginning of the project, and it's actually not in its best state [10:39] <ubitux> using it from the cmd line is mostly pretty fine [10:40] <ubitux> demuxing & decoding text subtitles is ok [10:40] <ubitux> demuxing & decoding bitmap sub is almost ok [10:40] <ubitux> but the whole thing is a bit clumsy [10:44] <feliwir> well i want to use it within c and there are so many different types like: SUBTITLE_ASS,SUBTITLE_TEXT and SUBTITLE_BITMAP it says nowhere how to handle those differentt types [10:45] <mustafa_muhammad> So is there anything I can do to find what causes this size difference? [10:46] <ubitux> feliwir: you can ignore SUBTITLE_TEXT [10:46] <ubitux> basically, when decoding text subtitles, you always get ass markup [10:46] <feliwir> and i need to parse and render that myself i guess? [10:46] <ubitux> (so you can pipe it to the only decent text subtitles renderer which is libass, as vf_subtitles.c does) [10:47] <feliwir> oh [10:47] <feliwir> i didn't know i need to get libass [10:47] <ubitux> if you want to render them, you'd better have it yeah [10:47] <feliwir> ah thank you for that tip :) [10:47] <ubitux> note that we planed to make possible to decode as "raw text" stripping markup [10:48] <feliwir> "planed" ? [10:48] <ubitux> but the redesigning of subtitles api is an monstruous work [10:48] <ubitux> i've been discussing this for years on the mailing list, and still haven't found a correct way of handling the situation properly [10:49] <sfan5> mustafa_muhammad: is the file still too big if you remux the seperate audio & video streams into one file again? [10:50] <feliwir> that's unfortunate :-/ may i ask which format the bitmap in the subtitles has? because it looks like i decoded it wrong: http://i.imgur.com/oSB4EAs.png :) [10:55] <mustafa_muhammad> sfan5: I'll try now [10:56] <ubitux> feliwir: look into ffplay [10:56] <ubitux> it's RGB [10:58] <feliwir> ah i thought it was rgba [10:58] <ubitux> ah mmh maybe it's rgba yeah [10:58] <ubitux> i meant rgb in the sense "not yuv" :P [10:59] <feliwir> thats weird not sure why i get fragments then [10:59] <feliwir> i was using rgba [10:59] <ubitux> it looks argb actually [10:59] <ubitux> unless i'm missing something [11:02] <mustafa_muhammad> sfan5: OK, sorry, it's all my fault, when I separated the audio and video kept the trimming. [11:02] <feliwir> ah then i might want to convert it before i put it into a rgba texture :D [11:03] <mustafa_muhammad> When trimming correctly the video is 202.3 M [11:08] <mustafa_muhammad> sfan5: Still, the video size is huge compared to the other static build, for the video only (202M vs 27M) [12:09] <feliwir> how would i convert an argb buffer to rgba? [12:09] <feliwir> the results i get look like i am doing sth. wrong [16:19] <feliwir> ubitux i tried to convert to convert the bitmap in the subtitles from argb to rgba but it didn't show good results. Are you sure the pixfmt is argb on those bitmaps? [16:20] <ubitux> i looked at libavcodec/dvdsubdec.c and it looked like it was loading argb [16:21] <feliwir> then i am doing something terrible wrong with the conversion: http://pastebin.com/9D3KTQvE (sorry never used sws_scale before :D) [16:24] <ubitux> you should just try to swap the value yourself for a start [16:24] <feliwir> well i thought sws_scale would perform better [16:24] <ubitux> i wouldn't be so sure about that, but maybe [16:24] <ubitux> and in anycase you want a working code first [16:25] <ubitux> so you'd better just make 2 for loop and a byte swap [16:25] <feliwir> ye the thing is i tried that before and i got an access violation [16:25] <feliwir> but i guess it was my loop being bad [20:51] <BlackDream> Hello, I RUN the following command: ffmpeg -i "http://87.121.34.253:9987/udp/239.100.202.104:1234" -acodec copy -vcodec copy -f mpegts /tmp/test.ts [20:52] <BlackDream> Can someone explain me why it says "frame= 0 fps=0.0 q=-1.0 q=-1.0 size= 0kB time=00:00:04.30 bitrate= 0.0kbits/s" all the time (just the time increasing) [20:52] <BlackDream> and ffmpeg unable to creat the file? [20:52] <BlackDream> the stream is working just fine , i tested in my vlc [20:52] <BlackDream> i was about to say that i dont get any error but ok wait to paste it there [20:53] <c_14> There's no error? Have you tried playing the stream with ffplay? [20:55] <BlackDream> http://pastebin.com/gXZZkcYF [20:55] <BlackDream> says frame 0 and fps 0 all the time [20:55] <BlackDream> no but its working on vlc [20:56] <c_14> Ye, but vlc does funky things sometimes so I want to see what ffplay does. [20:56] <BlackDream> a command example of ffplay [20:57] <c_14> ffplay http://[stream] [20:58] <BlackDream> Could not initialize SDL - DirectFBCreate: Initialization error! [20:59] <c_14> You probably don't have SDL or something. Anyways since the stream is public I'm going to test a few things. [20:59] <c_14> I think the stream is not sending or sending incorrect pts. [21:00] <BlackDream> Test it Please, cause if it is a bug we can report it [21:10] <BlackDream> c_14: Any news my friend :P [21:12] <c_14> Well, I've found that reencoding the video instead of just copying it fixes the issues. [21:13] <BlackDream> you mean something like -vcodec libx264 [21:13] <c_14> yeah [21:13] <BlackDream> but why without copy isnt working :( This will consume a lot of CPU [21:13] <BlackDream> :( [21:13] <c_14> The one thing that's confusing me though is that ffplay can play the stream after throwing errors for a bit, but won't play the streamcopy... [21:14] <BlackDream> Do you think it's ffmpeg issue? [21:20] <c_14> I think the issue is on both sides, but since ffplay can play the stream, ffmpeg _should_ imo be able to streamcopy it and have the result be playable as well, so it might be a bug. [21:21] <BlackDream> should i report it? [21:22] <c_14> Might be a good idea. [21:40] <feliwir> ubitux is the bitmap for subtitles planar? [21:43] <ubitux> i don't think so [22:09] <Mavrik> no it isn't... but it's usually palletized [22:09] <Mavrik> and I'm not sure if swscale added colorspace conversion for those palettes yet [22:26] <feliwir> what does palletized mean? [22:27] <Mavrik> this: http://en.wikipedia.org/wiki/Indexed_color [22:27] <Mavrik> ffmpeg will give you palette on one data plane [22:28] <Mavrik> and subtitle bitmap with indexes instead of ARGB values on another [22:28] <feliwir> does it have to do with the nb_color thing? [22:28] <feliwir> i really didn't understand the purpose of that [22:28] <feliwir> (bad documented ;) ) [22:29] <Mavrik> lemme whip out my code from a year ago o ros [22:29] <feliwir> ty good sir [22:32] <Mavrik> https://gist.github.com/izacus/26a624df7b5536191c84 [22:33] <Mavrik> you could probably do it more efficiently [23:02] <matham> I'm replicating the ff_load_image code, except I'm using frame referencing so I'm not copying the data. The result is I'm getting "data is not aligned" warnings from swscaler. Is there a way to set the format context to only generate aligned data? [23:03] <matham> also, do you think that it'd be less costly to copy the image to align vs converting a unaligned image? [23:31] <BlackDream> If i get the following error [23:31] <BlackDream> [segment @ 0x3965580] Invalid DTS: 3600 PTS: -292950 in output stream 0:0 [segment @ 0x3965580] Encoder did not produce proper pts, making some up. [23:31] <BlackDream> is there any fix [23:32] <c_14> >Encoder did not produce proper pts, making some up <- that would be ffmpeg fixing it itself, though you can try adding -fflags +genpts if you want [23:33] <BlackDream> i have already -flags -global_header [23:33] <BlackDream> in my command [23:34] <BlackDream> if i remove it , it works [23:34] <BlackDream> xmm [23:34] <BlackDream> since i 'm building an automated program for me is there any fix with a static command concerning flags? [23:34] <BlackDream> i found out that some streams wants -flags -global_header [23:35] <BlackDream> and this one i am trying needs +genpts [23:35] <BlackDream> is there something that i can use for both and it will work? understand me? [23:36] <c_14> you could just try it with both active or implement a fallback of some sort [23:36] <BlackDream> -flags -global_header +genpts [23:36] <BlackDream> sometihng like that? [23:36] <c_14> -flags -global_header+genpts [23:37] <c_14> If that throws an error, try -flags -global_header -fflags +genpts [23:37] <BlackDream> no i get the same error [23:38] <BlackDream> [mpegts @ 0x243ff20] pts (-385740) < dts (3600) in stream 0 av_interleaved_write_frame(): Invalid argument [23:39] <c_14> you can try some of the other format flags, ie igndts or sortdts or discardcorrupt maybe. [23:39] <BlackDream> ok i will play a bit with them [23:39] <BlackDream> thanks for your support. appreciate it [00:00] --- Sun Jul 20 2014

1 0

ffmpeg-devel.log.20140718
by burek 19 Jul '14

19 Jul '14

[00:08] <jamrial> and dsputil is no more [00:11] <michaelni> and the ac3 encoder now needs the motion estimation context hooray [00:13] <kierank> why [00:14] <kierank> oh sad [00:20] <jamrial> and alpha is broken again [00:20] <jamrial> meh, super easy to fix. i'll send a patch later tonight if nobody fixes it by then [00:23] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:162cffca802d: avcodec/me_cmp: restore author attribution and copyrights [00:46] <wm4> is there code anywhere in ffmpeg that rewrites timestamps of a packet stream with timestamp resets to continuous timestamps? I suppose at least ffmpeg.c with remuxing must have something like this [00:58] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:c527c14d6323: avcodec/dvenc: clear all dsp contexts [01:00] <michaelni> wm4, yes there is discontinuity removial code in ffmpeg.c [01:01] <wm4> aaaaand I'm spotting ffmpeg.c using libavformat internals [01:01] <wm4> it accesses pts_wrap_bits, which is not part of the public api [01:02] <j-b> and once again proves my point that ffmpeg should be splitted from the libav* repo [01:02] <wm4> yeah, that would be a good idea [01:02] <wm4> also, remove all non-public fields from the public header files [01:03] <j-b> eat your own dog food [01:06] <wm4> from what I can see, ffmpeg.c considers the timestamp differences between all streams to guess whether there is a reset [01:06] <wm4> doesn't this fail hard if the stream is not perfectly interleaved? [01:25] <Timothy_Gu> saste: someone donated $1000 in April? [01:25] <Timothy_Gu> sweet! [01:29] <wm4> (and the default is 10 seconds timestamp difference, now that's approximate...) [01:30] <wm4> it would all be so simple if libdvdnav and libbluray wouldn't stink to hell [02:31] <michaelni> wm4, the code should work fine if its crazily mis-interleaved [02:42] <cone-149> ffmpeg.git 03Ben Avison 07master:5c22e8e4ad08: armv6: Accelerate ff_imdct_half for general case (mdct_bits != 6) [02:42] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:4c735ee0e3d4: Merge commit '5c22e8e4ad0852d61d5c4ba8d67d33fd72339497' [03:12] <cone-149> ffmpeg.git 03Ben Avison 07master:87552d54d333: armv6: Accelerate ff_fft_calc for general case (nbits != 4) [03:12] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:c27adb37ef5b: Merge commit '87552d54d3337c3241e8a9e1a05df16eaa821496' [03:43] <cone-149> ffmpeg.git 03Ben Avison 07master:5edad2c4a1f4: armv6: Accelerate vector_fmul_window [03:44] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:7809c2600612: Merge commit '5edad2c4a1f46bcc56be755af86ab355c2f1b37f' [03:50] <cone-149> ffmpeg.git 03Ben Avison 07master:5a272190a046: armv6: Accelerate butterflies_float [03:50] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:c67cf84628bc: Merge commit '5a272190a04666f0fe41be767396b30712638c21' [04:53] <cone-149> ffmpeg.git 03Timothy Gu 07master:b64808e9ddc8: kerndeint: remove dead assignments [05:14] <cone-149> ffmpeg.git 03Timothy Gu 07master:cd3c85ac5cb2: a64multienc: remove dead assignment [05:14] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:d5ba19e611b0: avcodec/a64multienc use av_freep() [05:42] <cone-149> ffmpeg.git 03James Almer 07master:95a2d705cd98: alpha/idctdsp: move disabled cruft out of dsputil [05:42] <cone-149> ffmpeg.git 03James Almer 07master:277e5ca6175e: alpha/me_cmp: move code out of dsputil [11:28] <ubitux> ePirat: speaking of this, no other way of pushing to icecast? [11:28] <ubitux> like, icecast isn't supporting some http pushing? [11:28] <ePirat> http PUT is what should be used [11:28] <ePirat> at least with 2.4+ [11:28] <ubitux> and it doesn't work? [11:29] <ePirat> it works but ffmpeg can't do it [11:29] <ubitux> huh [11:29] <ePirat> it only does POST [11:29] <ubitux> ok [11:30] <ubitux> iirc i was able with chunked_post to push to some kind of icecast-like [11:30] <ubitux> but it wasn't icecast itself [11:32] <ubitux> it's kind of surprising if it doesn't support this [12:53] <cone-719> ffmpeg.git 03Martin Storsjö 07master:fe67f3fbb5f9: arm: cosmetics: Fix a misaligned asm operand [12:53] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:cd4497d8c5c4: Merge commit 'fe67f3fbb5f9f6a6b60f837f6bc5e087ac11f3bf' [13:31] <cone-719> ffmpeg.git 03Martin Storsjö 07master:7e18a727d2c2: arm: cosmetics: Consistently use lowercase for shift operators [13:31] <cone-719> ffmpeg.git 03Michael Niedermayer 07master:d13effb0b4d8: Merge commit '7e18a727d2c2a19f22fcf68875d1b05fd2eafcef' [13:35] <cone-719> ffmpeg.git 03Timothy Gu 07master:0035783488cc: smacker: remove dead code [14:24] <cone-719> ffmpeg.git 03Andrey Utkin 07master:cb8cb85da791: lavfi/drawtext: add alias "expr_int_format" to expansion function "eif" [16:13] <berzerka> hi there [16:14] <berzerka> i was looking at the git log to see when the avcodec_alloc_frame deprecation took place [16:14] <berzerka> the commit was: 2013-12-11 - 29c83d2 / b9fb59d,409a143 / 9431356,44967ab / d7b3ee9 - lavc 55.45.101 / 55.28.1 - avcodec.h [16:14] <berzerka> what do the two version numbers tell me? in which version was the change introduced, former, latter, both? [16:16] <berzerka> and how would i properly compile-time test for this in another library? [16:22] <Daemon404> one is the libav # and one is the ffmpeg # [16:22] <Daemon404> the .101 is the ffmpeg one and the version you'd check for ffmpeg [16:25] <berzerka> ok so i would check against >=libavcodec-55.28 [16:26] <berzerka> what's the proper #if macro to do so? [16:27] <berzerka> and how do i interpret the commits where there's only one version shown? does it mean the commit does not affect ffmpeg, yet? [16:29] <berzerka> oh looks like this project is including ffmpeg and not libavcodec after all [16:29] <berzerka> so how would i check against >=ffmpeg-55.45.101? [16:31] <wm4> ffmpeg versions start from x.y.100 [16:31] <wm4> while libav starts from x.y.0 [16:32] <wm4> it's stupid, but this is how it is [16:32] <berzerka> ok so i would only check against >=55.45 [16:33] <wm4> no, ffmpeg and libav version numbers are not aligned [16:33] <wm4> so you have to check for them separately [16:34] <berzerka> the project i'm dealing with with only uses the libavcodec version which comes which ffmpeg, so i think i only need to check for the ffmpeg version? [16:34] <berzerka> which comes with* [16:35] <berzerka> otoh, avcodec.h is part of libavcodec, so i would need to check only for that one? i'm confused. [16:37] <wm4> that should be sufficient [16:38] <berzerka> ok thanks! still, how do i interpret the commits which only show one version number. e.g.: 2013-07-23 - abc8110 - lavc 55.19.100 - avcodec.h [16:39] <berzerka> from the version number i guess the commit only affects only ffmpeg. however, avcodec.h was changeg, which is clearly part of libavcodec? [16:39] <berzerka> my god s/only// [16:39] <berzerka> changed* [17:15] <berzerka> whatever, have a nice weekend everyone! [18:34] <jamrial> libav is planning a release in a couple weeks. what are we going to do? [18:34] <jamrial> rush an ffmpeg 2.4/3.0 out of the door, break protocol and add non-abi-breaking features to 2.3.1 so it's in sync with libav11, or do nothing at all? [18:41] <ubitux> a couple of weeks = probably about 3 months, that will probably match 2.4 schedule [18:41] <jamrial> lol [18:54] <nevcairiel> Yeah they aren't exactly fast at releasing stuff [19:05] <jamrial> well, lu did mention he was targeting august, which is why i asked [19:06] <cone-444> ffmpeg.git 03Carl Eugen Hoyos 07master:5dcd43cd8666: Fix make checkheaders for libavcodec/proresdsp.h. [19:06] <nevcairiel> They will have a beta again and it'll all take months [19:08] <jamrial> Ah true [19:19] <ubitux> jamrial: he also said "but I'd start the first working prototype of avscale this week" in march [19:19] <Daemon404> lol avscale [19:19] <Daemon404> lol. [19:20] <kierank> good luck with that [19:20] <iive> ubitux: well, he said "start" not "finish" [19:46] <cone-444> ffmpeg.git 03Peter B 07master:0f7013108998: fate: Add test for wav Peak Envelope Chunk encoder (levl chunk). [21:29] <michaelni> mraulet, smarter, is "0718 17:41 Christophe Gisq (3.4K) [FFmpeg-devel] [PATCH] [RFC] hevc: report more precise progress" ok [21:29] <michaelni> ? [21:33] <mraulet> I am looking into it right now [21:36] <cone-444> ffmpeg.git 03Diego Biurrun 07master:cb44b21da1f5: dct-test: Move cpu_flags variable out of global scope [21:36] <cone-444> ffmpeg.git 03Michael Niedermayer 07master:2dbda9378c1e: Merge commit 'cb44b21da1f59923be577f08c267ec270529be97' [21:43] <cone-444> ffmpeg.git 03Diego Biurrun 07master:746ad4e0df7f: dct-test: Improve CPU flags struct member name [21:43] <cone-444> ffmpeg.git 03Michael Niedermayer 07master:29119aff6cc7: Merge commit '746ad4e0df7faf93329804e412ec53c1d929a75b' [21:48] <cone-444> ffmpeg.git 03Diego Biurrun 07master:24af1aa0f703: fft: Convert FFT/MDCT permutation type #defines to enums [21:48] <cone-444> ffmpeg.git 03Michael Niedermayer 07master:cff61d595e45: Merge commit '24af1aa0f70362a66cda04c9d7cd012e019f5572' [21:53] <cone-444> ffmpeg.git 03Diego Biurrun 07master:c0de9159a7ba: avdevice: Give names to anonymously typedeffed structs [21:53] <cone-444> ffmpeg.git 03Michael Niedermayer 07master:f6ed5df9c00c: Merge commit 'c0de9159a7ba5707aa0a5c2bc73ae78b7b87ec46' [22:14] <cone-444> ffmpeg.git 03Diego Biurrun 07master:b4987f72197e: idct: Convert IDCT permutation #defines to an enum [22:14] <cone-444> ffmpeg.git 03Michael Niedermayer 07master:42d326353c17: Merge commit 'b4987f72197e0c62cf2633bf835a9c32d2a445ae' [22:16] <mraulet> michaelni: patch partially dropped after a discussion with the author [22:17] <michaelni> mraulet, ok, if theres something i should apply/cherry pick ... [22:18] <mraulet> yes the sao stuff [22:18] <mraulet> 10fc29fc19a12c4d8168fbe1a954b76386db12d0 [22:24] <cone-444> ffmpeg.git 03Diego Biurrun 07master:8b0dd4942aac: idctdsp: prettyprinting cosmetics [22:24] <cone-444> ffmpeg.git 03Michael Niedermayer 07master:521f5697348d: Merge commit '8b0dd4942aac320d1ca3c40fa7ea1be342c71273' [22:25] <Timothy_Gu> michaelni: when cherry-picking please put the "(cherry-picked from" a separate line [22:26] <mraulet> michaelni: bitstreams for fate-hevc [22:26] <mraulet> https://dl.dropboxusercontent.com/u/588600/rext_biststreams.zip [22:26] <mraulet> for rext [22:26] <michaelni> Timothy_Gu, ok will do [22:27] <mraulet> let me know where they are in I will update the tests [22:28] <kurosu> 10fc29fc19a12c4d8168fbe1a954b76386db12d0 <- related to sao, but not to the patch I submited [22:28] <mraulet> yes [22:32] <cone-444> ffmpeg.git 03Nidhi Makhijani 07master:b4edbe9a0eb4: libgsm: Split decoder and encoder [22:32] <cone-444> ffmpeg.git 03Michael Niedermayer 07master:4b0fa45b842c: Merge commit 'b4edbe9a0eb48423fbf249b0ca7fb62838bdb7f5' [22:39] <cone-444> ffmpeg.git 03Diego Biurrun 07master:9e0b29911f1f: x86: dnxhdenc: Eliminate some unnecessary ifdefs [22:39] <cone-444> ffmpeg.git 03Michael Niedermayer 07master:097bf834bacd: Merge commit '9e0b29911f1f167381a7dbdfca68bf417b8c767b' [22:44] <michaelni> mraulet, rext uploaded [22:44] <mraulet> ok [22:45] <mraulet> I have them [22:45] <cone-444> ffmpeg.git 03Diego Biurrun 07master:913fa85a2c1f: dct-test: Skip indirection for MMX IDCT permutation [22:45] <cone-444> ffmpeg.git 03Michael Niedermayer 07master:2897447ddf80: Merge commit '913fa85a2c1fcd71802f7decdef5f355e0d9c757' [23:04] <cone-444> ffmpeg.git 03Mickaël Raulet 07master:d249e6828e8a: hevc/sao: optimze sao implementation [23:04] <cone-444> ffmpeg.git 03Michael Niedermayer 07master:ffa0b510cd15: avcodec/hevc: fix pict_type [23:04] <mraulet> michaelni: for rext: with fate I will have to put pixfmt for each bitstream? [23:05] <michaelni> fate must pass on all archs, big endian and little, not sure but i would guess that means choosing one endianness and forcing it [23:06] <mraulet> why do we need it for [23:06] <mraulet> ok [23:06] <mraulet> forget my last comment [23:06] <mraulet> a bit painfull for rext then [23:40] <mraulet> michaelni: tests for hevc rext https://github.com/OpenHEVC/FFmpeg/compare/10fc29fc19...e6ee89e6fa [23:40] <cone-444> ffmpeg.git 03Carl Eugen Hoyos 07master:0e70266bbfad: Write valid free atom when -moov_size is specified. [23:40] <cone-444> ffmpeg.git 03Carl Eugen Hoyos 07master:ee88a2080ef0: Parse dri when receiving jpg via rtp. [00:00] --- Sat Jul 19 2014

1 0

ffmpeg.log.20140718
by burek 19 Jul '14

19 Jul '14

[00:49] <acuozzo> I'm working with 10-bit V210 (YUV422P10LE) material and my goal is to compress to 8-bit Huffyuv (YUV422P) for post-work. [00:49] <acuozzo> Two questions: [00:50] <acuozzo> 1. Do YUV422P10LE and YUV422P have the same luma range? (64-940 and 16-235, respectively) [01:02] <acuozzo> Is there some way I can extend the luma range before dropping to 8-bit with FFmpeg? [01:03] <acuozzo> I want to scale from (64, 888) -> (64, 940) and clamp at (64, 940) [10:20] <waressearcher2> when encoding I have many these strings: "frame= 975 fps=121 q=2.0 size=3472kB time=00:00:39.00 bitrate= 729.2kbits/s" now what is "q=2.0" ? [10:21] <waressearcher2> anyone ? [10:41] Last message repeated 1 time(s). [10:51] <waressearcher2> anyone [10:52] <Riviera> "quality" [10:52] <Riviera> man ffmpeg, search for "q=" [11:04] <Mavrik> it's quantizer actually, but yeah, "quality" is close enough explanation :) [11:09] <waressearcher2> why its changing, some lines with "q=1.6" some with "q=2.0" ? [11:15] <Riviera> Mavrik: ah :) [11:18] <Mavrik> waressearcher2, because quantizer is calculated practically on per-frame basis [11:18] <Mavrik> it's how internal encoder controls quality and bitrate [11:19] <waressearcher2> there is that filter "smartblur" it supports "enable" option, like: -vf smartblur=enable='between(t,10,3*60)' so some parts of video can be blurred, is it possible to do "enable" and "disable" for other filters in a similar way ? say I have that filter: -vf "setpts=1.8*PTS" but it speds up entire video and I want to speed up some parts of the [11:19] <waressearcher2> another question [11:19] <waressearcher2> video and slow down others, is it possible to use "enable" and "disable" for "setpts" ? [11:30] <termos> Is there a way using ffmpeg audio filters to set the volume for each channel? [11:31] <termos> Say there are two audio channels and I'd like to mute the left one [11:43] <OnkelTem> Hi all [11:44] <Mavrik> termos, hmm, you should check the list of filters [11:45] <Mavrik> but as it seems now you'll have to split audio, remove one channel and use only the righ tone [11:45] <OnkelTem> I use ffmpeg to convert RTSP stream to RTMP from inside nginx's rtmp module. For some reason ffmpeg process exists randomly. More info and logs: https://github.com/arut/nginx-rtmp-module/issues/436 Any ideas? [11:45] <OnkelTem> Or how can I get more information about why this happens? [11:45] <termos> hmm okey, I noticed some channelsplit filter that I might be able to use [11:46] <OnkelTem> ffmpeg version: N-63893-gc69defd [11:50] <OnkelTem> I'm not sure what does this mean but there is 17 (SIGCHLD) in logs [11:57] <FrEaKmAn_> hi all.. I'm confused how to use concat [11:57] <FrEaKmAn_> when I load files from .txt it works [11:57] <FrEaKmAn_> but when I can ffmpeg -i concat:"file1.mp4|file2.mp4" -c copy output.mp4 [11:57] <FrEaKmAn_> it concats just 1 file [11:57] <FrEaKmAn_> "concat:file1.mp4|file2.mp4" also doesnt work [11:58] <FrEaKmAn_> when I call* [11:59] <cuba> OnkelTem: you should log ffmpegs output to a file [11:59] <cuba> and check whats up with it [11:59] <Mavrik> FrEaKmAn_, pipe and quotes are usually special chars in shell [11:59] <Mavrik> so you probably have to escape them [12:00] <FrEaKmAn_> really? [12:00] <cuba> or use single quote ' [12:01] <FrEaKmAn_> hm [12:01] <cuba> your command looks right tho FrEaKmAn_ [12:01] <cuba> ah no [12:02] <cuba> try ffmpeg -i "concat:file1.mp4|file2.mp4" -c copy output.mp4 [12:02] <FrEaKmAn_> no.. just uses first file [12:02] <FrEaKmAn_> checking the docs online, this is the right command [12:02] <FrEaKmAn_> so no idea [12:03] <sacarasc> IIRC, MP4 can't be concatted like that, you have to use the concat filter... [12:03] <FrEaKmAn_> but when using files it works [12:03] <FrEaKmAn_> so probably it's not format problem [12:04] <cuba> Instructions [12:04] <cuba> ffmpeg -i "concat:input1.mpg|input2.mpg|input3.mpg" -c copy output.mpg [12:04] <cuba> ... [12:06] <FrEaKmAn_> nop.. I really dont understand why [12:07] <FrEaKmAn_> can someone please try on your computer? [16:15] <alonsovb> Hi, I'm trying to set up a video streaming server with ffmpeg. The idea is to stream from a Linux server running ffmpeg towards a Windows Server running IIS Media Services. I want to be able to use IIS Publishing Points, via the Push method. Could you give me an example of would I set the ffmpeg command? [16:16] <Fjorgynn_> fish [16:17] <Fjorgynn_> no [16:17] <Fjorgynn> but stay here for a few min or an hour [16:17] <alonsovb> Sure, I can even stay for 8 hours [16:18] <Fjorgynn> good. Most people leave after 2 min ;) [16:50] <OnkelTem> How to get latest ffplay? Why it is not in ffmpeg static downloads? [16:53] <sfan5> it's probably not included because it requires SDL [16:53] <OnkelTem> ffmpeg question. How to disable audio on fly when using ffmpeg as stream converter from rtsp to rtmp for example? [16:54] <c_14> -an [16:57] <OnkelTem> c_14: strange, it didn't help - when I view the stream using cmdline vlc rtmp://neganov.ru:1935/full/go VLC claims there is second stream [16:58] <OnkelTem> Type: Audio, Codec: undf [17:01] <OnkelTem> why it leaves Audio stream? This is my command line: [17:02] <Freakshow> what about ffprobe against the input, what does that give you? [17:02] <OnkelTem> ffmpeg -i rtsp://185.8.127.10:554 -an -c copy -f flv rtmp://localhost:1935/full/go [17:02] <c_14> I'm not sure, but the rtmp spec might require an audio track so ffmpeg creates an empty one, or maybe nginx is creating it. [17:02] <OnkelTem> Freakshow: let me check, I've never used ffprobe :) [17:03] <Freakshow> I suspect it will tell you the same thing, I don't think that the rtmp spec requres an audio track [17:03] <OnkelTem> rtmp://neganov.ru:1935/full/go [17:03] <OnkelTem> this is the stream [17:03] <Freakshow> we spin rtmp feeds all day without audio [17:03] Action: OnkelTem really loves ffmpeg and the tools [17:04] <c_14> OnkelTem: how bout if you output to a file, does that file then have an audio track? [17:05] <OnkelTem> (trying to realize what ffprobe printed) [17:06] <OnkelTem> ffprobe sees 2 streams [17:06] <OnkelTem> Stream #0:1: Data: none - the second [17:07] <OnkelTem> So I think VLC just goes crazy a bit, misunderstanding what it gets [17:07] <OnkelTem> Probably (just a suggestion) ffmpeg gets two streams of which the second is empty and just retranslates them? [17:08] <OnkelTem> Which means the second stream is just empty... FLV spec? [17:09] <OnkelTem> Nah, initial RTSP has exactly one stream [17:16] <OnkelTem> Erm.. how to make ffprobe to not exist and continue monitor the stream, as I suspect there are problems with it [17:16] <OnkelTem> honestly I read ffprobe man but can see appropriate option [17:17] <c_14> If you mean exit you can't, but you can do something like ffmpeg -i stream -c copy -f null /dev/null [17:17] <c_14> That way ffmpeg should report any errors in the source. [17:18] <OnkelTem> Ah, I see, thanks [17:19] <alonsovb> Hi, I'm trying to set up a video streaming server with ffmpeg. The idea is to stream from a Linux server running ffmpeg towards a Windows Server running IIS Media Services. I want to be able to use IIS Publishing Points, via the Push method. Could you give me an example of would I set the ffmpeg command? [17:20] <OnkelTem> There are huge number of messages like: [null @ 0x3203780] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 759600 >= 733500 [17:24] <c_14> alonsovb: https://ffmpeg.org/ffmpeg-formats.html#mov_002c-mp4_002c-ismv check the example, might be what you want. [17:26] <c_14> OnkelTem: that usually isn't that big an issue, it just means that the timestamps are out of order or broken [17:26] <OnkelTem> Hm, what is the thing called ffserver? Do I need it instead of ffmpeg? [17:26] <OnkelTem> c_14: can't I fix this when converting rtsp to rtmp using ffmpeg? [17:26] <alonsovb> Thank you <c_14>. The server I'm using is password protected. How should I send the username and password through ffmpeg? [17:27] <c_14> alonsovb: you can try http://username:password@server [17:28] <alonsovb> awesome, I'll try that, thanks again c_14 [17:28] <c_14> OnkelTem: I'm pretty sure FFmpeg will fix it as much as it can automatically, but sometimes the only fix is to drop the packets because it doesn't know where to put them. [17:28] <OnkelTem> c_14: yes, then let them be dropped. You see I get them *after* ffmpeg [17:29] <c_14> so you get them in the rtmp stream not the ffmpeg rtsp -> rtmp conversion? [17:30] <OnkelTem> c_14: exactly [17:30] <OnkelTem> ffmpeg -i rtmp://neganov.ru:1935/full/go -c copy -f null /dev/null [17:30] <OnkelTem> not at the start, later [17:32] <c_14> but the conversion process isn't complaining? [17:32] <c_14> If the conversion process were also complaining I might know why. [17:34] <OnkelTem> c_14: hm, how can I redirect output in some file? I mean during conversion? [17:34] <OnkelTem> console output [17:36] <c_14> ffmpeg [command] &>file [17:47] <OnkelTem> c_14: I'm not sure, but I can't use &>file in nginx.conf [17:47] <OnkelTem> with nginx rtmp, like: exec_pull ffmpeg .... &>file; [17:48] <c_14> add -report to the ffmpeg commandline [17:48] <c_14> It'll create a program-YYYYMMDD-HHMMSS.log in the current dir. [17:49] <c_14> (you'll have to know where nginx is executing ffmpeg) [17:50] <OnkelTem> ok nice [18:01] <OnkelTem> Ok, I got the output: https://gist.github.com/OnkelTem/f0ce9ed7dc414df160b4 [18:01] <OnkelTem> There are another problem which I was talking some hours ago [18:01] <OnkelTem> as you can see stream is deleted! [18:02] <OnkelTem> I don't know why this happens, and looks like this occures only when ffmpeg is launched by nginx [18:02] <OnkelTem> when I run ffmpeg from command line, it doesn't exist [18:05] <OnkelTem> OMG [18:05] <OnkelTem> Sorry, I stupid [18:05] <OnkelTem> I suddenly noticed that it is another version, old 0.10 [18:05] <OnkelTem> But HOW [18:05] <c_14> do you have an old version of ffmpeg in your path somewher? [18:05] <c_14> +e [18:06] <OnkelTem> yes! but: which ffmpeg ---> /usr/local/bin/ffmpeg [18:06] <c_14> Maybe nginx is using a different path? [18:06] <OnkelTem> sudo -u www-data which ffmpeg ---> /usr/local/bin/ffmpeg [18:06] <c_14> hmm, you could just give nginx the path to the binary [18:06] <OnkelTem> I didn't specified any path. Ok, I'll write absolute [18:09] <OnkelTem> c_14: thank you very much! this explains everything :) with -report I could catch this crap [18:09] <OnkelTem> so the video was laggy with outdated 0.10.x [18:10] <OnkelTem> now I expect everything to change [18:10] Action: OnkelTem is dancing [18:30] <Diogo> hi this is possible to stream to rtmp server but using a multiple input ... if one of the stream stop receiving pass to second stream without change the final content.. [18:31] <sfan5> https://www.ffmpeg.org/ffmpeg-filters.html#concat could be what you're looking for [18:34] <Freakshow> can anyone think of a reason I wouldn't be able to grab from a multicast source when that network has been added through route add on centos? [18:34] <Freakshow> this is correct isn't it: [18:34] <Freakshow> ffmpeg -i "udp://230.1.1.27:9000?fifo_size=4915200" [18:34] <Diogo> sfan5: ffmpeg -i rtmp://streamA -i rtmp://streamB -filter_complex '[0:0] [0:1] [0:2] [1:0] [1:1] [1:2] [2:0] [2:1] [2:2] concat=n=3:v=1:a=2 [v] [a1] [a2]' -map '[v]' -map '[a1]' -map '[a2]' -f flv rtmp://finalSTREAM [18:34] <Diogo> something like this sfan5 ? [18:34] <Diogo> using filter_complex this is possible to to the fallback? [18:35] <sfan5> looks right [18:35] <sfan5> you may want to specify encoders and encoder arguments though' [21:11] <optix2> I'm trying to burn in subtitles, is the AlphaLevel setting in ssa format not supported? I get solid black irrespective of the value [21:23] <optix2> http://pastebin.com/izq4xBjJ [21:24] <optix2> here is the important part of the ut.ssa http://pastebin.com/qgAzgWcs [21:25] <optix2> so my question: does ffmpeg support AlphaLevel of ssa subtitles or is that value ignored? [21:40] <alonsovb> What reasons could cause IIS Media Services to not recognize a ffmpeg stream? I'm using all these options: -movflags isml+frag_keyframe -f ismv -threads 0 -c:a libfaac -ac 2 -b:a 64k -c:v libx264 -preset fast -profile:v baseline -g 48 -keyint_min 48 [21:41] <Hello71> and we can't really support closed-source software [21:42] <sfan5> optix2: did you try -vf ass=<subfile> ? [21:42] <alonsovb> sorry, here it is my full command http://pastebin.com/FfpStGMK [21:42] <optix2> not yet - wait I'll try [21:43] <c_14> alonsovb: the output would also be helpful so we can see what is happening [21:43] <alonsovb> the output is the usual output when ffmpeg is working, there are no errors [21:44] <alonsovb> the only problem is that the stream is not being recognized in IIS [21:44] <c_14> So FFmpeg thinks it's outputting to the http server correctly? ie no http errors etc? [21:44] <alonsovb> it may be an ISS problem, but I just wanted to make sure I don't have any errors on the ffmpeg side [21:44] <alonsovb> yes, there are no errors whatsoever [21:47] <optix2> sfan5: no doesnt work - it does work in media player classic though, so the value is read by some software [21:47] <c_14> alonsovb: Then I'd blame ISS. [22:43] <jdolan> why would avfilter_graph_parse2 complain about parsing the string "null; anull" ? [22:44] <jdolan> No such filter: '' [23:24] <Einewton> Hi [23:24] <Einewton> I have a question about overlays [23:25] <Einewton> I've sucessfully followed the instructions to do overlays the way I want it [23:25] <Einewton> https://trac.ffmpeg.org/wiki/Create%20a%20mosaic%20out%20of%20several%20inp… [23:25] <Einewton> I cannot find a way to toggle on/off the overlays during an encode process. [23:25] <Einewton> While it's processing to turn on/off a certian overlay. [23:26] <Einewton> Can anyone help? [23:27] <Einewton> Can I send a command to a specific fiter while it's encoding to change it's position off the screen or something like that? [00:00] --- Sat Jul 19 2014

1 0

ffmpeg-devel.log.20140717
by burek 18 Jul '14

18 Jul '14

[02:45] <BBB> oh thats true, I saw that too, the mp4toannexb bsf crash [02:45] <BBB> I forgot to report that [02:45] <BBB> I think it crashes if you forget -vcodec copy or something like that, I forgot what exactly triggers it [02:46] <BBB> ubitux: also want to implement skip_* in vp9, and skip_idct in vp8? [02:46] <BBB> ubitux: I think vp8 does skip_loopfilter etc. but I forgot the rest [03:28] <cone-68> ffmpeg.git 03Simon Thelen 07master:fa6716c66d31: libavformat/segment: change segment_list_size behavior to match hls_list_size behavior. [06:23] <cone-68> ffmpeg.git 03Michael Niedermayer 07master:7cd6d61da53c: avutil/opt: try to fix the case where src==dst in av_opt_copy() [09:12] <ubitux> BBB: another time, i'm mostly interested in having that feature in h264 so far [09:15] <ubitux> projectsymphony is koda? [13:27] <cone-149> ffmpeg.git 03Stefano Sabatini 07master:f3e886c7dfcc: lavu/eval: add clip function [13:27] <cone-149> ffmpeg.git 03Andrey Utkin 07master:dec87454deb2: lavfi/drawtext: add expansion function eif() [13:27] <cone-149> ffmpeg.git 03Andrey Utkin 07master:e9ff5df0b530: lavfi/drawtext: add fontcolor_expr option [13:27] <cone-149> ffmpeg.git 03Stefano Sabatini 07master:3babbf3bf4c6: doc/filters/drawtext: simplify example, use clip() eval function [13:54] <cone-149> ffmpeg.git 03Diego Biurrun 07master:a11ef610eec5: h264: Remove some commented-out, broken cruft [13:54] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:b884599f6a1c: Merge commit 'a11ef610eec5241e226efbd1c5bb5eb45c88d2c2' [14:10] <cone-149> ffmpeg.git 03Luca Barbato 07master:2db953f84671: h264: K&R formatting cosmetics [14:10] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:5fce29ef29e4: Merge commit '2db953f84671997e936f91140ffb5143c1537844' [14:16] <ePirat> Can I somehow use PUT for sending to HTTP with ffmpeg? The docs only mention POST. [14:28] <thardin> it used to be possible [14:28] <thardin> check ffmpeg --help [14:28] <thardin> in the http section [14:31] <cone-149> ffmpeg.git 03Nidhi Makhijani 07master:324ff59444ff: avpacket: Check for and return errors in ff_interleave_add_packet() [14:31] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:e5cb7795fcaf: Merge commit '324ff59444ff5470bb325ff1e2be7c4b054fc944' [14:32] <ePirat> thardin, there is no option that permits that, everything is just for POST [14:32] <thardin> huh [14:34] <thardin> anyway, just pipe to curl [14:34] <ePirat> doing this currently but having issues with laggy video [14:34] <ePirat> thardin, thats how I am doing currently https://gist.github.com/anonymous/0cf50b64b6b5398b12e3 [14:35] <thardin> looks decent [14:36] <ePirat> thardin, any idea what could cause the lagging? [14:37] <ePirat> seems like dropping a lot of frames but ffmpeg always displays 25fps so it should work& [14:38] <thardin> try piping to a file and see how that works [14:38] <thardin> ffmpeg ... - > foo.ogg [14:40] <ePirat> this seems to work fine. [14:41] <thardin> and if you curl the file? [14:43] <ePirat> I curl it to icecast (recent version has HTTP PUT support) and I get lagging video when playlig the stream. (video stops and continues later on, while audio plays along fine) [14:44] <ePirat> (I can't just cat it, since icecast needs to receive the video with proper timing) [14:44] <thardin> this sounds like possible ogg horribleness [14:45] <thardin> try another container, like mpegts [14:45] <ePirat> hmm not sure if icecast supports mpegts, I'll try webm& but ogg should work, at least it works with vlc [14:46] <ePirat> (vlc on windows can stream to icecast) [14:46] <thardin> ogg muxer in ffmpeg may be incorrect [14:47] <thardin> because ogg is a nasty format [14:48] <ePirat> nasty? [14:49] <thardin> iirc timestamping is a mess [15:06] <Daemon404> someone should make a bot that just spams mans' blog every time ogg comes up [15:15] <ubitux> only the dc version of the idct are written in hevc asm? [15:23] <ubitux> saste: ping [15:23] <saste> ubitux, pong [15:23] <ubitux> saste: you worked on the segmenter, how feasable do you think it would be to define a segmentation list file? [15:24] <saste> ubitux, do you mean a file listing the segmentation points? [15:24] <ubitux> my goal being to construct such list using the scene detection, and use the segmenter to split the file that way [15:25] <ubitux> saste: yes [15:25] <saste> ubitux, we already can provide a list of splitting points, through the commandline [15:25] <saste> -segment_times T1,T2,... [15:25] <ubitux> mmh yeah i could build that [15:25] <ubitux> i guess [15:34] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:88f038ac97a8: avformat/dv: implement fallback in dv_extract_pack() [16:06] <Daemon404> https://www.dropbox.com/s/mpqwofcfpotlnbx/t.png <-- it's trying so hard to give me targeted ads [16:06] <Daemon404> but just slightly failing [16:09] <ubitux> >opus [16:09] <ubitux> ? [16:13] <ubitux> cool stuff of the day with ffmpeg: [16:13] <ubitux> ffprobe -v error -f lavfi "movie=$HOME/samples/thelondonfeeling.mp4, select='gt(scene,0.25)'" -select_streams v:0 -show_entries frame=best_effort_timestamp_time -of csv | cut -d, -f2 > /tmp/video.cut [16:13] <ubitux> ffmpeg -i ~/samples/thelondonfeeling.mp4 -map 0:v -c:v copy -f segment -segment_times $(xargs --arg-file=/tmp/video.cut | tr ' ' ',') /tmp/out-%03d.mp4 [16:54] <cone-149> ffmpeg.git 03Stefano Sabatini 07master:5acad50056f6: lavf/segment: only use reference frames for computing the segment end time [16:54] <cone-149> ffmpeg.git 03Stefano Sabatini 07master:b6a8c5245f5c: lavf/segment: show incoming packet duration in debug message [16:54] <cone-149> ffmpeg.git 03Stefano Sabatini 07master:3d38c9904121: lavf/segment: fix weird indent [16:54] <cone-149> ffmpeg.git 03Stefano Sabatini 07master:526bd7bdfad7: lavf/segment: set segment end time when the first packet arrives [17:06] <Compn> ubitux : whats so interesting about a fingerprint filter ? [17:06] <Compn> sounds like framemd5 to me [17:09] <ubitux> Compn: assuming it can handle different resolution & encode qualities, it can be used to detect similar content [17:11] <ubitux> but maybe that's something more stupid [17:11] <ubitux> ...haven't seen the code [17:14] <ePirat> is this https://www.ffmpeg.org/doxygen/trunk/avfoundation_8m_source.html outdated? [17:45] <Compn> ubitux : it sounded like copyright detection to me :P [17:46] <ubitux> that's one usage, but dups detection in whatever youtube-like service is useful [17:46] <ubitux> for copyright purpose or not [18:02] <kierank> Daemon404: mumble mumble vfr [18:02] <kierank> [FFmpeg-devel] Issue 3052 [18:05] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:f46e87602700: ffserver: use av_mallocz_array() [18:05] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:ad516dd271ff: avcodec/resample2: Use av_malloc(z)_array() [18:05] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:dca22ab055bd: avcodec/resample: Use av_m/realloc_array() [19:13] <ubitux> where is llogan? [19:14] <ubitux> so he can reply to https://twitter.com/ID_AA_Carmack/status/489813195373035520 with swscale [19:14] <ubitux> (thx Daemon404) [19:14] <ubitux> ;) [19:14] <Daemon404> "single file" [19:15] <ubitux> cat libswscale/*.c > scale.c [19:16] <Daemon404> "simple" [19:16] <Daemon404> >needs config [19:16] <Daemon404> also he means to statically link ofc [19:16] <Daemon404> so MIT licensed [19:16] <ubitux> :( [19:17] <Daemon404> game devs for you :P [19:17] <Daemon404> interestingly enough, the thing he talks about has a single-file MIT vorbis implementation [19:17] <Daemon404> seems pretty silly... libvorbis is MIT [19:18] <ubitux> maybe not single file [19:18] <Daemon404> game devs are concerned about an extra 40-100 kb in the exe? [19:18] <Daemon404> wolfenstein *language pack* alone is 14 gb iirc [19:18] <Daemon404> not the game. the language pack. [20:39] <cone-149> ffmpeg.git 03Nicolas George 07master:a65c0a3fe822: lavd/x11grab: disable drawing mouse without XFixes. [20:39] <cone-149> ffmpeg.git 03Nicolas George 07master:36fbe3c78974: lavd/x11grab: change error code for unsupported visuals. [20:39] <cone-149> ffmpeg.git 03Nicolas George 07master:16c6795465fd: lavd/x11grab: check 32-bits color masks. [20:39] <cone-149> ffmpeg.git 03Nicolas George 07master:1d12df1ad78c: lavd/x11grab: add an option to disable MIT-SHM. [20:39] <cone-149> ffmpeg.git 03Nicolas George 07master:099aff5c42ae: lavfi: check refcount before merging. [20:39] <cone-149> ffmpeg.git 03Nicolas George 07master:8e2976860ffc: lavd/x11grab: reindent after last commit. [22:19] <cone-149> ffmpeg.git 03Diego Biurrun 07master:c23ce454b3e3: x86: dsputil: Coalesce all init files [22:19] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:d6676a160575: Merge commit 'c23ce454b3e33634a188d6facfd2b7182af5af93' [22:26] <cone-149> ffmpeg.git 03Martin Storsjö 07master:a578b0407dc9: configure: Assume runtime cpu detection on arm on --target-os=android as well [22:26] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:6be71e995595: Merge commit 'a578b0407dc983aecd72028e1127062689b67089' [22:44] <saste> michaelni: https://co.clickandpledge.com/advanced/default.aspx?wid=56226 [22:45] <michaelni> saste, thats cool! [23:51] <cone-149> ffmpeg.git 03Diego Biurrun 07master:2d60444331fc: dsputil: Split motion estimation compare bits off into their own context [23:51] <cone-149> ffmpeg.git 03Michael Niedermayer 07master:3a2d1465c886: Merge commit '2d60444331fca1910510038dd3817bea885c2367' [00:00] --- Fri Jul 18 2014

1 0

ffmpeg.log.20140717
by burek 18 Jul '14

18 Jul '14

[00:04] <IamTrying> c_14: when i apply ffmpeg -f concat -i nl.txt -c copy test.flv its saying "Unknown input format concat" ffmpeg version 0.8.6 [00:05] <c_14> the concat demuxer was added in FFmpeg 1.1, grab a static binary from [00:06] <IamTrying> OK - Thank you [00:28] <Peter_Occ> I'm sending output from a cronjob to start ffmpeg to a log file. the cron is working and ffmpeg is working. The first time the log file is created, it looks fine. The next time the command runs, the log file becomes unreadable because it full of binary code. Is there something wrong with the way I'm sending the output? Here is the line from the crontab [00:28] <Peter_Occ> /2 * * * * /var/www/html/ffmpeg/start.sh>/var/www/html/ffmpeg/ffmpeg.log 2>&1 [00:52] <c_14> it looks fine. You could always try redirecting the output within start.sh directly instead of later in the cron line see if that helps. [00:55] <Peter_Occ> Ok I'll try that. [03:28] <ePirat> Can I use ffmpeg to output some input file for streaming? [03:28] <ePirat> need it to send to icecast, but icecast can't do timing itself so it needs receive it timed correctly [03:31] <relaxed> ePirat: ffmpeg -h full | less +/'-re ' [03:34] <ePirat> relaxed, thanks! [03:47] <rcombs> can I interface between multiple complex filter graphs? [03:48] <c_14> I'm pretty sure an ffmpeg command can only have one filter_complex [05:59] <laoshanlong> So I'm trying to figure out how to create a webm with embedded, hard subtitles. My source is a .MKV with .ass formatted subtitles. I extracted the .ass and tried the flag -vf "ass=sub.ass" but there were no hardsubs in my output. [06:31] <Dark-knight> just use mov_text [06:31] <waressearcher2> how do I fade in first second of output video ? if I speed up video using "-vf setpts=0.02*PTS,fade=in:0:24" it doesn't works because it takes 24 frames from original video but because it was speed up so much there is no fade in, but it kind of works if I use "fade=in:0:1000", so should I calculate amount of frames myself ? [06:32] <Dark-knight> if you only want hardsubs then nvm [06:33] <waressearcher2> other words if I encode 1 minutes video to 1 minutes video "fade=in:0:24" is working for fading in first second, but if I take 1 hour long video and speed it up to 1 minute long, "fade=in:0:24" doesn't work, I need something like "fade=in:0:1000", looks like that filter takes into account initial video on the output one [07:16] <waressearcher2> if I say have audio that is 2m55s long and video that is 2m58s long its will be allright to mix them ? ffmpeg will add silence at the end automatically ? [07:54] <laoshanlong> (Dark-knight): What's that? [07:54] <Dark-knight> nvm [07:54] <laoshanlong> Ah. [07:55] <laoshanlong> I attempted "ffmpeg -i dal.mkv -ss 04:37.021 -to 05:49.182 -c:v libvpx -b:v 2M -vf subtitles=out.ass -crf 4 -f webm out.webm" [07:55] <laoshanlong> but the subtitles are offset by 04:37:021 when I play it in MPC, and they don't appear at all in a web browser. [08:03] <relaxed> laoshanlong: what do you mean they don't up in your web browser? either they're hardsubbed or they're not. [08:03] <relaxed> dont show up* [08:06] <laoshanlong> They're not hardsubbed. I suspect MPC is auto-loading the external .ass file [08:07] <laoshanlong> Yep, that's exactly what's going on [08:07] <relaxed> laoshanlong: did you build ffmpeg? [08:07] <laoshanlong> It's not hardcoding anything [08:07] <laoshanlong> No. [08:08] <relaxed> Version? Where did you get it? [08:08] <laoshanlong> I suspect I got it from the download page on ffmpeg.org, Zeranoe FFmpeg Builds [08:09] <laoshanlong> folder name is ffmpeg-20140406-git-59c1023-win64-static [08:09] <laoshanlong> I got it ages ago. [08:09] <relaxed> try a newer build [08:10] <laoshanlong> downloading from: http://ffmpeg.zeranoe.com/builds/ [08:10] <relaxed> also, simply your command to see if it works at all- "ffmpeg -i dal.mkv -t 60 -vf subtitles=out.ass out.webm" [08:10] <relaxed> simplify* [08:15] <laoshanlong> running it [08:19] <laoshanlong> Nope [08:19] <laoshanlong> No hardsubs [08:23] <relaxed> pastebin the output [08:24] <laoshanlong> ? [08:24] <laoshanlong> The output is a video file [08:25] <relaxed> the console output from the ffmpeg command. [08:25] <waressearcher2> video inside console ? [08:25] <waressearcher2> ASCII video [08:25] <sacarasc> The text output from ffmpeg. [08:26] <laoshanlong> cmd line buffer wasn't long enough, one sec [08:26] <relaxed> I assume if MPC is using out.ass it's a valid file. [08:27] <laoshanlong> Hmm... [08:27] <laoshanlong> How do I pipe cmd line output to a text file? [08:28] <waressearcher2> laoshanlong: > textfile ? [08:28] <laoshanlong> Doesn't seem to have worked, waressearcher2 [08:28] <relaxed> I don't know how to using windows. [08:28] <relaxed> google knows [08:28] <waressearcher2> laoshanlong: "echo 111 > file" doen't create file with "111" in windows ? [08:29] <laoshanlong> it does [08:29] <laoshanlong> but running the ffmpeg command > log.txt doesn't pipe the text to log.txt [08:29] <relaxed> ffmpeg's output is stderr, not stdout [08:30] <relaxed> on *nix it would be ffmpeg ... 2>ffmpeg.log [08:30] <waressearcher2> laoshanlong: try to put it in 1.bat and then run "1.bat > file.log" [08:30] <laoshanlong> In order to redirect STDERR you have to specify "2>" for the redirection symbol. This selects the second output stream which is STDERR. [08:30] <waressearcher2> so looks like "2>" works in windows too [08:30] <laoshanlong> there we go [08:30] <laoshanlong> log.txt is doing stuff [08:32] <laoshanlong> http://pastebin.com/uz283kZf [08:32] <laoshanlong> the subtitle file is named out2.ass in order to prevent mpc from loading it [08:36] <relaxed> See this? --> Fontconfig error: Cannot load default config file [08:36] <laoshanlong> yeah [08:39] <relaxed> I guess that's the problem. [08:39] <relaxed> Zeranoe: ^^^ [08:40] <laoshanlong> Well, any idea what that error really means? [08:42] <Zeranoe> http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=10&t=318&start=20#p2463 [08:44] <laoshanlong> (Zeranoe): System or User variables? [08:57] <laoshanlong> success! [08:58] <Dark-knight> Hey Zeranoe! [09:19] <laoshanlong> ... [09:19] <laoshanlong> So [09:19] <laoshanlong> I now have an output file [09:20] <laoshanlong> and it plays just fine for me [09:20] <laoshanlong> but others say it's corrupt [09:20] <laoshanlong> http://a.pomf.se/oqkxzw.webm What the hell [09:20] <MrPingouin> hey people [09:20] <heftig-z> the ffmpegs in this pipe keep eating memory: ffmpeg -v quiet -f lavfi -i color -f nut -c:v copy - | ffmpeg -i - -f rawvideo -c:v copy -y /dev/null [09:20] <heftig-z> what's going on here? [09:21] <MrPingouin> despite my -pix_fmt yuv420p, I can't managed to play a h264 1080p movie on quicktime, is it a bandwidth issue ? [09:22] <MrPingouin> note that the pix_fmt yuv trick does work, as if I don't put this option, quicktime opens a "white image video" [09:23] <MrPingouin> sorry, not bandwidth, bitrate [09:28] <termos> what can I do to maintain constant segment size for HLS .ts files? I think the problem is varying gop due to scene cut detection? [09:29] <heftig-z> termos: try no-scenecut x264opt [09:30] <termos> ah thanks [09:48] <heftig-z> can I somehow disable ffmpeg's building of a stream index? it's useless to me and keeps eating memory [10:05] <relaxed> MrPingouin: see if adding -level 41 makes a difference. create a small sample with ffmpeg -i input -t 20 -level 41 .... [10:29] <MrPingouin> relaxed: I'm testing right now [10:30] <Roee> hi guys [10:30] <Roee> has anynoe failed to build ffmpeg with the following error: ERROR: librtmp not found [10:31] <Roee> I installed various versions of rtmp and nothing works :( [10:31] <relaxed> you're probably missing the dev package that includes the headers. [10:32] <relaxed> look at the bottom of config.log for the reason [10:32] <Roee> i586-poky-linux-gcc -Wl,-O1 -Wl,--hash-style=gnu -Wl,--as-needed --sysroot=/opt/clanton-full/1.4.2//sysroots/i586-poky-linux -Wl,--as-needed -o /tmp/ffconf.co8wH3bV /tmp/ffconf.IRLF576F.o -lm -lbz2 -lz -pthread check_pkg_config librtmp librtmp/rtmp.h RTMP_Socket false --exists --print-errors librtmp ERROR: librtmp not found [10:33] <relaxed> Roee: which distro? [10:33] <Roee> ubuntu [10:33] <Roee> let me just verify the version [10:33] <Roee> just a sec [10:33] <Roee> i think 12.04 [10:33] <relaxed> install librtmp-dev and try it again [10:34] <Roee> oops [10:34] <Roee> 13.1 [10:34] <Roee> 10* [10:35] <Roee> i586-poky-linux-gcc -Wl,-O1 -Wl,--hash-style=gnu -Wl,--as-needed --sysroot=/opt/clanton-full/1.4.2//sysroots/i586-poky-linux -Wl,--as-needed -o /tmp/ffconf.XIwS4CSg /tmp/ffconf.CDxsf621.o -lm -lbz2 -lz -pthread check_pkg_config librtmp librtmp/rtmp.h RTMP_Socket false --exists --print-errors librtmp ERROR: librtmp not found [10:35] <Roee> same error i believe [10:35] <Roee> after doing apt-get remove and than install [10:36] <relaxed> Do you even need librtmp? [10:36] <Roee> yes [10:36] <Roee> it's actually the only thing i need :| [10:37] <relaxed> pastebin.com the last couple of pages of your config.log [10:38] <Roee> http://pastebin.com/EJ7aHnR0 [10:39] <Roee> Do you wish to see the Makefile? [10:41] <relaxed> pastebin the whole log [10:42] <Roee> http://pastebin.com/qxG8duLr [10:42] <MrPingouin> relaxed: it's better a little bit ! still laggy though [10:44] <relaxed> does quicktime have multithreaded decoding? [10:46] <MrPingouin> I don't know, but the thing is [10:46] <MrPingouin> the same video encoded with adobe premiere runs perfectly [10:46] <relaxed> Roee: does it work if you simply run "./configure --enable-gpl --enable-librtmp" ? [10:47] <relaxed> MrPingouin: pastebin the output of ffmpeg -i [10:47] <relaxed> (and the video that's giving you trouble) [10:48] <MrPingouin> relaxed: http://bpaste.net/show/3se5gyuuK6qOdi3p4AT7/ [10:49] <Roee_> chrome stopped working for some reason [10:49] <Roee_> yasm/nasm not found or too old. Use --disable-yasm for a crippled build. [10:49] <relaxed> install yasm [10:50] <relaxed> MrPingouin: output.mov is what you encoded with ffmpeg? [10:51] <MrPingouin> yes [10:51] <MrPingouin> oh, you want the input ? [10:51] <relaxed> no [10:51] <Roee_> I think it worked [10:51] <Roee_> got a pretty huge output [10:51] <Roee_> pastebin? [10:52] <relaxed> MrPingouin: why does is say "major_brand : qt", are you sure you encoded this with ffmpeg? [10:52] <MrPingouin> eeeeer yes lol [10:52] <MrPingouin> here is the command line [10:53] <relaxed> Roee_: something in your original ./configure line is conflicting with librtmp support. [10:53] <MrPingouin> relaxed: http://bpaste.net/show/0hJfb0GIzqdoDbAbZf0C/ [10:54] <Roee_> How can I detect which one? [10:54] <relaxed> Roee_: start small [10:55] <relaxed> MrPingouin: That command looks fine. Does it play fine in vlc or mplayer? [10:56] <MrPingouin> in vlc it works perfectly, the issue is, I've been asked for it to play smoothly on quicktime :) [10:58] <Roee_> I'm using a specific toolchain designed for Intel Galileo [10:58] <Roee_> maybe this could be the issue [10:59] <relaxed> MrPingouin: add -tune fastdecode [11:00] <MrPingouin> lol [11:00] <relaxed> ? [11:00] <MrPingouin> I've been dealing with ffmpeg command line since 2 days, I'm the master of the command line ! [11:00] <MrPingouin> encoding a video with ffmpeg is like hacking the fbi in scifi movies :) [11:02] <MrPingouin> better ! [11:03] <MrPingouin> thanks a lot relaxed, I'm forwarding the result to my bosses [11:03] <MrPingouin> I keep you in touch :) [11:03] <relaxed> it could be that your pc is two slow to decode it with one thread. [11:03] <relaxed> too slow* [11:23] <muhammet> I am using Ubuntu and read https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu for compiling FFmpeg. In the output of one statically built FFmpeg executable the configuration parameters are present, so I managed to compile using those options but the resulting "ffmpeg" file is approximately 10M smaller. How is your procedure for creating a static build of FFmpeg? [11:25] <MrPingouin> relaxed: by the way, isn't it "-level 4.1" instead of "-level 41" ? [11:27] <JEEBsv> at least the latter should work, possibly the dotted one too [11:27] <JEEBsv> if the dotted one actually works, then feel free to use it [11:27] Action: JEEBsv prefers the non-dotted one for some reason [11:28] <relaxed> last time I tried dotted did not work [12:00] <FrEaKmAn_> hi.. can I execute 2 things at once? for example cut first file 00:05 - 00:10 AND cut second file 00:10 - 00:15? [12:01] <FrEaKmAn_> I trying to put everything in 1 command for I have total progress [13:50] <ePirat> I am using -re option to output live video with ffmpeg to stream it to icecast, but it is very laggy for some reason, are there more things I can do to optimize it? [13:51] <c_14> FrEaKmAn_: ffmpeg -ss 5 -t 5 -i file1 -ss 10 -t 5 -i file2 -map 0 out1 -map 1 out2; <- this should work [13:52] <FrEaKmAn_> c_14: nice.. let me try [13:52] <FrEaKmAn_> but to also add concat to this? possible [13:55] <ePirat> c_14, here is the full output https://gist.github.com/anonymous/0cf50b64b6b5398b12e3 [13:55] <c_14> FrEaKmAn_: ffmpeg -ss 5 -t 5 -i file1 -ss 10 -t 5 -i file2 -filter_complex '[0][1]concat=n=2:v=1:a=1[v][a]' -map '[v]' -map '[a]' out; assuming you want to concat the two streams (also you might need to specify the streams one by one. ie [0:0][0:1][1:0][1:1] etc [13:56] <FrEaKmAn_> nice [13:58] <c_14> ePirat: is the encoding too slow or what's the problem? [13:59] <ePirat> c_14, the video is lagging for some reason, like a lot missing frames but since ffmpeg always display 25fps not sure how this is possible [14:00] <ePirat> even happens when I use an ogg file and codec copy [14:00] <c_14> does it also happen if you take say an ogg file and cat it into curl? [14:00] <ePirat> thats not possible [14:00] <ePirat> it needs to have proper timing [14:01] <ePirat> icecast doesn't do any timing [14:01] <c_14> oh, right [14:01] <relaxed> ePirat: can you streamcopy the original file or does icecast require ogg? [14:01] <c_14> have you tried getting FFmpeg to output directly to the http stream? [14:02] <ePirat> c_14, no, how would I do that? [14:02] <ePirat> relaxed, I tried -c copy -f ogg with an ogg (video+audio) input file, same problem. [14:03] <c_14> ffmpeg -i [..] -f ogg http://localhost:8000/video [14:05] <ePirat> c_14, can I somehow specify the http method it should use? seems it does the wrong thing, I get error 400 (Bad request) [14:08] <c_14> hmmm, it seems your curl command is specifying a user/password. You can probably set that in FFmpeg with the headers option, but I don't know how. [14:09] <ePirat> I just put it in the url like http://user:pass@localhost:8000/video, ffmpeg doesn't complained about it so I expected it can handle it [14:10] <ePirat> but it seems ffmpeg uses POST instead of PUT [14:10] <ePirat> and the docs provide no info at all how to change it, seems impossible [14:13] <c_14> Yeah, hmm. The reason I asked was that I think either curl or the pipe might be buffering the input partially then emptying the buffer and then buffering again and then emptying. That could explain why the stream might be laggy. [14:13] <ePirat> hmmh yes [15:45] <Dark-knight> can anybody else confirm that youtube just changed its font so that blind people can now read it? [15:46] <sacarasc> Braille? [15:46] <Dark-knight> no its just ridiculously huge [15:46] <Dark-knight> anybody else see that? [15:47] <sacarasc> Nope. [15:47] <Mavrik> you have zoom broken in browser [15:47] <Mavrik> or you're being trolled by A/B test. [15:47] <Dark-knight> when you click a video all the other videos on the side have very large text [15:47] <Dark-knight> its not my zoom [15:49] <Dark-knight> can anyone confirm? [15:49] <sacarasc> I already confirmed the opposite. [15:49] <Dark-knight> did you clikc a video? [15:50] <sacarasc> http://i.imgur.com/A75nDSm.png [15:50] <sacarasc> Looks fine. [15:50] <Dark-knight> wtf [15:51] <Dark-knight> you haven't cleared your cookies yet have you? [15:51] <Dark-knight> dont do it [15:52] <Dark-knight> can you give me your YT cookies so i can get the old font back? [15:53] <sacarasc> Looks fine in a browser I have never been to Youtube on before. [16:00] <stephenmac7> I'm having an issue when streaming from rtsp: When the camera gets too much sun, the image gets corrupted and the output gets filled with "RTP: Missed X packets". Here is the output: http://sprunge.us/GgRP and the command: [16:00] <Dark-knight> oh thank code wizard [16:00] <Dark-knight> i fixed it [16:00] <stephenmac7> ffmpeg -i 'rtsp://radbear:[password_omitted]@98.231.68.88/axis-media/media.amp' -c:v copy -map 0 -an -flags:v -global_header -bsf:v dump_extra -f hls -hls_time 5 -hls_list_size 4 -hls_wrap 8 surf_stream.m3u8 [16:00] <Dark-knight> i just logged out, cleared the cache, then logged in again [17:10] <ePirat> how can I list avfoundation devices? list_devices is an invalid option [19:11] <Peter_Occ> ffmpeg -f mjpeg -r 8 -i http://192.168.0.7/video/mjpg.cgi -vf "drawtext=/usr/share/fonts/dejavu/DejaVuSans-Bold.ttf: text='\%D \%T': fontcolor=white(a)0.8: x=7: y=20" -r 10 /var/www/html/captures/videoMain2.mp4 [19:11] <Peter_Occ> The date and the time are now being added to the capture, and I'm getting this message. I'm pretty sure the syntax is correct, so what is the problem? [19:11] <Peter_Occ> [Parsed_drawtext_0 @ 0x9de4060] Stray % near 'D %T'00:10.20 bitrate= 389.6kbits/s dup=29 drop=0 [19:12] <Peter_Occ> The date and time are not being added [19:23] <sfan5> use text='%{localtime\:%D %T}' [19:34] <Peter_Occ> sfan5 thanks for the tip, works now [19:50] <Peter_Occ> There seems to be a seven second delay from when the command to capture is sent and the actual capture starts. Any way around that? [19:53] <Peter_Occ> Actually, the delay is in the timestamp itself [19:56] <Peter_Occ> If I start the video on the minute, the video starts correctly, but starts with a timestamp of 07 seconds, keeps that time for 7 seconds and then continues correctly when the video gets to the 7 second mark. [21:03] <gaffa> Hi, what would you think was wrong if ffmpeg stopped at some point saving to the output file, but it keeps processing and encoding. Just nothing is written to the file. What are other likely causes than ffmpeg? Just from the top of your heads to give me some ideas and input to debugging the problem. [21:04] <Plorkyeran> disk full? [21:04] <gaffa> Heh, I've already eliminated that :P [21:08] <Hello71> other disk errors. [21:12] <gaffa> One would think that ffmpeg would give an error message, but I can't be sure. I guess I'll just have to run traces for errors first on an inconsistent I/O problem :/. I will see if I can somehow make the process stop for debugging, when it stops writing output. [21:12] <gaffa> s/One would/I [21:51] <optix2> is there a way to generate an image sequence based on a subtitle only? [21:52] <ePirat> does ffmpeg supports ogm? [21:53] <JEEB> yes [21:53] <JEEB> as in, you can at least read it - not sure about muxing, but you definitely shouldn't be muxing OGM any more [21:53] <JEEB> for /any/ reason [22:12] <bgchris> Hello, i am using this command -f segment -segment_format flv ....... and it works just fine . My question is: It is possible to stip the metadata duration from the output segment? [22:12] <bgchris> Cause i output flv with multiple segments [22:12] <bgchris> but al the players think that flv is duration of 10 (given by -segment_time) and then it stops [00:00] --- Fri Jul 18 2014

1 0

ffmpeg-devel.log.20140716
by burek 17 Jul '14

17 Jul '14

[00:18] <cone-756> ffmpeg.git 03Vignesh Venkatasubramanian 07master:5a206569468a: lavf/matroska: Add functions for WebM DASH Manifest [00:18] <cone-756> ffmpeg.git 03Vignesh Venkatasubramanian 07master:3e73d1429045: lavf: Add WebM DASH Manifest Muxer [00:18] <cone-756> ffmpeg.git 03Michael Niedermayer 07master:e240d01c120e: avformat/matroskadec: fix declaration after statement [00:18] <cone-756> ffmpeg.git 03James Almer 07master:ad24256e7e9f: diracdec: remove unused dsputil context [00:41] <cone-756> ffmpeg.git 03Hanspeter Niederstrasser 07master:04980dbee805: avdevice/avfoundation: kCVPixelFormatType_OneComponent8 only exists from 10.8 onward [00:41] <cone-756> ffmpeg.git 03Marc Jeffreys 07master:a0b71e9f3e95: avfilter/drawtext: Add basic text shaping using libfribidi [00:54] <cone-756> ffmpeg.git 03Marc Jeffreys 07fatal: ambiguous argument 'refs/heads/release/2.3': unknown revision or path not in the working tree. [00:54] <cone-756> Use '--' to separate paths from revisions [00:54] <cone-756> refs/heads/release/2.3:HEAD: avfilter/drawtext: Add basic text shaping using libfribidi [01:41] <cone-756> ffmpeg.git 03Michael Niedermayer 07master:cf92cc8751bb: avcodec/hevc: clear HEVClcList[i] on allocation [01:59] <cone-756> ffmpeg.git 03Michael Niedermayer 07release/2.3:cf92cc8751bb: avcodec/hevc: clear HEVClcList[i] on allocation [01:59] <cone-756> ffmpeg.git 03Michael Niedermayer 07release/2.3:7fa72ff19cc0: update for FFmpeg 2.3 [02:02] <cone-756> ffmpeg.git 03Michael Niedermayer 07release/2.3:e32249605456: RELEASE_NOTES: update version numbers [02:08] <cone-756> ffmpeg.git 03Michael Niedermayer 07master:13a72d9b08c9: doc/APIchanges: update [02:08] <cone-756> ffmpeg.git 03Michael Niedermayer 07release/2.3:2678b2509910: doc/APIchanges: update [02:08] <jamrial> michaelni: change "version <next>" to "version 2.3" in the release/2.3 changelog file [02:08] <jamrial> also in master for that matter [02:10] <cone-756> ffmpeg.git 03Michael Niedermayer 07master:faafd1e4f1fc: Changelog: change "version <next>" to "version 2.3" [02:10] <cone-756> ffmpeg.git 03Michael Niedermayer 07release/2.3:bc4f6ae88e5f: Changelog: change "version <next>" to "version 2.3" [02:11] <michaelni> jamrial, done thx, would have forgotten that one probably [02:15] <Timothy_Gu> michaelni: do you think after this release we should change RELEASE to 2.4.git? Currently it's 2.2.git, and it is uncommon that 2.2.git > 2.2.5 [02:21] <michaelni> no objections from me [02:29] <jamrial> BBB: the patch merely moves draw_edges from one context to another. it doesn't change any decoder or encoder behavior [02:29] <jamrial> BBB: h264, vp8 and vp9 have nothing to do with it. they don't depend on draw_edges afaics [02:30] <jamrial> unless i'm missing something... [02:31] <BBB> jamrial: exactly [02:31] <BBB> jamrial: so right now, a ffmpeg built with just vp8, vp9 and h264 (think chrome) does not include draw_edges [02:31] <BBB> jamrial: your patch changes that, by moving it into a context used by vp8/vp9/h264 [02:31] <BBB> theyre not _using_ it, but its included in the binary [02:31] <BBB> I dont want that [02:32] <BBB> that old stuffy mpeg encoder code needs to stay where it is now; in the mpeg encoder [02:32] <BBB> decoders shouldnt touch it [02:32] <BBB> emu_edge is always faster [02:34] <jamrial> i'm ok with any solution, really. i was merely following michaelni's suggestion to remove the mpegvideoenc dependency from snow and dirac decoders [02:36] <michaelni> i suggested it as it seemed like the simplest way to improve it, i dont claim it leads to the best code independant of effort [02:36] <jamrial> although i'm not going to remove the draw_edges usage from those two decoders myself. i'll let someone else do that if needed [02:40] <Timothy_Gu_> michaelni: did you see my patch on RELEASE_NOTES? 2.3 won't be complete without it. [02:42] <BBB> so does libavs dirac decoder depend on mpegenc? [02:43] <BBB> or does libav not have a dirac decoder? [02:44] <jamrial> looks like they don't have a dirac decoder at all [02:44] <jamrial> only parser [02:45] <kierank> they don't have a dirac decoder [02:45] <kierank> there was a lot of cosmetic madness iirc that meant it never got merged [02:45] <kierank> the guy got a bit fed up [02:46] <wm4> lol, so that does happen [02:46] <jamrial> yeah, no wonder diracdsp was untouched as part of all the dsputil massacre from the past few months [02:47] <kierank> wm4: imo that gsoc was probably the worst [02:47] <kierank> and it's got better now [02:47] <wm4> cosmetic gsoc? [02:47] <kierank> usually the commitors who care about the cosmetics do them now [02:47] <kierank> wm4: people writing good code and then getting blasted for cosmetics [02:50] <cone-756> ffmpeg.git 03Timothy Gu 07release/2.3:bef4d9bf87f7: RELEASE_NOTES: update [02:52] <michaelni> Timothy_Gu_, applied, thx [02:54] <Timothy_Gu_> Yeah. Most of their patches are like "LGTM, there are only 60 nits. Please fix them before pushing." [02:55] <kierank> Timothy_Gu_: in the past yes [02:55] <kierank> but these days no [02:56] <Timothy_Gu_> because no new developers are coming to Libav, and the only new dev in the past 12 (?) months, koda, has been thoroughly trained to avoid DonDiego's blasts [02:58] <michaelni> BBB, i can look into converting snow to emu_edge but not sure whe ive time [03:00] <jamrial> for that matter, diracdec is calling ff_emulated_edge_mc_8 directly instead of using the function pointer from videodsp [03:00] <jamrial> is there a reason for that? [03:01] <jamrial> changing it to use the pointer doesn't break fate here [03:01] <j-b> emu_edge \o/ [03:13] <BBB> jamrial: I worked on libav when I converted emu_edge to be a dsp function [03:13] <BBB> jamrial: libav probably didnt have a dirac decoder back then [03:13] <BBB> so it was naturally not converted when the patch was merged here [03:14] <wm4> [FFmpeg-devel] [PATCH] lavfi: check refcount before merging. <- wow this format negotiation stuff is hitler [03:15] <BBB> godwins law! [03:15] <BBB> :D [03:15] <Daemon404> wm4, more like the SS [03:15] <Daemon404> since lavfi is hitler [03:16] <wm4> somehow this negotiation stuff is terribly complicated because it tries really hard to insert "optimal" conversion filters [03:16] <Daemon404> you probably dont want to see dshow pin negotiation code [03:16] <Daemon404> :P [03:19] <jamrial> BBB: ah good. i'll send a patch changing it then [03:19] <cone-756> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n2.3': unknown revision or path not in the working tree. [03:19] <cone-756> Use '--' to separate paths from revisions [03:19] <cone-756> refs/tags/n2.3:HEAD: Changelog: change "version <next>" to "version 2.3" [03:19] <Daemon404> dumb bot is dumb [03:20] <Timothy_Gu> It has been like this for years if not months [03:20] <Daemon404> cia.vc existed in the not so distant past. [03:20] <kierank> wm4: all filter negotiation and/or reconfiguration is messy [03:20] <Daemon404> be thankful the caller doesnt have to manually do negotiation [03:20] <Daemon404> via some callback [03:20] <Daemon404> or something [03:21] <kierank> " IRC and Subversion have taken a back seat to Twitter and Github" [03:21] <kierank> lol [03:21] <kierank> development via twitter [03:21] <Daemon404> im trying to deal with a github-tracker-only project right now [03:21] <Daemon404> its pain. [03:21] <kierank> IRC is still the best way to do collaborative development [03:21] <Daemon404> worse because the entire community around it is super naive / wtf [03:22] <Daemon404> (mozjpeg) [03:22] <kierank> lol [03:23] <Daemon404> im trying to help them out, but damn [03:23] <Daemon404> i might just give up [03:23] <j-b> twitter is the new IRC [03:23] <j-b> lol [03:23] <Daemon404> i thought that was hipchat [03:24] <Daemon404> or is there somthing more hip now [03:24] <kierank> I've not found anything better than IRC where I can just get people to click a button and visit my channel [03:25] <wm4> obviously you should use facebook for development [03:25] <jamrial> #bugsquashing #optimizing #testinginproduction #YOLO [03:25] <kierank> I get people trying to friend me on facebook [03:25] <jamrial> though YOLO is probably outdated by now [03:25] <kierank> and then asking for help [03:35] <Daemon404> oh [03:35] <Daemon404> so thats where i remember this guy on mozjpeg from [03:35] <Daemon404> https://github.com/vapoursynth/vapoursynth/issues/85 [03:35] <Daemon404> SIGH [03:36] <wm4> lol [03:36] <wm4> did he make the same post on mozjpeg or what [03:36] <Daemon404> https://github.com/mozilla/mozjpeg/issues/66 [03:36] <Daemon404> that's this [03:37] <Daemon404> there are far more annoyings things though [03:37] <wm4> uargh [03:37] <wm4> nnedi in a jpeg decoder? [03:37] <Daemon404> in fucking cjpeg [03:37] <wm4> *encoder [03:38] <Daemon404> https://github.com/mozilla/mozjpeg/issues/21#issuecomment-48390582 [03:38] <Daemon404> and [03:38] <wm4> Daemon404: I see you're trying hard to prevent his retarded ideas to make it into the project [03:38] <Daemon404> https://github.com/mozilla/mozjpeg/issues/67 [03:38] <Daemon404> these anger me. [03:39] <Daemon404> theyre breaking ABI silently and leaving it as libjpeg.so with the jpeg6 soname [03:39] <Daemon404> silently. [03:40] <kierank> is there anything that explains the mapping from av_opt to the values in avcodeccontext? [03:42] <Timothy_Gu> kierank: lavc/options_table.h [03:43] <kierank> ah thanks [03:56] <BBB> jamrial: can you remove the prototype of ff_emulated_edge_mc_8_c from videodsp.h if we no longer need to expose it? [03:56] <BBB> jamrial: unless its used in other palces [03:56] <BBB> jamrial: (not sure, didnt check) [04:01] <Timothy_Gu_> BBB: clang's static analyzer reports some dead code in vp9 [04:01] <BBB> uhm& ok [04:01] <BBB> which? [04:01] <Timothy_Gu_> http://50d2506d.ngrok.com/report-c92ac6.html http://50d2506d.ngrok.com/report-6e0817.html and http://50d2506d.ngrok.com/report-52fa84.html [04:04] <jamrial> BBB: it's still being used by gmc_mmx, form MpegVideoDSPContext [04:04] <jamrial> *from [04:04] <BBB> the first two seem correct [04:04] <BBB> yeah theyre all correct [04:05] <BBB> Timothy_Gu_: patch to remove these is fine [04:05] <BBB> jamrial: ah I see [04:21] <cone-756> ffmpeg.git 03Vittorio Giovara 07master:14b4e64eabc8: g2meet: allow size changes within original sizes [04:21] <cone-756> ffmpeg.git 03Michael Niedermayer 07master:f00bb086cb97: Merge commit '14b4e64eabc84c5a5e57c8ccc56bbeb95380823b' [04:21] <cone-756> ffmpeg.git 03James Almer 07master:b67a0e99ee82: diracdec: don't call ff_emulated_edge_mc_8 directly [04:21] <cone-756> ffmpeg.git 03Michael Niedermayer 07master:8156e036e527: avcodec/snowdec: remove mpegvideoencdsp dependency [04:33] <Timothy_Gu_> BBB: I decided not to make a patch because that would decrease readability. If you wanna do this 3-line commit, go ahead. [04:33] <BBB> why does it decrease readability? [04:35] <Timothy_Gu_> like the one at l.2481 if I remove the unused h4 but leave w4 other people will be like "is this wrong?" [04:35] <Timothy_Gu_> w4 is used [04:35] <BBB> are those people that understand the fundamental nature of this video codec? [04:35] <BBB> I mean, there is a topright but not bottomleft, so its obvious to me that you need a w but not a h variable for that [04:36] <BBB> (unlike hevc, which has both) [04:36] <Timothy_Gu_> OK, I have no idea what you are talking about, but sure, I'll send a patch [04:36] <Timothy_Gu_> (I mean I understand it in words but not the function) [04:37] <BBB> you know spatial (intra) prediction right? [04:38] <Timothy_Gu_> nope :p [04:39] <Timothy_Gu_> I haven't even read JPEG specs [04:39] <BBB> hm... [04:39] <BBB> ok [04:39] <Timothy_Gu_> I know what Intra is though [04:39] <BBB> so [04:39] <BBB> intra prediction is taking edge pixels to predict a blocks content [04:39] <Timothy_Gu_> what is a block? [04:39] <Timothy_Gu_> a part of a frame? [04:39] <Timothy_Gu_> what is a macroblock? a group of blocks? [04:40] <BBB> the base subdivision unit of a frame in which the actual video content is coded [04:40] <BBB> for h264, 16x16 is the base block unit size [04:40] <Timothy_Gu_> ok [04:40] <BBB> vp8 same [04:40] <BBB> vp9/hevc its 64x64 -> 8x8 (with 4x4 support throug some hacks) [04:40] <Timothy_Gu_> huh? [04:40] <BBB> so to predict a block, you take edge pixels, apply a function over it, and thats the blocks predicted content [04:40] <Timothy_Gu_> where did the 64 come from? [04:41] <BBB> vp9/hevc have bigger blocks [04:41] <BBB> basically gives better compression for some hd content [04:41] <Timothy_Gu_> then how about 8? [04:41] <BBB> for complex content [04:41] <BBB> i.e. adaptive to content of frame [04:41] <BBB> a 64x64 block can divide to 4 32x32 blocks, or be coded as-is [04:41] <BBB> 32x32 -> 16x16 -> 8x8 -> (hacky) 4x4 [04:42] <Timothy_Gu_> ok, so HEVC allows any of those (64, 32, 16, 8, 4)? [04:42] <BBB> yes [04:42] <BBB> and the encoder will select what makes most sense given the content [04:42] <BBB> for static motion (or no motion), youll likely get bigger blocks for hd content [04:43] <BBB> for highly complex motion patterns, maybe smaller blocks [04:43] <Timothy_Gu_> and then the block's predicted content is compared with the actual content? [04:43] <Timothy_Gu_> and a "diff" is coded? [04:43] <BBB> right, the difference is coded as transformed coefficients [04:43] <BBB> diff = quantize(transform(src[]-pred[])) [04:44] <BBB> and thats your basic video encoder right there [04:44] <Timothy_Gu_> how does transform() work? [04:44] <BBB> typically just a forward dct [04:45] <Timothy_Gu_> how does DCT/Fourier transform work? [04:46] <BBB> Id just read the wikipedia page :D [04:46] <Timothy_Gu_> BTW I asked my homeroom teacher (who teaches AP Stats) once on Fourier transform for a "for dummies" definition, and he was like "oh I remember back in college I did a project on it. I'm pretty sure it is covered in multi-variable calculus" [04:47] <BBB> its basically a function that finds frequency patterns in the set of difference values [04:47] <BBB> so think of the coefficients as weights of sine wave functions [04:47] <BBB> (ignore the 2d aspect for now, think just 1d) [04:47] <BBB> so if I have an array of 4 diff values double x[4] [04:47] <Timothy_Gu_> Then he pulled up the Wikipedia page (¡qué coincidencia!) that has a chart I absolutely don't understand [04:48] <Timothy_Gu_> ok continue [04:48] <BBB> and their values are like 0, 1, 1, 0 [04:48] <BBB> you can see this as a wave function that goes up towards the middle of the line, and down at the edges [04:48] <BBB> now imagine that I predefined 4 sine wave functions with different frequencies [04:49] <BBB> first DC (i.e. infinite wavelength), and the others something like pi/n wavelength (or maybe 2pi/n, I forgot), where n is 1, 2, 3 [04:50] <BBB> then the values (0,1,1,0 in this case, but really any set of values) can also be represented as a vector of 4 multipliers times each of these 3 wave functions (plus the one with infinite wavelength - the one I called dc) [04:50] <BBB> in this case, the 0,1,1,0 have an average value of 0.5, so the weight of dc would be 0.5 [04:51] <BBB> then after I subtract that, Im left with -0.5, 0.5, 0.5, -0.5 [04:51] <BBB> and the 2pi/2 describes that quite well (if you assume the function is centered at the middle), again with a multiplier of 0.5 (if I assume the peak to be 1) [04:51] <BBB> so then the coefficients are 0.5, 0, 0.5 and 0 [04:52] <BBB> for typical natural patterns, which are common in video, transforms decrease the amount of information that needs to be coded [04:52] <BBB> the inverse transform is just the opposite, its basically multiply the wav functions bu the cofficients (multipliers), add them all up, and you have your original diff back [04:52] <Timothy_Gu_> is transform lossless? [04:53] <BBB> no [04:53] <BBB> it can be [04:53] <BBB> but dcts typically arent [04:53] <BBB> vp9s dht (4x4) is lossless [04:53] <BBB> dwt, sorry [04:53] <BBB> h264s dht is maybe lossless? [04:53] <BBB> but these are dct approximations that are changed to be less dct-like (thus worse compression) to become lossless [04:54] <Timothy_Gu_> ok [04:54] <BBB> also quantization is not lossless, so lossless dht/dwt only works for a quantizer of 1.0 (i.e. none) [04:55] <BBB> vp8 has no lossless transform [04:55] <BBB> Im honestly not sure about hevc [04:56] <BBB> bedtime now, sorry... [04:56] <Timothy_Gu_> how did -.5, .5, .5, -.5 become .5, .0, .5, .0 again? [04:56] <Timothy_Gu_> last question [04:57] <BBB> -.5, .5, .5, -.5 became the third coefficient [04:58] <BBB> look at this image: http://en.wikipedia.org/wiki/Discrete_cosine_transform#mediaviewer/File:Dct… [04:58] <BBB> imagine that each small square boxed in red edges is a 2d sine function [04:58] <BBB> just look at the top 8 (in fact, just the left 4 of the top row) [04:58] <BBB> imagine now that white is 1.0, and black is -1.0 [04:59] <BBB> the topleft one is a sine function where all values are 1.0 [05:00] <BBB> the second one is a sine function that goes from 1.0 to -1.0, like 1.0, 0.4, -0.4, -1.0 or so [05:00] <Timothy_Gu_> If all the values are the same, is it even a sine function? [05:00] <BBB> with infinite wavelength, sure [05:00] <Timothy_Gu_> ok [05:00] <BBB> (but I mean, its a special case, yes) [05:01] <BBB> the third one is something like 1.0, -0.7 -0.7, 1.0 [05:01] <BBB> and the fourth one is & 1.0, -1.0, 1.0, -1.0 [05:01] <BBB> makes sense? [05:01] <Timothy_Gu_> So... are these "coefficients"? [05:02] <BBB> no, these are the base functions [05:02] <BBB> the coefficients are the multipliers times each of these base functions to get the diff [05:02] <BBB> like a matrix multiplication (in fact, slow implementations of f/idct are like a matrix multiply) [05:03] <BBB> so if your diff was 1.0, 1.0, 1.0, 1.0 [05:03] <BBB> you only need the first base function to describe this (multiplier=1.0) [05:03] <BBB> the multipliers for the other base functions are 0 [05:03] <BBB> so your coefficients are then 1.0, 0.0, 0.0, 0.0 [05:03] <Timothy_Gu_> oh, now I get it [05:03] <BBB> and so for more complex diff patterns youll use all 4 base functions with various coefficients [05:04] <BBB> the typical effect is that the earlier coefficients are bigger and the later are smaller, so that after quantization, they are zero [05:04] <cone-756> ffmpeg.git 03Michael Niedermayer 07master:aa1d096d027b: avcodec/snow: only allocate space for edges when encoding [05:04] <BBB> which means you can code only nonzero ones and thus save space [05:04] <BBB> thats why transforms help compression [05:05] <BBB> instead of coding 4x4 pixels for a 4x4 block, you code only& say, 13, or 7, or even just 2 [05:05] <Timothy_Gu_> So if I understood correctly, DCT = avg all the members out -> use a wave form (or some wave forms) to simulate it by "assembling" different basic waves together [05:06] <Timothy_Gu_> And then after DCT you quantize ~0 values as 0 [05:06] <Timothy_Gu_> ~ as in approximated [05:10] <BBB> well, theyre not floating point values [05:10] <BBB> theyre fixed point [05:11] <BBB> so 1.0 would be like 1000, and 0.1 would be 100 [05:11] <BBB> then you use a quantizer [05:11] <BBB> say my quantizer is 200 [05:11] <BBB> 1000/200=5 [05:11] <BBB> 100/200=0 [05:11] <BBB> so the 100 fell off the radar [05:11] <Timothy_Gu_> ok, in floating point concept mine is correct, right? [05:12] <BBB> yeah [05:12] <BBB> and thats your basic fdct [05:12] <BBB> idct is the same, but inverse [05:12] <BBB> and dequant is just multiply coded coefficients by the quantizer value [05:12] <BBB> so 5*200=1000, 0*200=0, etc. [05:13] <BBB> I guess now is bedtime :-p [05:13] <Timothy_Gu_> ok, thanks for everything [05:14] <Timothy_Gu_> good night! [05:14] <Timothy_Gu_> still 2 hours before my bed time [05:15] <Timothy_Gu_> I gotta save this somewhere [05:33] <fionag> if you have any other questions I can try to help! [05:49] <Timothy_Gu_> fionag: thanks, but I think this is enough for today :D [05:52] <fionag> okay :P [05:52] Action: fionag looks at the log. okay that is quite a lot [08:43] <db0> Compn: about the .js files, this is just the recommended header so it works well on IE9 and I dont know exactly so we can also remove this part of the header, or we can download them locally. I assume it's recommended to include the url in case the content needs to be changed for some reason [09:17] <ubitux> oh, --enable-memory-poisoning found an actual issue? [09:17] <ubitux> cool [09:25] <rrouton> hello everyone [09:27] <rrouton> anyone here have any experience with the C/C++ libs? I have written a decent transcoder in order to learn ffmpeg's api and was wondering if anyone knows how the memory works under the hood, specifically the buffering of av packets when using av_interleaved_write [09:28] <rrouton> is it smart enough to know not to buffer anything before the stream if there is an audio frame adjacent to a video ? [09:31] <rrouton> i am not sure I like this chat client (LimeChat) [10:02] <plepere> morning [11:46] <ubitux> frame side data doesn't have a variable size? [11:46] <ubitux> :( [13:03] <BBB> Timothy_Gu: thats actually a fair point, michaelni or #x264dev are good places to learn more (or write your own video decoder [or encoder?]) [14:38] <cone-165> ffmpeg.git 03Timothy Gu 07master:ea6178fff8ee: get_bits: remove unused assignment [14:49] <cone-165> ffmpeg.git 03Timothy Gu 07master:9bc0410e4f89: vp9: remove unused assignment [15:46] <ubitux> TIL i learned michaelni is a place [15:50] <Daemon404> wat [15:50] <Daemon404> also s/i learned // [16:01] <gnafu> Daemon404: RIP in peace. [16:02] <ubitux> Daemon404: yeah, i learned twice [16:02] <Daemon404> ;p [16:02] <ubitux> Daemon404: 13:03:19 <@BBB> Timothy_Gu: thats actually a fair point, michaelni or #x264dev are good places to learn more (or write your own video decoder [or encoder?]) [16:02] <Daemon404> o [16:03] <Daemon404> how i learned to write a decoder in ffmpeg: cp simpledec.c mydec.c [16:03] <Daemon404> then hack mydec.c [16:03] Action: Daemon404 runs [16:03] <ubitux> that's how you're supposed to write filters [16:03] <ubitux> ;) [16:03] <Daemon404> lol [16:04] <Daemon404> the only thing i ever had trouble with in lavc was the vlc stuff [16:04] <Daemon404> ff_init_vlc_sparse is confusing [18:06] <nevcairiel> GCC 4.9.1 is out, anyone run a full fate on it yet? [18:07] <JEEB> oh, finally_ [18:07] <JEEB> motherfucking finally I should say [18:07] <JEEB> :V [18:07] <JEEB> it should have at least the FLAC fix [18:07] <JEEB> since that got backported to 4.9.x [18:08] <Case> backported from what? Isn't 4.9.x latest branch? [18:08] <JEEB> all fixes go into master first [18:08] <JEEB> and then get backported [18:08] <JEEB> because releases get branched off some time before release [18:09] <JEEB> I think at the RC point [18:10] <Case> no new lovely Windows packages at http://files.1f0.de/mingw/ or xhmikosr's site yet :( [18:10] <JEEB> nev builds those :P [18:10] <JEEB> and he probably wanted to know if anyone ran full FATE on it yet exactly for that [18:11] <Case> damn you nevcairiel [18:11] <Case> weren't those tests already ran with some earlier builds and confirmed that things work? [18:13] <JEEB> yeah, but you want to make sure that the current state works [18:22] <nevcairiel> Will take me a bit to build them, away from my systems for a couple more days [18:24] <Daemon404> \o/ [18:25] <jamrial> i assume the celebrations are for gcc 4.9.1? :P [18:26] <Daemon404> yes [18:29] <Daemon404> that but title makes no sense... [19:59] <ubitux> michaelni: -skip_idct is nice; anything working with h264? [20:05] <michaelni> adding skip_idct support to h264 should be possible [20:06] <wm4> what for [20:06] <ubitux> wm4: see [RFC] avcodec: export MB information in frame side data [20:07] <ubitux> tl;dr i'm looking for a way to avoid decoding the frames so the MV extraction doesn't need to be so slow [20:08] <ubitux> and in my personal use-case i'm interested in h264 in particular [20:08] <wm4> how is that useful [20:08] <ubitux> > faster [20:08] <ubitux> i don't need the frames, i just want the MV information [20:08] <wm4> also prefixing the option name with sd_ is a bad idea [20:08] <ubitux> ah? [20:09] <ubitux> (it's not an option but a flag value) [20:09] <ubitux> (so it won't be a -sd_mb_info, but -flags2 +sd_mb_info) [20:12] <wm4> sd sounds like low resolution [20:14] <ubitux> oh, mmh [20:49] <cone-609> ffmpeg.git 03Diego Biurrun 07master:adff0a816634: arm: dsputil: Coalesce all init files [20:49] <cone-609> ffmpeg.git 03Michael Niedermayer 07master:21dfabfa6461: Merge commit 'adff0a8166345bb9513f0f658043fb6387e90122' [21:04] <cone-609> ffmpeg.git 03Timothy Gu 07master:1b034483853b: aaccoder: remove unused assignment [21:34] <cone-609> ffmpeg.git 03Michael Niedermayer 07master:a3f752bcee4d: avcodec/snow: remove unused variables [21:52] <cone-609> ffmpeg.git 03Ben Avison 07master:649c666137f4: armv6: Accelerate vector_fmul_window [21:52] <cone-609> ffmpeg.git 03Ben Avison 07master:57641410d1a3: armv6: Accelerate butterflies_float [22:56] <cone-609> ffmpeg.git 03Michael Niedermayer 07master:7da2592f5677: avformat/md5enc: add format_version, to allow selecting which version to use [22:57] <jamrial> michaelni: micro bump? [23:12] <cone-609> ffmpeg.git 03Michael Niedermayer 07master:37a0ac1d8227: avformat: Micro bump for "md5enc: add format_version, to allow selecting which version to use" [23:12] <michaelni> jamrial, done thx [00:00] --- Thu Jul 17 2014

1 0

ffmpeg.log.20140716
by burek 17 Jul '14

17 Jul '14

[00:02] <Dark-knight> I re-muxed a .mkv video with a subtitle stream that automatically played by default. when i remuxed it to .mp4 the subtitle stream no longer automatically played. i had to select it. [00:02] <Dark-knight> how do i fix this? [00:05] <sobaah> I've actually been doing some of this stuff myself Dark-knight, the issue I see is that [00:05] <sobaah> ffmpeg seems to lose the language ID for the audio or sub track [00:05] <sobaah> try adding it in by remuxing with this option: [00:05] <sobaah> -metadata:s:2 -language=eng [00:05] <sobaah> The "s" is the stream index [00:05] <sobaah> I'm guessing it's 2 [00:05] <sobaah> 0=video, 1=audio, 2=sub, but check for sure [00:06] <Dark-knight> alright [00:06] <Dark-knight> whats the "-language=eng" for? [00:06] <sobaah> oops [00:06] <sobaah> sorry just language=eng [00:06] <sobaah> no dash [00:06] <sobaah> that is the actual tag you are adding to the stream [00:07] <sobaah> the -metadata:s:2 just means apply something to this stream [00:07] <Dark-knight> can you explain to me what those commands do. I would like to understand why things work. [00:07] <sobaah> sure [00:07] <sobaah> -metadata:s:n tag=value [00:07] <Dark-knight> i trying to learn, not just copy paste [00:07] <sobaah> so the "n" is the stream number which ffmpeg will show you if you do "ffmpeg -i file" [00:08] <sobaah> yep, I hear ya [00:08] <sobaah> the tag=value is the actual data that will be added to the stream [00:08] <sobaah> there are many tags you can add, "language" is one of them [00:08] <Dark-knight> ok im thing im getting it [00:09] <sobaah> once you add that language tag, your video player should show you that the language ID is set to "English" [00:09] <sobaah> but you can put whatever ISO language code in there [00:09] <sobaah> jpn, spa... et cetera [00:09] <Dark-knight> how does that make the subtitles play by default? [00:10] <sobaah> well, I believe that in some players the preferred language will choose the subs [00:10] <sobaah> but I could be wrong [00:11] <sobaah> so because you lost any language ID in the file, the player didnt choose anything [00:11] <Dark-knight> when i view the file in mediainfo it shows the 3 different substreams i have and the 1st one has "default: Yes" next to it [00:11] <sobaah> I see [00:12] <sobaah> so maybe that's a tag [00:12] <Dark-knight> itsn't it as simple as flipping a swich? [00:12] <Dark-knight> i assume all i have to do it tell the container which stream to play by default [00:13] <Dark-knight> is* [00:13] <sobaah> try this [00:13] <sobaah> -metadata:s:2 default-flag=1 [00:13] <sobaah> make sure Stream2 is your subs though [00:14] <Dark-knight> well whatever number the subs are anyway [00:14] <sobaah> yep [00:14] <Dark-knight> what does the "default-flag=1" do? [00:14] <sobaah> 1 generally means "True" in programming [00:14] <sobaah> 0 means "False" [00:15] <sobaah> or perhaps try Yes instead of 1 [00:29] <sobaah> hi all, I am having a bit of an issue with a large MKV file (9GB), where I am using the "-attach FONTFILE.ttf" option. The issue is that ffmpeg seems to write the entire file to memory first but I don't have the 9GB it would need. Is there any way around this? [00:29] <sobaah> this seems to happen also if you try to add an image file to the file as well [00:55] <Dark-knight> added to my notes [00:55] <Dark-knight> thanks [00:55] <sobaah> it worked? [00:56] <Dark-knight> no but i will try it later [00:56] <sobaah> ahh [00:56] <Dark-knight> i havent done it yet [00:56] <sobaah> yah, I read some more into it and there seems to be some confusion with how the default flag works [00:56] <Dark-knight> i was going to after dinner [00:56] <Dark-knight> ok [00:56] <Dark-knight> care to share? [00:56] <sobaah> I couldnt find a clear answer on how to set a default sub or audio stream on a file...no idea why [00:56] <Dark-knight> wow thanks alot google [00:56] <sobaah> haha [00:57] <Dark-knight> lol [00:57] <sobaah> apparently the metadata option may not work [00:57] <Dark-knight> really? [00:57] <sobaah> it seems there is another way that the stream is set as default [00:57] <Dark-knight> does ffmpeg even have an option to set a default stream? [00:58] <sobaah> I looked for about 10 minutes with various searches and found a lot of info of issues with how to set a stream to non-default [00:58] <sobaah> ... [00:58] <Dark-knight> if not, why? and how come nobody else has thought of adding an option to do that? [00:58] <sobaah> but nothing about setting it to default [00:58] <Dark-knight> haha [00:58] <Dark-knight> that figures [00:59] <sobaah> someone had an issue where they had multiple subs and ffmpeg set the original sub that had the default flag as non-default [00:59] <sobaah> and all the other subs got the default flag [00:59] <sobaah> strange =) [00:59] <sobaah> hope you ge tit [00:59] <sobaah> get it* [01:02] <sobaah> well I cant find a soltuon to my own issue [01:02] <sobaah> solution [01:04] <Dark-knight> clear up some memory or defrag your harddrive [01:06] <Dark-knight> how do i highlight certain lines in pastebin? [01:07] <c_14> I'm not sure that works, why? [01:08] <Dark-knight> ive seen people make specific lines a different color in PB before [01:08] <Dark-knight> i want to highlight things [01:13] <c_14> Maybe some part of the syntax highlighting? [01:14] <Dark-knight> no idea [01:34] <Dark-knight> wow [01:34] <Dark-knight> http://pastebin.com/faq#13 [01:34] <Dark-knight> there it is [01:35] <Dark-knight> can't believe i never saw that before [01:35] <Dark-knight> its so obvious now' [01:35] <Dark-knight> giant yellow pen in the top right of the text box [01:37] <c_14> pfft, giant icons were made to be overlooked [01:44] <Dark-knight> http://pastebin.com/XWhiZdVX [01:45] <Dark-knight> sobaah as you see, the subtitle stream "Signs" is the default [01:46] <Dark-knight> how do i make stream 6 the default? [01:46] <Dark-knight> or even stram 5 [01:46] <Dark-knight> i just want to know "How". [01:54] <c_14> Can't seem to find an option to do that with FFmpeg, but you could use something like mkvtoolnix. [02:04] <Dark-knight> the mkv file is not the problem [02:05] <Dark-knight> its when i re-mux it to mp4, that the sub stream no longer plays by default [02:05] <Dark-knight> for some reason ffmpeg looses the default flag during re-muxing [02:06] <Dark-knight> if only ffmpeg and an option to assing default streams [02:06] <sacarasc> Try using mp4box, then. [02:06] <Dark-knight> assign* [02:06] <Dark-knight> i don't understand why this isn't already in ffmpeg [02:11] <debianuser> Hello, I have two short questions. Looks simple, but I can't find an answer somewhy. First: that line "Press [q] to stop, [?] for help"... How can I disable this feature? I mean, print everything as usual, but don't read any commands. Is there some -no-console or -no-commands option for ffmpeg? [02:14] <debianuser> Second quesion: ffmpeg is able to detect h264 profile ("Video: h264 (High)"). I guess it's stored somewhere in file header. Can it change just that information without reencoding the entire stream? [02:17] <klaxa> as to your first question: see -loglevel [02:17] <satanist> debianuser to the first question: you can change the stdin with your shell by add " < /dev/null" [02:17] <klaxa> as to your second question: no [02:23] <c_14> debianuser: the correct command for the first question is actually -nostdin [02:25] <debianuser> klaxa: Tried '-loglevel info' but ffmpeg still reacted to console input. [02:26] <c_14> try -nostdin [02:26] <Engin> my best bet to encode a 3 channel wav in terms of compression/quality efficiency? [02:29] <debianuser> satanist, c_14: About "</dev/null", nice idea, I have not thought about it. -nostdin... wow, couldn't find that. Thank you! [02:29] <debianuser> The only disadvantage is that with any of those options it also does not ask about file overwriting, just exits. I would prefer it to still ask about overwriting, just don't expect any commands during file encoding. [02:30] <c_14> not sure that works, but you can use -n to disable overwriting [02:35] <Freakshow> quick ffprobe question: how can I sniff the incoming TSoIP unicast stream for channel info? [02:37] <c_14> ffprobe -show_streams -select_streams a [02:37] <c_14> and then check for channels/channel_layout [02:45] <Dark-knight> official link of mp4box? [02:47] <sacarasc> http://gpac.wp.mines-telecom.fr/ I think. [02:48] <Dark-knight> you sure? [02:48] <sacarasc> No, which is why I said "I think". [02:48] <sacarasc> But the project it is from is called gpac. [02:48] <Dark-knight> ok thanks :) [02:59] <Freakshow> c_14: thanks, trying that now [02:59] <Freakshow> what is the 'a' for after -select_streams? [02:59] <c_14> audio streams [03:11] <debianuser> Maybe there's some way to manually edit h264 profile in mp4 container? I need that just for a few tests, so I don't mind doing that in hex-editor, just can't find how to do that. http://coolsoft.altervista.org/en/h264leveleditor -- this tool kind of does that, but it's not open source. [03:12] <sacarasc> To edit that, you'd have to reencode... The profile tells the decoder what the stream might have in the way of features. [03:14] <debianuser> I understand that the file may become invalid if any of those features were used. It's just there was a guess, that a device in question can actually decode all the High-profile features, but its software player checks profile and refuses to start trying. [03:15] <debianuser> To check that guess I wanted to patch "High" to "Baseline" and see what happens... [03:15] <debianuser> But I can't find how to do that [03:16] <Hello71> profiles don't work the way you think they do. [03:16] <sacarasc> Dark_Shikari could probably help with this, but I don't know if he even comes to this network any more. [03:18] <Hello71> amendment: that sounds highly unlikely. [03:19] <debianuser> I thought "profile" is just a shortcut for settings. I.e. "baseline" == "--no-8x8dct --bframes 0 --no-cabac --cqm flat --weightp 0". At least that's how it looks from `x264 --fullhelp` output. So it's just saved in the header to help player quickly understand whether it can decode that file without actually decoding everything. Where am I wrong? [03:22] <Hello71> see my amendment. [04:04] <Dark-knight> is the only pre-built ffmpeg build for windows really Zeranoe's? [04:05] <Dark-knight> how often does he update that? [04:05] <sacarasc> Most days. [04:05] <sacarasc> http://ffmpeg.zeranoe.com/builds/win64/static/ [04:19] <debianuser> I found it! In case someone else asks how to get a broken mp4 file by chaning video profile/level: https://stackoverflow.com/a/21121914 Using that I was able to successfully change "Main" to "Baseline" and the video even played fine in mplayer after that. [04:27] <Dark-knight> why has he stopped coming to this chat? [04:33] <debianuser> But hexediting a High-profile 3.0 h264(mp4) have not worked. By changing that to Baseline 3.0 I got a broken resolution: 720x406 turned into 16x160, and there were lots of decoding errors. I may understand where decoding errors came from, but have no idea why resolution have changed. [04:33] <Hello71> you are surprised that your broken mp4 is broken? [04:34] <Dark-knight> lol [04:34] <debianuser> No. :) I just surprized why it was broken much more than I was breaking it. [04:40] <Dark-knight> what code syntax in the file container determines which subtitle stream is set to default? If we can figure that out, then a reverse-engineered command line can be determined. [04:41] <Dark-knight> thus a solution to my, or anybody elses problem can be achieved. [07:04] <Dark-knight> anybody actually looking for that code syntax? [07:04] <Dark-knight> maybe im using the wrong words [07:05] <Dark-knight> code string? perhaps is a better word? [07:19] <relaxed> Dark-knight: For the ps3 you could use the m2ts container and mark the default subtitle track with tsmuxer [07:19] <relaxed> http://forum.doom9.org/showthread.php?t=168539 [07:20] <Dark-knight> why cant ffmpeg just do it? [07:20] <Dark-knight> thanks though [07:20] <Dark-knight> ill just use mp4box [07:20] <Dark-knight> sucks that i have to dl another prgram just to get the job done [07:34] <anshul> I am getting same pts=1668330 for all packets for subtitle stream [07:34] <anshul> In many videos that i have pts value is same 166830 [07:35] <anshul> does 1668330 is an magic number that specify that this is not pts [09:20] <rrouton> hello all [09:20] <rrouton> trying irc for the first time in 12 years, can anyone see this? [09:21] <Dark-knight> dont be startled by the silence young one. if you speak it, they will come. [09:21] <Dark-knight> hmmmmmmmmm [09:22] <rrouton> I haven't used this in ages since mirc back in the day =) [09:24] <rrouton> anyone here have any experience with the C/C++ libs? I have written a decent transcoder for my purposes and was wondering if anyone knows how the memory works under the hood, specifically the buffering of av packets when using av_interleaved_write [09:28] <Dark-knight> sacarasc c_14 klaxa sobaah Paranoialmaniac relaxed sfan5 Can any of you help this gentleman? [09:29] <sfan5> mass-highlighting is usually not appreciated... [09:29] <Dark-knight> sry [09:29] <Dark-knight> wont happen again [09:29] <Dark-knight> 12 years is crazy [09:30] <Dark-knight> i couldn't go more than 3 [09:32] <rrouton> to be honest it might be more than that actually [09:33] <rrouton> ages ago when i was in the warez scene, what client is used mostly these days? Not liking this one [09:36] Action: sfan5 uses XChat [09:40] <rrouton> I have to say though that this is a very unforgiving api =) fun to play with though when you can get something out of it [09:42] <rrouton> ah i see this is more for the command line tool and less the actual libraries eh? [10:56] <FrEaKmAn_> hi all.. I'm following tutorials at https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join%2C%20merge)%20m… [10:56] <FrEaKmAn_> but I'm always getting Unrecognized option 'c' [10:59] <FrEaKmAn_> nevermind [11:13] <FrEaKmAn_> hi, is there anything for stretching video or duplicating frames? I have a video with white background which is 5 seconds long.. I want to make it 10 seconds long [11:13] <FrEaKmAn_> or 13.. [11:26] <Dark-knight> if your talking about extending the length to your video and adding a white background in the extended section by editing it? then i believe you are unaware that ffmpeg doesn't do that and are in the wrong place. [11:27] <FrEaKmAn_> ow ok [11:27] <Dark-knight> sry man [11:27] <FrEaKmAn_> I could have 1 second video and just concat it to desired length [11:28] <Dark-knight> sure [11:28] <Dark-knight> you could keep adding a white jpg over and over [11:29] <Dark-knight> or you could join 2 video files into one video [11:29] <Dark-knight> http://www.ffmpeg.org/faq.html#How-can-I-join-video-files_003f [11:31] <Dark-knight> just make a duplicate of the one video file you have and rename it, then combine them [11:32] <Dark-knight> http://superuser.com/questions/683643/how-to-add-my-logo-for-the-first-30-s… [11:32] <Dark-knight> this could also help [11:33] <FrEaKmAn_> thanks [11:41] <FrEaKmAn_> one more thing.. what format is the best if I want to do concat, trim, split and audio replacement? I'm looking for a fastest solution [11:41] <FrEaKmAn_> mp4? [11:43] <Dark-knight> mp4 is the least flexible container. use .mkv [11:44] <FrEaKmAn_> ok [11:48] <Dark-knight> what do you plan i using this video for? [11:48] <Dark-knight> plan on* [11:49] <FrEaKmAn_> I'm doing a very simple online video editing solution [11:49] <Dark-knight> sorry [11:49] <FrEaKmAn_> so user uploads multiple videos, defines start and end and hits play [11:49] <Dark-knight> Where are you going to be playing this video file? [11:49] <Dark-knight> what will it be played from? [11:49] <FrEaKmAn_> at the moment, with my flash player thru wowza [11:50] <Dark-knight> i dont think your understanding me [11:50] <FrEaKmAn_> sorry [11:50] <Dark-knight> What hardware are you playing this from? [11:50] <FrEaKmAn_> simple ubuntu 12.04 machine [11:50] <FrEaKmAn_> VPS [11:50] <Dark-knight> ok pc [11:50] <Dark-knight> its a pc [11:50] <FrEaKmAn_> yes yes [11:51] <Dark-knight> .mkv should be fine then [11:51] <FrEaKmAn_> probably you are thinking about osx and mov? [11:51] <sfan5> Dark-knight: the container format does not have anything to do with hardware [11:51] <sfan5> FrEaKmAn_: if you want to be 100 percent compatible I'd advise using .mp4 [11:52] <sfan5> but as long as you know the software it will be player with supports mkv, go with mkv [11:52] <Dark-knight> if all he is going to be using this video on, is a PC then why not use the easier container? [11:52] <FrEaKmAn_> it's my software so... [11:52] <FrEaKmAn_> like? [11:53] <sfan5> Dark-knight: an iPhone is a computer too, does it support .mkv: no [11:53] <Dark-knight> lol an iphone is not a computer [11:53] <sfan5> it is [11:54] <sfan5> any smartphone is a computer [11:54] <Dark-knight> some dude came to a convention with his tablet and said "I brought my computer and for some reason i can't hook it up to this projector" [11:54] <Dark-knight> thats because its not a computer, its a tablet [11:54] <Dark-knight> same with phone [11:54] <sfan5> IT IS A COMPUTER [11:55] <FrEaKmAn_> lol [11:55] <FrEaKmAn_> offtopic [11:55] <sfan5> nobody said a computer needs to have VGA/HDMI connector [11:55] <sfan5> or even an usb port [11:55] <FrEaKmAn_> anyways, thanks for inof [11:55] <FrEaKmAn_> info [11:55] <sfan5> your router is a computer too [11:56] <Dark-knight> alright alright, im just saying that an iphone or tablet can't do everything a PC can do [11:56] <sfan5> I didn't say that [11:56] <sfan5> but theoretically it could do the same [11:56] <sfan5> you can install a full-features Linux system on some Tablets [11:57] <Dark-knight> any major software that a PC runs on will use .mkv [11:58] <Dark-knight> unless your running windows 98 then your pc will play .mkv files [11:58] <sfan5> I don't think Windows supports .mkv out-of-the-box [11:58] <sfan5> and what is a PC? [11:58] <sfan5> PC = computing device with Windows installed? [11:59] <Dark-knight> personal computer [11:59] <sfan5> I'm talking about your definition of PC [11:59] <sfan5> if PC = computing device [11:59] <sfan5> then <Dark-knight> any major software that a PC runs on will use .mkv [11:59] <sfan5> is wrong [11:59] <Dark-knight> major software includes, OSX, Windows, linix, Ubuntu. [12:00] <sfan5> a computing device with absolutly nothing installed won't do anything [12:00] <sfan5> why are you listing both Ubuntu and Linux? [12:00] <Dark-knight> you know what im trying to say and you're still giving me a hard time [12:00] <Dark-knight> lol [12:01] <sfan5> I just want you to know that Ubuntu is Linux [12:01] <sfan5> (basically) [12:01] <Dark-knight> you might get some blowback on that one from other people, but im not going to disagree with you [12:02] <sfan5> I don't there is Ubuntu with kernels not Linux [12:07] <Dark-knight> albeit my original comment, I was incorrectly using the word PC to refer to a computer with windows. container format DOES have something to do with the hardware. [12:07] <Dark-knight> "[05:51] sfan5 Dark-knight: the container format does not have anything to do with hardware" [12:07] <Dark-knight> if he had said he was playing the video from a iphone or ps3 i would have told him that mkv doesn't work and to use mp4. [12:07] <sfan5> that is the software [12:08] <sfan5> if you were to run a better program on the ps3 or iPhone it would play .mkv [12:08] <sfan5> and the codecs matter too [12:08] <sfan5> unless something has hardware decoding [12:08] <sfan5> it will probably not play HEVS [12:08] <sfan5> HEVC* [12:09] <Dark-knight> i thought it was a moot point that different hardware has different software (most of the time) [12:10] <Dark-knight> its obvious that an iphone wouldnt play .mkv [12:10] <sfan5> how is it obvious? [12:10] <Dark-knight> it runs on OSX [12:11] <sfan5> correct [12:11] <sfan5> and how is it obvious that OSX won't play mkv? [12:11] <Dark-knight> my old mac didnt like playing mkv videos [12:11] <sfan5> the codec matter too [12:11] <Dark-knight> had to dl stuff to make it work [12:11] <sfan5> you can put many different video codecs into mkv [12:11] <Dark-knight> i understand that [12:12] <Dark-knight> i was assuming he was going to change the codecs to fit his platform [12:12] <Dark-knight> my bad [12:14] <waressearcher2> anyone ? [12:14] <waressearcher2> how to add running clock in video ? [12:15] <sfan5> http://einar.slaskete.net/2011/09/05/adding-time-stamp-overlay-to-video-str… [12:15] <waressearcher2> I want to speed up 3 hours video into 3 minutes, sort of timelapse, and I want to add running timear at the bottom left corner, is it possible ? [12:16] <sfan5> yes [12:16] <waressearcher2> I want to add fast running timer [12:16] <sfan5> that could be tricky [12:17] <Dark-knight> just add the normal timer and then speed up the video afterwards [12:17] <waressearcher2> Dark-knight: so its twice processing ? [12:17] <waressearcher2> can I speed up added timer ? [12:17] <Dark-knight> did you already speed up the video? [12:18] <waressearcher2> wait, I can do both filters at the same time ? use "-filter:v setpts=0.02*PTS" and it will speed up timer with video ? or in resulting video timer will be normally running ? [12:19] <Dark-knight> i'd do it one at a time [12:19] <waressearcher2> so will it add timer first then speed up it with video or first it will speed up video and then add timer normally running ? [12:19] <Dark-knight> sfan5? [12:20] <sfan5> -filter:v drawtext=text='%T',setpts=0.02*PTS [12:20] <sfan5> that probably works [12:34] <waressearcher2> it says: No such filter: 'drawtext' [12:35] <waressearcher2> I use ffmpeg-2.0.2, where can I get that filter or should I enable when compiling ffmpeg ? [12:37] <john___> Hello. I got a ... weird problem with ffmpeg and hoped you guys could help me there. [12:38] <waressearcher2> looks like I need to rebuild it with --enable-libfreetype option [12:38] <john___> At the beginning I recorded a video with fraps, resulting in a 2.1 GB video file (Windows 7). [12:38] <waressearcher2> john___: then ? [12:39] <waressearcher2> can you write entire story down ? [12:39] <john___> I am trying right now, but it's a bit confusing. :) [12:39] <john___> OK, [12:39] <ubitux> waressearcher2: take the opportunity to pick a more up-to-date, and --enable-fontconfig is also recommended to avoid hardcoding font paths [12:40] <john___> Wait, I am going to reconnect if I wrote it down. Not much sense writing it right now. [12:40] <Dark-knight> ??????? [12:40] <Dark-knight> what does that even mean? [12:42] <anshul> john__ was in trans state [12:42] <anshul> :D [12:42] <Dark-knight> lol [12:52] <Dark-knight> what is the code syntax in a mkv or mp4 container that determines which stream is the "default" stream? [12:52] <Dark-knight> like what inside the coding of the container determines the default stream? [12:53] <luc4> Hello! Im trying to play a variable frame rate h264 annex b stream using ffplay. I see the speed of the video is completely wrong. Is this because there is no timing information inside the stream itself? [12:53] <Dark-knight> id like to know so i can create a command line that changes the default stream without the need to dl extra prgrams or re-mux it [12:53] <Dark-knight> help him first though^ [12:54] <c_14> Dark-knight: http://matroska.org/technical/specs/index.html Down by Track it's called FlagDefault. [12:55] <c_14> luc4: speed as in what you're seeing visually or the fps or what? [12:55] <luc4> c_14: the speed of what I see. [12:56] <c_14> What are you using to play the file and can you pastebin an ffprobe of the file? [12:58] <luc4> c_14: Im using ffplay. Ill pastebin the output. [13:00] <luc4> c_14: http://pastebin.com/tvYher3F [13:04] <Dark-knight> c_14: ok now how do i remove that specification from one stream and apply it to another using ffmpeg? [13:04] <c_14> luc4: Yeah, I'd guess it has to do with ffmpeg not being able to find any timestamps in the input file. It can't even guess a duration. [13:05] <luc4> c_14: so should I conclude that timestamps are not contained into the h264 stream but only in the container? [13:07] <c_14> They are usually in the container, yes. [13:13] <c_14> Dark-knight: I don't think FFmpeg supports setting the default track(at least from what I've found reading the mailing list archives and such). The options for setting default-flags and such seem to set the tag element instead of the track element. [13:13] <c_14> You could try them and see if it's been fixed in the meantime. [13:15] <Dark-knight> ill contact zeranoe and ask him if he could work on that feature [13:37] <Dark-knight> i doubt he will though [13:37] <Dark-knight> in the meantime [13:37] <Dark-knight> somebody else can fix thius [13:37] <Dark-knight> this* [13:39] <Dark-knight> List: Things that need to be done/fixed: add the ALAC tag to the .mp4 container, add support for setting the default track. [13:43] <soffio> hi i've got problem to input a video: "crazy start time, iam scared, giving up" [13:43] <soffio> "Invalid data found when processing input" [13:45] <soffio> http://pastebin.com/raw.php?i=CQxjbzJT [13:46] <klaxa|work> that is neither the command you used, nor is it complete [13:46] <soffio> if you prefer everything: http://pastebin.com/raw.php?i=a31TQYqt [13:46] <soffio> well, it's kind a waste of space [13:47] <Dark-knight> not really [13:47] <Dark-knight> it lets us know how and why the problem happened [13:48] <soffio> seems that only vlc like to play it [13:48] <soffio> mplayer does not, probably it use ffmpeg (?) [13:49] <soffio> mediainfo output: http://pastebin.com/raw.php?i=kNe8kDpc [13:50] <soffio> kind a old codec [13:50] <soffio> XviD 1.1.2 (UTC 2006-11-01) [13:50] <soffio> nvm, works on other video with same codec version [13:52] <Dark-knight> c_14: can you use that in a command line? [13:52] <Dark-knight> don' [13:53] <Dark-knight> i don't really know how to word it [13:53] <sfan5> Dark-knight: ffmpeg -i <file> is a perfectly valid command [13:54] <Dark-knight> what are you talking about sfan5? [13:54] <sfan5> >can you use that in a command line? [13:54] <sfan5> maybe I understood your question wrong [13:55] <Dark-knight> i was addressing an earlier convo with c_14 about something entierly different [13:55] <sfan5> then ignore what I said [13:56] <Dark-knight> i did [13:57] <soffio> so i can't debug this?.. [13:57] <klaxa|work> doubt it [13:58] <klaxa|work> maybe vlc can remux it [13:58] <soffio> nice idea, let me try [14:00] <soffio> klaxa|work: omg, thanks :D working [14:01] <klaxa|work> cool cool [14:07] <heftig-z> I've got two almost identical ffmpeg instances here, both do some complex filtering then push streams to rtmp; one receives input via x11grab and alsa, the other receives a matroska stream over stdin; the latter's memory usage creeps ever higher and higher [14:07] <heftig-z> is there a memory leak in the matroska decoder? [14:10] <klaxa|work> maybe run it with valgrid? (also, would be matroska demuxer, not decoder) [14:14] <waressearcher2> I use that filter line: "drawtext=fontfile=/usr/share/fonts/TTF/DejaVuSans-Bold.ttf:text='%T'" and it says: "Stray % near 'T'", where is the problem ? I used also '\%T' the same [14:16] <sfan5> waressearcher2: try this http://einar.slaskete.net/2011/09/05/adding-time-stamp-overlay-to-video-str… [14:16] <john_____> Hello, here I am again. I wrote down my problem there: http://pastebin.com/1qEBkXtz [14:18] <waressearcher2> sfan5: if I replace '%T' with '%m/%d/%y %H\:%M\:%S' it says: "Stray % near 'm/%d/%y %H:%M:%S'" [14:18] <sfan5> hm [14:20] <sfan5> -vf drawtext='fontfile=/usr/share/fonts/truetype/ttf-dejavu/DejaVuSans-Bold.ttf:text=%{localtime\\:%T}' [14:21] <sfan5> waressearcher2: that one works for me [14:21] <Dark-knight> excellent work john_____ i impressed at your description [14:22] <Dark-knight> although i have no idea how to help you atm [14:22] <Dark-knight> somebody else can [14:22] <sfan5> waressearcher2: you probably want text=%{pts\\:hms} [14:23] <waressearcher2> sfan5: now it says: "%{pts} requires at most 0 arguments" [14:24] <waressearcher2> how can I check with any text not just time ? that is my entire filter: "-filter:v "drawtext=fontfile=/usr/share/fonts/TTF/DejaVuSans-Bold.ttf:text='1111111':fontcolor=white@0.8:x=7:y=460" but there is no "1111111" text in resulting video [14:24] <sfan5> which ffmpeg version? [14:25] <john_____> Dark-knight: Hm, OK. I will just wait here nonetheless. [14:25] <waressearcher2> sfan5: ffmpeg-2.0.2 [14:25] <waressearcher2> just don't tell me to upgrade [14:25] <waressearcher2> for how long that feature --enable-libfreetype there ? [14:25] <waressearcher2> it should work even in 0.6 version [14:27] <sfan5> waressearcher2: the filter your just posted works perfectly for me [14:27] <sfan5> it draws 1111111 at the bottom of the video as expected [14:28] <waressearcher2> no there is actually "111111" in video, I changed "y=460" to "y=260" because wideo is "672x384" [14:28] <sfan5> waressearcher2: did you make sure your video is big enough so that the text is not out-of-bounds [14:28] <sfan5> .. [14:29] <Dark-knight> john_____: if nobody helps just wait a while and then post "Help! (your pastebin URL)" [14:29] <Dark-knight> it alot to read [14:29] <Dark-knight> might take a min for somebody to help you [14:30] <sfan5> john_____: it might be possible that fraps generated data ffmpeg cannot yet fully decode [14:30] <john_____> Dark-knight: that's what I planned to do. :) [14:30] <waressearcher2> but '%T' still not working it says "Stray % near 't'", mabe there is something wrong with my versio of strftime() ? it uses it to get time right ? [14:31] <sfan5> waressearcher2: that would show the time of _encoding_, I'm sure you want the time in video [14:31] <sfan5> there were some changed [14:31] <sfan5> changes* [14:31] <waressearcher2> I want time of the video, yes [14:31] <john_____> sfan5: any suggestions on how I get my footage encoded to h.264 then? [14:31] <sfan5> you would use %{localtime\\:%T} [14:31] <sfan5> <sfan5> waressearcher2: you probably want text=%{pts\\:hms} [14:32] <sfan5> that would show the time in the video [14:32] <john_____> Because all programs that I tried to use on Windows crahed. Literally. [14:32] <sfan5> john_____: convert the footage into ffv1 or huffyuv on windows [14:33] <john_____> And Handbreak wasn't even compilable. How am I supposed to convert them when the convertes I used so far crashed? :) [14:33] <john_____> converters* [14:34] <sfan5> doesn't Fraps provide a .dll to decode the fraps codec? [14:35] <john_____> I don't know. I got three DLLs here: frapslcd.dll, fraps64.dll and fraps32.dll [14:35] <john_____> None of them really make the impression of containing a decoder for me. [14:36] <waressearcher2> sfan5: it prints "23:36:36" in video, but I need sort of a timer from "0:0:0" to "3:0:0" because video is 3 hours long [14:37] <sfan5> john_____: http://www.fraps.com/faq.php 'Where can I find the custom Fraps FPS1 codec?' [14:37] <sfan5> maybe that helps [14:37] <sfan5> waressearcher2: use text=%{pts\\:hms} [14:38] <john_____> sfan5 I got there as well. :) But that does not tell me where it is. [14:40] <john_____> Whatever. I will just look around and try to get it encoding to ffv1 or huffyuv. I will join again if it worked or not. [14:40] <john_____> Bye [14:42] <waressearcher2> sfan5: it says: "%{pts} requires at most 0 arguments" [14:42] <sfan5> try text=%{pts} [14:47] <waressearcher2> sfan5: I got seconds running in form like: "5.250000", but I want them to be like: "2h 12m 30s" or "2:12:30.2000" and counting [14:47] <waressearcher2> is it possible ? [14:47] <sfan5> %{pts\\:hms} [14:47] <sfan5> would do exactly that [14:47] <sfan5> but your ffmpeg seems to be too old [14:47] <waressearcher2> so it should be obvious that its timelapse and it should be clearly seen how minutes and hours are flying [14:49] <waressearcher2> so I will update it later and try it again, and if you wrong and I will return and say "I told you that, it should work even in older version" [14:49] <sfan5> my ffmpeg is built on Jul 10 2014 16:02:44 [14:49] <waressearcher2> "built on Jul 16 2014 22:17:47 with gcc 4.4.4" [14:49] <waressearcher2> so what ? [14:49] <waressearcher2> sfan5: you can build 0.6.3 version right now [14:50] <sfan5> https://www.ffmpeg.org/ffmpeg-filters.html says there is pts:hms [14:50] <sfan5> it works for me [14:50] <sfan5> even if my version is older than yours [14:51] <MrPingouin_> hello world [14:52] <MrPingouin_> hope everyone's good [14:53] <waressearcher2> MrPingouin: what, "MrPinguin" was taken ? [14:53] <waressearcher2> MrPingouin: "MrPenguin" [14:53] <MrPingouin> non, it's in french [14:54] <MrPingouin> the truth is, I initially wanted "monsieur" pingouin [14:54] <MrPingouin> but in french, "monsieur" is "M." [14:54] <MrPingouin> and the dot was not allowed I think [14:54] <waressearcher2> isn't "manchot" is penguin in french ? [14:54] <MrPingouin> yes [14:54] <MrPingouin> BUT [14:54] <MrPingouin> manchot, is also what you called someone without arms [14:54] <jjohn> Ahoy, folks. s [14:54] <waressearcher2> MrPingouin: or you mean "auk" ? [14:55] <MrPingouin> or especially clumsy (if the google translation is ok) [14:55] <MrPingouin> I didn't get you, what's auk ? [14:55] <MrPingouin> oh, it's penguin lol [14:55] <waressearcher2> "pingouin" is "auk" in french [14:55] <jjohn> sfan5: I tried to encode the footage, once as huffyuv, once as ffv1. But I guess I got something wrong in my head. [14:56] <jjohn> Because the errors still appear. [14:56] <MrPingouin> auk or awk, one letter and you're a clumsy animal, or the king of the birds [14:57] <MrPingouin> anyway [14:57] <MrPingouin> I have issues while encoding a video (jpeg2000 max quality) to h264 [14:57] <MrPingouin> the output is very noisy [14:57] <relaxed> command? [14:57] <MrPingouin> in spite of my -qp 0 [14:57] <MrPingouin> command is... [14:58] <klaxa|work> -qp 0 should be lossless [14:58] <sfan5> jjohn: on windows? [14:58] <MrPingouin> ffmpeg -i jpeg2000stuff.mov -c:v libx264 -preset veryslow -qp 0 -c:a copy -f 25 myOutput.mov [14:58] <sfan5> MrPingouin: you probably want -crf [14:59] <relaxed> -f 25 is wrong [14:59] <jjohn> On windows, and with the encoders I used so far, I don't even have these options. On linux ... well, it doesn't matter here because the FPS doesn't seem to work. [14:59] <MrPingouin> sfan5: oh, I'll try then. I was following the h264 faq on trac.ffmpeg.org [14:59] <MrPingouin> You can use -qp 0 or -crf 0 to encode a lossless output. Use of -qp is recommended over -crf for lossless because 8-bit and 10-bit x264 use different -crf values for lossless [14:59] <sfan5> jjohn: you might be able to use directshow to decode the fraps stream with fraps own decoder [14:59] <MrPingouin> trying this right now [15:01] <jjohn> sfan5 could it be that directshow is included in the K-Lite codec package? [15:01] <jjohn> Sorry, I am still a newby with this decoder businiss. [15:02] <jjohn> business* [15:04] <jjohn> But I always thought that K-Lite was used to update all codecs that might be installed on one's PC, because most installers won't check for the codec version (also see DLL hell) [15:05] <sfan5> jjohn: hm, seems like what I suggested is not possible [15:05] <jjohn> Oh, OK. I was already doubting my own sanity. :) [15:06] <jjohn> Another suggestion? [15:07] <MrPingouin> there's no "-crf" in ffmpeg doc! [15:07] <MrPingouin> what are the possible values ? [15:08] <klaxa|work> MrPingouin: https://trac.ffmpeg.org/wiki/Encode/H.264 [15:09] <MrPingouin> oh thanks, it was right in front of me [15:10] <jjohn> help: http://pastebin.com/1qEBkXtz [15:13] <c_14> jjohn: Have you tried using an up to date version of FFmpeg? [15:15] <relaxed> jjohn: install ffmpeg-9999 from portage or try http://johnvansickle.com/ffmpeg/ [15:19] <MrPingouin> hmmmm. [15:19] <MrPingouin> style noisy [15:19] <MrPingouin> still [15:19] <MrPingouin> sorry [15:20] <MrPingouin> less noisy than with -qp 0 (I set -crf 18) [15:20] <klaxa|work> is the source material noisy? [15:20] <MrPingouin> nope [15:20] <MrPingouin> but [15:20] <MrPingouin> there's an interesting fact [15:20] <MrPingouin> I asked my colleague to produce me two video source [15:20] <MrPingouin> 1. quicktime animation 100 (huuuuge file) [15:21] <MrPingouin> 2. jpeg2000 quality max (supposed to be lossless too) [15:21] <MrPingouin> same command line, different sources [15:21] <MrPingouin> 1. perfect [15:21] <MrPingouin> 2. noisy [15:21] <MrPingouin> the thing is [15:21] <klaxa|work> maybe the jpeg2000 decoder is buggy [15:21] <MrPingouin> as it was noisy, we used a computer with adobe premiere [15:21] <MrPingouin> and it has produced a non-noisy h264 file, but the process time was huge [15:25] <relaxed> MrPingouin: try the libopenjpeg decoder [15:25] <relaxed> ffmpeg -vcodec libopenjpeg -i input ... [15:27] <MrPingouin> exactly what I typed 10 sec ago :) [15:27] <MrPingouin> I found the trac ticket 2683 [15:27] <MrPingouin> note : I'm currently using the nightly build [15:28] <relaxed> typed where? [15:28] <MrPingouin> in my console [15:29] <MrPingouin> muuuuuuuuuuuuuuch better [15:29] <MrPingouin> thanks a lot relaxed [15:29] <relaxed> you're welcome [15:29] <MrPingouin> crf 10, I guess I can try 18 [15:30] <MrPingouin> I'm using the pix_fmt yuv420p for quicktime player to read it, but it can't read the file in realtime, vlc can [15:30] <MrPingouin> video format is 1080p [15:33] <jjohn> c_14: not yet. relaxed: johnvansickle does not provide the sources, and I'd like to compile it on my own machine. [15:33] <jjohn> I am going to find out how to manually install it. [15:34] <c_14> https://trac.ffmpeg.org/wiki/CompilationGuide/Generic [15:35] <jjohn> c_14 I thought about that but I do not want to mess up the portage tree. [15:35] <relaxed> jjohn: That's my site, and sources are available upon request. If you're going to compile it you might as well use portage. [15:35] <c_14> how would that mess up the portage tree? [15:36] <jjohn> Because ffmpeg is already installed, and portage knows that. And if I compile and install a vanilla version, future updates might break. [15:37] <jjohn> I once had a guy explaining that to me far and wide. [15:37] <c_14> you don't have to globally install it [15:37] <c_14> Just install it in a local dir. [15:37] <klaxa|work> compile separately, run with /whatever/prefix/you/used/ffmpeg [15:39] <jjohn> Damn, I need some sleep ... of course I can do that. [15:45] <heftig-z> klaxa|work: valgrind doesn't seem to find the heap ffmpeg uses [15:48] <jjohn> c_14: ./configure --prefix=/opt/local/ffmpeg && make -j9 && make install && /opt/local/ffmpeg/bin/ffmpeg -i input.avi -c:v libx264 -c:a copy output.mov [15:48] <MrPingouin> next step, setting a maxrate for quicktime player [15:48] <jjohn> Shows the error: Unknown encoder 'libx264'. Do I have to install that codec manually as well? [15:48] <c_14> you have to enable it in the build [15:49] <c_14> --enable-libx264 should be enough if you have x264 installed. [15:50] <jjohn> c_14 I am already compiling it. But you need to specify --enable-gpl as well. :) [15:55] <jjohn> c_14 thank you very much! Using the latest ffmpeg version did the trick. :) [15:56] <jjohn> The video and audio now runs flawlessly. [15:59] <jjohn> Although mplayer still complains that my machine is too slow. And the "picture is too late to be displayed (missing X ms)" messages are still there. [16:01] <klaxa|work> does that also happen with ffplay? [16:05] <jjohn> klaxa no, it doesn't. [16:06] <jjohn> What would it mean if it had? [16:06] <klaxa|work> dunno [16:07] <klaxa|work> this means that mplayer is decoding slower apparently [16:07] <klaxa|work> or rather, ffplay is not telling you that it is decoding too slow? [16:08] <klaxa|work> have you tried some of the suggestions mplayer gives you to increase playback speed? [16:08] <sfan5> klaxa|work: I don't think ffplay has 'slowness' detection [16:08] <klaxa|work> like disabling audio or stuff [16:08] <klaxa|work> sfan5: yeah that's what i thought afterwards too [16:10] <jjohn> No, I guess I know what the difference is: mplayer is single-threaded, ffplay multi-threaded. ffplay renders the file fast enough, but uses two cores. mplayer is not that fast and uses only one core. [16:11] <klaxa|work> maybe try out mpv or mplayer2 [16:11] <klaxa|work> (mpv is what the cool kids use these days) [16:11] <jjohn> If I wanna be cool I'd use Arch. ;) [16:12] <klaxa|work> i... i run arch [16:12] <MrPingouin> lol [16:12] <jjohn> Well, congrats. You are a cool kid then. [16:12] <MrPingouin> buuusted [16:13] Action: sfan5 runs (X)ubuntu [16:13] <sfan5> :c [16:13] <jjohn> No, I guess that's how fast my linux can render the file without VDPAU. But at the moment it's enough for me. [16:13] <jjohn> sfan5 tell me you at least compile your own kernel. [16:14] <sfan5> no [16:14] <sfan5> but I compile my own ffmpeg [16:14] <sfan5> and most of the libs it needs [16:14] <sfan5> and I use mpv [16:14] <jjohn> Well, I compile my whole system on my own. :) [16:14] <klaxa|work> who's the hipster now? [16:14] <sfan5> :c [16:14] <jjohn> But I am still using vlc. :/ [16:14] <sfan5> ew vlc [16:15] <jjohn> >inb4 not hipster enough [16:15] <klaxa|work> i think that's enough circlejerking for a support channel [16:15] <sfan5> yeah [16:16] <jjohn> Right. However, thanks for the help! I now have a way to encode my footage. [16:16] <jjohn> Bye. [16:20] <waressearcher2> are libx265 and libx264 two different codecs or they will be merged into one one day ? [16:21] <klaxa|work> they are different [16:21] <klaxa|work> and will most likely never be merged [16:22] <sfan5> waressearcher2: libx265 is for HEVC and libx264 for H.264 [16:23] <waressearcher2> I noticed in ffmpeg 2.3 there is new option "--enable-opengl" what its for ? [16:23] <waressearcher2> is it accelerate things by using video card ? [16:23] <waressearcher2> I now compile 2.3 should I enable it ? [16:24] <klaxa|work> if anything it would be opencl, opengl is not for general purpose computing but hardware accelerated rendering so it's maybe used by ffplay? [16:24] <klaxa|work> check the changelog maybe? [17:00] <waressearcher2> sfan5: and here we go, you here ? [17:00] <sfan5> yes [17:03] <waressearcher2> sfan5: I tryed to build 2.3 version there http://sprunge.us/fAZf at the begining is compilation options and after ============ is output with errors [17:03] <waressearcher2> so 2.0.2 compilled fine but 2.3 cased errors [17:04] <waressearcher2> I assume they are specific to libxvid so maybe I can update that libxvid or something or is it the part of ffmpeg package ? [17:04] <waressearcher2> I don't want to disable it [17:05] <sfan5> waressearcher2: can you try compiling the development version of ffmpeg? [17:05] <sfan5> (from git) [17:06] <waressearcher2> how can I get it from git ? [17:06] <waressearcher2> is there tarball or should I use git client ? [17:06] <waressearcher2> sfan5: if you give me command of downloading from git [17:07] <sfan5> git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg [17:09] <waressearcher2> I also have packages installed xvidcore-1.2.2-i486-1 and xvidtune-1.0.2-i486-1, should I update them ? [17:09] <waressearcher2> are they related ? [17:10] <sfan5> maybe [17:10] <heftig-z> the memory use of this pipeline grows steadily: [17:10] <heftig-z> ffmpeg -v quiet -f lavfi -i testsrc -f matroska -c:v rawvideo -pix_fmt yuv420p - | ffmpeg -i - -f matroska -c:v copy -y /dev/null [17:11] <heftig-z> is that normal? [17:11] <heftig-z> er, hang on [17:11] <heftig-z> actually have a simpler version [17:11] <heftig-z> ffmpeg -v quiet -f lavfi -i color -f nut -c:v copy - | ffmpeg -i - -f rawvideo -c:v copy -y /dev/null [17:17] <waressearcher2> by the way I have similar to heftig-z question: I have 1GB of ram, and I was trying to make slideshow video from photos, there were 90 photos of 1920x1080 resolution, and I tryed to make video of 1920x1080 using h264 codec, I used that command: "cat *.png | ffmpeg -f image2pipe -vcodec png -r .4 -i - -vcodec h264 -vb 3000k -r 25 out.avi" and after [17:17] <waressearcher2> I started process it started to consume memory and after it consumed about 900MB it was killed by system, and I got message in dmesg: "Out of memory: kill process 20688 (ffmpeg) score 116212 or a child" why it was killed ? [17:17] <Hello71> heftig-z: well, you output one format [17:17] <Hello71> er, never mind [17:18] <Hello71> waressearcher2: get ffmpeg to open the images itself [17:18] <waressearcher2> on the other hand [17:18] <waressearcher2> Hello71: [17:19] <waressearcher2> and on the other hand, I tryed make "1920x1080 h264" video from pieces, the difference from previous case I used video pieces as a source instead of photos, the command was: "ffmpeg -f concat -i list.txt -vcodec h264 -vb 3000k out.avi" and it did it fine, while processing it it consumed only 437MB of RAM and haven't been killed, so why when I [17:19] <waressearcher2> tryed to make "1920x1080 h264" video out of photos it lacked RAM and was killed but there was no problems in second case of converting video pieces to "1920x1080 h264" video ? [17:19] <Hello71> probably because ffmpeg is reading in all the images before processing them [17:20] <Hello71> also, use mp4 container [17:20] <waressearcher2> Hello71: instead of "-f avi file.avi" use "-f mp4 file.mp4" ? why is it better ? [17:20] <waressearcher2> because some rules are ? [17:21] <Hello71> don't use image2pipe [17:21] <Hello71> get ffmpeg to open the images itself [17:22] <waressearcher2> Hello71: that way: "-i %05d.png", the thing is I want to have full control on what images I want to use [17:23] <waressearcher2> Hello71: with "cat *.png | ffmpeg" I can use images from many sources like: "cat _/{0001..10000..500}.png /tmp/{1..200}.png | ffmpeg" [17:24] <Hello71> 1. you'll run out of ARG_MAX fast [17:24] <waressearcher2> Hello71: without pipe using "-i %05d.png" I need to stack all images and rename them consequently [17:24] <klaxa|work> you can use -i *.png [17:25] <waressearcher2> klaxa|work: and what if I have images like 00538.png ? should I rename them all without leading zeroes ? [17:25] <waressearcher2> no [17:26] <heftig-z> ffmpeg -f image2 -pattern_type glob -i '*.png' -r .4 [17:27] <waressearcher2> Hello71: I remember did same piping with 10000 png files of about 650x350 resolutino and it didn't run out of RAM [17:27] <waressearcher2> Hello71: and didn't run out of command line arguments [17:40] <OnkelTem> Hi all [17:40] <OnkelTem> I can't get ffmpeg to read rtsp stream, all I see is a bunch of "[NULL @ 0x338acc0] RTP: missed 212 packets" messages [17:41] <OnkelTem> My line is: ffmpeg -re -i rtsp://web:q1w2e3r4@185.8.127.10:554 -c copy -f flv rtmp://localhost:1935/mytv/go [17:41] <OnkelTem> what I do wrong? [17:44] <OnkelTem> My line is: ffmpeg -re -i rtsp://web:q1w2e3r4@185.8.127.10:554 - this part is wrong, right? Why? [17:45] <waressearcher2> sfan5: you here ? [17:46] <waressearcher2> sfan5: for that fresh git version it gives same error about libxvid http://sprunge.us/QNdV [17:46] <waressearcher2> I don't know if I need that libxvid, I usually use mpeg4 of h264 codecs [17:48] <waressearcher2> or wait, I will try with updated xvidcore [17:49] <waressearcher2> so much hassle for a timer on top of video [17:50] <waressearcher2> the version my xvidcore-1.2.2 dates back to Jun 13 2011 [18:01] <_Vi> Filed a bug: https://trac.ffmpeg.org/ticket/3773 [18:02] <waressearcher2> _Vi: its like: "anyone here want to fix that bug" querry ? [18:04] <_Vi> Is this bug already known? [18:09] <Justice> i have a videofile that has the wrong lenght is there a way to manually set the lenght ? [18:10] <Justice> length* [18:10] <_Vi> Justice, Maybe remux it? ffmpeg -i youvideo.mkv -acodec copy -vcodec copy newvideo.mkv [18:11] <Justice> _Vi: doesnt that reencode it ? [18:12] <_Vi> Justice, No, only demux and mux. "-vcodec copy -acodec copy" is to tell ffmpeg not to re-encode. [18:20] <Justice> _Vi: hm still wrong length [18:22] <_Vi> Justice, How do you measure the length? Does "ffmpeg -i newvideo.mkv" show the correct length? [18:23] <_Vi> Justice, Does the file plays to its end if you play it in vlc or mplayer or ffplay? [18:24] <Justice> _Vi: ffmpeg -i shows wrong length but i fixed it finaly =) [18:25] <Justice> when i tried without the copy it said that aac codec is experimental and did not want to continue [18:25] <Justice> ffmpeg -i infile -acodec copy -vcodec copy -t 02:09:41.023 outfile seems to fixed it [18:26] <Justice> though the file is smaller i wonder why [18:26] <_Vi> Justice, "without copy" means you are re-encoding. As for AAC, you can either force ffmpeg to use experimental codec, or choose another aac codec implementation (maybe "-acodec libfaaac"). [18:26] <_Vi> Much smaller or few kilobytes smaller? [18:27] <Justice> _Vi: from 1.8GiB to 1.6GiB [18:27] <Justice> The video was "7hours long" but should be ~2hours [18:27] <_Vi> Is it after "-vcodec copy -acodec copy -t"? [18:27] <Justice> yea [18:28] <_Vi> Probably everything after 02:09:41.023 got trimmed. [18:28] <Justice> hm i thought it was just a file error but i guess [18:30] <Justice> _Vi: thanks for your help =) [18:34] <waressearcher2> _Vi: "-vcodec copy -acodec copy" is equals to "-c copy" right ? [18:34] <_Vi> -c ? [18:34] <_Vi> Probably. [18:41] <OnkelTem> Ok, I managed to get ffmpeg to read rtsp stream and convert it to rtmp [18:41] <OnkelTem> But it exits after video starts in few seconds with message: [18:41] <OnkelTem> Application provided invalid, non monotonically increasing dts to muxer in stream 0: 20840 >= 20840 [18:41] <OnkelTem> av_interleaved_write_frame(): Invalid argument [18:42] <OnkelTem> How to make it NOT quit on such silly errors? [18:43] <OnkelTem> Googling I found posts about similar issues and there people recommends using option: -fix_sub_duration [18:44] <OnkelTem> but for some reason my ffmpeg claims it doesn't recognize it [18:44] <OnkelTem> Any ideas why? [18:44] <OnkelTem> ffmpeg version 0.10.12-7:0.10.12-1~precise1 [18:44] <_Vi> OnkelTem, I remember just commenting out this check from source code. [18:45] <OnkelTem> is my version outdated? [18:45] <JEEB> quite [18:46] <OnkelTem> Oh... I knew that! [18:46] <OnkelTem> So how can I install fresh ffmpeg on Ubuntu? [18:47] <_Vi> OnkelTem, 1. Connect some PPA; 2. install ffmpeg from PPA? [18:47] <OnkelTem> or what would be the best option in my case? I don't need ffmpeg systemwide as I use it on server just to recode a video stream [18:47] <JEEB> build it yourself or grab a static binary if you only need the command line app [18:47] <waressearcher2> OnkelTem: get it from git and compile [18:47] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=commit;h=7e16636995fd6710164f7622cd… <- this is how old 0.10.x is btw :P [18:47] <OnkelTem> _Vi: but that's exacty whay I did. I followed the link to PPA from https://www.ffmpeg.org/download.html [18:47] <_Vi> OnkelTem, Do you want pre-built ffmpeg binary? [18:47] <OnkelTem> JEEB: I see :) [18:48] <JEEB> yeah, PPAs usually try to be replacements :P So they are limited to the old versions if your distro bases itself on older APIs [18:48] <waressearcher2> why they renamed files "INSTALL LICENSE README" to "INSTALL.md LICENSE.md README.md" ? [18:48] <OnkelTem> _Vi: any, I just need to run it on Ubuntu 12.04. Be it static or not - not really care [18:48] <JEEB> there's static binaries' downloads listed on ffmpeg.org [18:48] <JEEB> try those if you only need the command line tool [18:48] <OnkelTem> ok, thanks [18:48] <_Vi> waressearcher2, Because they are in Markdown and are rendered as such if named *.md? [18:49] <waressearcher2> after waiting for 40 minutes for compilation to end it failed because couldn't find ""INSTALL LICENSE README" files [18:49] <_Vi> OnkelTem, There are http://vi-server.org/pub/ffmpeg2013 and http://vi-server.org/pub/ffmpeg2012 . I also have ARM version and (for lulz) version for DOS. [18:49] <waressearcher2> I mean waiting for script to end build ffmpeg package, now I need to restart and wait another 40 minutes [18:50] <JEEB> you have a funky script then :P [18:50] <waressearcher2> _Vi: what is Markdown ? [18:51] <_Vi> waressearcher2, It's a wiki-like document format with [this sort of links](http://example.com). [18:51] <_Vi> waressearcher2, It is used, for example, on Github and on stackoverflow.com. [18:53] <OnkelTem> URA!!! [18:53] <OnkelTem> http://neganov.ru/home.html - video! [19:47] <waressearcher2> sfan5: you here ? [19:49] <Schalla> Hello folks! First of all, thanks a lot to the devs for creating ffmpeg, doing some video slicing and its just amazing fast & stable. I got a small issue, I basicly want to use instead -ss 00:15:32 -t {DURATION} the endtimestamp, so from -ss to -to, that is possible, or I must do a little math on my own?= [19:50] <waressearcher2> sfan5: I updated xvidcore to 1.3.2 and now there is no errors about libxvid, also now with ffmpeg 2.3 version that option "text='%{pts\\:hms}'" is working [19:52] <c_14> Schalla: what exactly do you want? I don't understand the question. [19:53] <waressearcher2> c_14: instead of length to use end position [19:53] <c_14> ye, just use -to [19:54] <Schalla> c_14: That doesn't work, it seems [19:54] <Schalla> Sec [19:54] <c_14> can you pb the commandline [19:54] <Schalla> Option to (record or transcode stop time) cannot be applied to input file [19:54] <Schalla> sure, sec. [19:55] <Schalla> http://pastebin.com/FjcauTZA [19:55] <c_14> -to is only an output option [19:55] <c_14> and if you use -ss as an input option it moves the start time by that point so -to turns into time+seek [19:56] <Schalla> ah well, alright. Guess the easiest thing I could do would be do quick the math and just stay with -t, since I only have to slice 5 parts. [19:57] <Schalla> Thanks for your help and have a nice evening! [20:03] <tesh> *-0J1 [20:03] <tesh> warning [20:03] <tesh> you may be watched [20:03] <tesh> do usa&israel use the internet(facebook,youtube,twitter, chat rooms ..ect)to spy?? [20:03] <tesh> do usa&israel use the internet 2 collect informations,,can we call that spying?? [20:06] <waressearcher2> tesh: try #hardware [20:21] <klaxa> after snowden this is general knowledge [20:30] <waressearcher2> are there ways,tricks,filters to improve video ? [20:30] <waressearcher2> or all I can do is change hue,saturation,contrast ? [20:33] <klaxa> waressearcher2: http://www.ffmpeg.org/ffmpeg-filters.html#toc-Video-Filters [20:33] <klaxa> if you are further interested, look at vapoursynth and/or avisynth [20:33] <waressearcher2> any filters made specifically to improve video make it more vibrant ? [20:34] <klaxa> increasing saturation should make it more "vibrant" [20:38] <JEEB> lol [20:46] <brontosaurusrex> waressearcher2, if we are talking a camera like RED or something that produces raw or close to raw data, then there is bunch of trciks [20:46] <brontosaurusrex> tricks* [20:46] <brontosaurusrex> none involves ffmpeg [20:47] <waressearcher2> brontosaurusrex: so you are last dinosaurus standing ? [20:48] <brontosaurusrex> waressearcher2, no, linux ppl told me that "b" is too short for a nick [20:49] <waressearcher2> [20:49] <waressearcher2> your is longer [20:49] <brontosaurusrex> i still have better ideas than searching for a superfancy sharpener [20:49] <brontosaurusrex> sometimes at least [20:50] <waressearcher2> what is hue ? [20:51] <brontosaurusrex> radial position of color in colorwheel < out of my head [20:51] <waressearcher2> there is an option "hue[=hue:saturation]" so I tryed "mp=hue=5:1" but nothing was changed [20:51] <waressearcher2> but on the other hand "mp=hue=0:2" with increased saturation changed look and feel of video, made it warmer [20:53] <waressearcher2> brontosaurusrex: ""b" is too short" so you wanted like: "Float like a butterfly, sting like a b" ? that "b" ? [20:53] <klaxa> have you tried hue=80:0 or something? [20:53] <waressearcher2> no [20:53] <klaxa> that should move the colors around quite a bit [20:53] <JEEB> my personal opinion is that unless you are the one having the original content on your hand and doing the mastering as a part of the production, then you should keep your hands off the content unless something is actually wrong [20:54] <klaxa> sounds about right [20:56] <klaxa> i want to write that cropping and scaling should be fine, but that can be done at playback time anyway [20:56] <JEEB> well, stuff like cropping is something that you should do when recompressing [20:57] <JEEB> and resizing is something that might be needed for specific plastic toys [20:57] <JEEB> and IVTC/deinterlacing is at times needed for some content [20:57] <klaxa> those probably can't play back at the original bitrate anyway so re-encoding to a lower bitrate is inevitable [20:58] <JEEB> anyways, I think I've had like two cases where I've had to touch the colors of a source: a company had mastered their content with either the top or the bottom of the YCbCr values going full range instead of the usual limited range values [20:58] <JEEB> I felt sorry for the people who had actually watched the stuff on a TV or a dvd/blu-ray player [20:58] <Fjorgynn> :D [20:58] <JEEB> since those would just cut off the content over/under the limited range values [20:59] <JEEB> the second case was actually similar, but not fixable because the company had decided to limit values to valid range at the end [20:59] <JEEB> so you were actually lacking the values [21:00] <JEEB> (meanwhile the TV spot for the same thing had a completely correct range, no idea how the hell they did that) [21:00] <waressearcher2> klaxa: hue of 80 made yellow into green [21:01] <waressearcher2> so its just moving colors around ? [21:01] <waressearcher2> like shifting pallete ? [21:01] <klaxa> yes, did you google hue? http://en.wikipedia.org/wiki/Hue [21:03] <waressearcher2> no [21:04] <waressearcher2> there is that "drawtext" parameter and it has options: "box=1:boxcolor=black" is there a way to make that box bigger than text ? [21:04] <waressearcher2> now its just a width and heigth of a text no more [21:05] <JEEB> if you need fancy text subtitling just use ASS [21:05] <JEEB> aegisub is the app to create those subtitles with [21:05] <JEEB> and then use ffmpeg's ASS rendering filter to draw them on your video surface [21:09] <waressearcher2> JEEB: why are you writing ass in upper case ? [21:09] <JEEB> Advanced SubStation [21:10] <JEEB> (if I recall correctly) [21:13] <waressearcher2> just call it ass in context of the channel it will be understood correct way [23:47] <IamTrying> Remotely i am in a enterprise network connected. But i need to make 2 video clips into 1 video clip. And i do not have FFmpeg only i have VLC. How can i merge 2 video and make it as 1 ? [23:48] <c_14> either download a static build of FFmpeg, or ask in #videolan [23:51] <IamTrying> c_14: OK - somehow using proxy server i was able to install the apt-get install ffmpeg, how can you now without breaking video/audio codecs just merge 2 video into 1? (untouched the video /audio codec cause it was made using Sony/Windows tools therefore on Linux the encoder/decoder breaks the quality) [23:52] <c_14> https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join,%20merge)%20med… [23:53] <c_14> Though if you installed "ffmpeg" via apt-get you actually installed libav instead. You might be better of grabbing the static build and running that. [23:55] <IamTrying> c_14: OK great. So this won't modify my existing audio/video codecs right? e.g: ffmpeg -i "concat:input1.mpg|input2.mpg|input3.mpg" -c copy output.mpg [23:56] <c_14> that command will only work if you actually have mpeg streams, but yes it won't modify the settings. You can add -map 0 if there are multiple audio/video/subtitle streams that you want. (Also, the concat demuxer is more reliable than the concat protocol so you might want to use that - and you will probably have to for all non-mpeg streams). [23:58] <IamTrying> OK - c_14 thank you very much [00:00] --- Thu Jul 17 2014

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