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October 2012

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ffmpeg-devel.log.20121025
by burek 26 Oct '12

26 Oct '12

[00:00] <saste> BTW i'll push ffescape if no one will review/comment in a day [00:01] <saste> people is having an hard time at getting how to escape weird filenames [00:01] <llogan> The Escapist. [00:01] <saste> maybe i should put an escaping intro in filters.texi [00:02] <saste> saste houdini [00:02] Action: llogan recalls some problem with commas in file names...maybe. [00:02] <saste> look ma, i can escape a filename tied to a filterchain [00:03] <saste> llogan: you have to keep in mind that there are two levels of escaping [00:03] <saste> plus the shell escaping [00:04] <cone-609> ffmpeg.git 03Clément BSsch 07a96b39de6225: lavf/srtdec: simplify start/end computation. [00:04] <cone-609> ffmpeg.git 03Clément BSsch 0760d9ee1b75b3: lavc/utils: make sub decode consistent with A/V. [00:04] <cone-609> ffmpeg.git 03Clément BSsch 074d46fd0b3ed1: lavc: add AV_PKT_DATA_SUBTITLE_POSITION side data type. [00:04] <cone-609> ffmpeg.git 03Clément BSsch 07c27b3816e4a7: srt: make the demuxer output SubRip packets. [00:04] <ubitux> commits with "long" descriptions like you seem to appreciate saste ^ [00:05] <saste> well why don't you write lavf/srt then? ;-) [00:06] <ubitux> cause it was a bit related to codec [00:06] <saste> also my religion is against the trailing dot in the first line [00:06] <ubitux> (ok that's a lame excuse) [00:06] <ubitux> ok :( [00:07] <saste> all trasgressors burn in hell for 2^20 years [00:07] <saste> (which is infinitely better than burning in hell for all the eternity) [00:08] Action: ubitux feels lucky [00:13] <cone-609> ffmpeg.git 03Clément BSsch 07fd090a5d0923: lavf/srtenc: simplify timing printing. [00:13] <cone-609> ffmpeg.git 03Clément BSsch 072ecf9492ff3b: lavf/srtenc: set codec to subrip by default. [00:30] <cone-609> ffmpeg.git 03Michael Niedermayer 07aa4782134459: cmdutils: remove writes in never read variable [00:30] <cone-609> ffmpeg.git 03Michael Niedermayer 0758c2c17f1dd2: mov_probe: use correct variable [00:47] <michaelni> ubitux, saste theres a new memleak in lavfi.c (CID739866) [00:47] <saste> new? [00:47] <saste> michaelni, i still have no coverity account [00:48] <ubitux> saste: http://ubitux.fr/pub/pics/_lavfi-leak.png [00:49] <ubitux> saste: have you really checked your spambox? [00:50] <michaelni> saste, nothing in your spam folder ? [01:01] <llogan> i see no saste in "owners" [01:02] <llogan> oh, duh. that's different [01:02] <llogan> i don't know. </ignorant> [01:04] <saste> fuck i deleted all spam by chance [01:05] <saste> but i checked before and couldn't find anything related to "coverity" [01:05] <Daemon404> nice job [01:07] <saste> ok but *that* fix is easy [01:11] <michaelni> saste, ive requested a new account to be created for your gmail address and a seperate one for your poste.it address [01:11] <michaelni> i hope one will go through [01:12] <saste> and yet none arrived... i'll check my poste.it account (i'm not even sure it is still enabled, they were going to close email support) [01:14] <michaelni> well, its just one 2 minutes since so ... [01:39] <cone-609> ffmpeg.git 03Michael Niedermayer 070008e0d6321d: cmdutils: fix unclosed file on error [01:39] <cone-609> ffmpeg.git 03Michael Niedermayer 07ed68085104c2: asfdec: fixed signedness in comparission [01:39] <cone-609> ffmpeg.git 03Michael Niedermayer 074a2297294fa2: qt-faststart: check return of ftello() [01:39] <cone-609> ffmpeg.git 03Michael Niedermayer 07a1af505d6640: roqaudioenc: remove dead code [01:39] <cone-609> ffmpeg.git 03Michael Niedermayer 07fa48da1ee9ce: ffmpeg: fix null ptr deref in psnr printing code [01:52] <llogan> what is the "dmo" in "ffmpeg-dmo" in debian-multimedia? [02:00] <Daemon404> DMO on windows is directshow related [02:00] <Daemon404> no idea what it means for linux [03:00] <durandal_1707> michaelni: why voc demuxer split packets into smaller ones, this makes seeking almost impossible to implement [03:03] <michaelni> i dont know/remember without searching/reading old ML threads, maybe the packets where too large or maybe theres no reason [03:04] <durandal_1707> max possible packet size is 16mb [03:05] <michaelni> thats a bit large [03:05] <michaelni> for audio [03:12] <durandal_1707> michaelni: so how than to make seeking possible? [03:15] <michaelni> if the packets really are that large, then some index can be build that stores whatever [03:15] <michaelni> is needed [03:17] <durandal_1707> and there are functions that can store at least one int per index? [03:27] <durandal_1707> michaelni: the only thing required is the remaining size of chunk for each packet - and i do not see nice way to store it [03:28] <michaelni> you could store the whole packets in the index so the packet size is the remaining size [03:30] <durandal_1707> index is AVIndexEntry? [03:42] <michaelni> probably easiest to use, but something else could be used too in theory [04:17] <cone-609> ffmpeg.git 03Michael Niedermayer 071a535fc477cd: fate: reenable some recently lost audio tests [04:17] <cone-609> ffmpeg.git 03Michael Niedermayer 07d312ffdd79bd: mpegvideo: fix lowres on field pictures [08:22] <nevcairiel> ubitux: how would msys figure out that it needs to replace a string you printed into a file? [08:24] <nevcairiel> it only translates the path if its calling a win32 app, it doesnt translate it when calling a msys builtin [08:26] <nevcairiel> easiest solution would be to just fix the code in the fate script that tries to detect an absolute path [08:27] <nevcairiel> why not simply check if the path exists, and if it does be happy [08:27] <nevcairiel> if it doesnt, try to check if we need to pretent the context [08:27] <nevcairiel> prepend* [08:28] <nevcairiel> that way i could specify a real windows path, and wouldnt need msys conversion [08:53] <ubitux> < nevcairiel> it only translates the path if its calling a win32 app, it doesnt translate it when calling a msys builtin // "ffmpeg.exe" is treated differently than "ls.exe"? [08:55] <nevcairiel> actually, yes [08:55] <nevcairiel> ls.exe is a msys app, while ffmpeg.exe in those fate machines is a native win32 app [08:56] <ubitux> and so, ls.exe /d/foo/bar will receive "/d/foo/bar", while ffmpeg.exe will receive "d:\foo\bar"? pretty weird [08:57] <nevcairiel> pretty much [08:57] <ubitux> mmh. [09:00] <ubitux> nevcairiel: printf '#!/bin/sh\nprintf $@' > ffprintf && chmod +x ffprintf && ./ffprintf '/d/foo/bar' [09:00] <ubitux> what does this print? [09:00] <ubitux> does it needs to be a program? [09:01] <ubitux> if i do the same with a ffprintf.c built with mingw, will it print "d:\foo\bar"? [09:03] <ubitux> printf '#include <stdio.h>\nint main(int ac,char**av){if(ac==2)puts(av[1]);return 0;}' > ffprintf.c && gcc ffprintf.c -o ffprintf && ./ffprintf '/d/foo/bar' [09:03] <ubitux> i'd be curious about what happens in this case [09:05] <ubitux> and sure if you think we should fix the absolute path detection, why not [09:05] <ubitux> unfortunately i can't help much here [09:07] <nevcairiel> the first prints /d/foo/bar [09:07] <nevcairiel> as does the second [09:07] <ubitux> huh? [09:08] <ubitux> then when is supposed to happen the replace? [09:08] <nevcairiel> when you call a win32 app [09:08] <nevcairiel> let me build the second with msvc [09:10] <nevcairiel> build with msvc, the second outputs d:/foo/bar [09:10] <nevcairiel> like i expected [09:10] <ubitux> right ok [09:10] <ubitux> so. [09:11] <ubitux> if i add this stupid ffputs to the tools [09:11] <ubitux> it will work, right? :) [09:12] <ubitux> (yes I know it's ugly) [09:15] <nevcairiel> if that tool is build with msvc during the fate process, it will do what you want [09:16] <ubitux> i guess all the tools are [09:16] <ubitux> at least i hope so :p [09:16] <nevcairiel> one would think so [09:16] <nevcairiel> what else would be there to build it [09:16] <nevcairiel> anyway, meeting [13:26] <cone-621> ffmpeg.git 03Martin Storsjö 07c44cef978bd5: smoothstreamingenc: Don't assume streams start from timestamp 0 [13:26] <cone-621> ffmpeg.git 03Mans Rullgard 07d4c99513f412: configure: generalise 64-bit test [13:26] <cone-621> ffmpeg.git 03Mans Rullgard 072acda282eb8e: configure: detect mips64 automatically [13:26] <cone-621> ffmpeg.git 03Mans Rullgard 0756203596aea3: configure: detect ppc64 automatically [13:26] <cone-621> ffmpeg.git 03Mans Rullgard 07a6e9d6497739: configure: detect parisc64 automatically [13:26] <cone-621> ffmpeg.git 03Diego Biurrun 075bac2d0c3020: avutil: Move memcpy_backptr() to mem.c [13:26] <cone-621> ffmpeg.git 03Diego Biurrun 072a91ada8282f: avutil: Make LZO decoder code configure-time selectable [13:26] <cone-621> ffmpeg.git 03Michael Niedermayer 07aa604e8e33ae: Merge remote-tracking branch 'qatar/master' [13:46] <nevcairiel> ubitux: ignore the path replacement issue for now, its a purely fate thing, no-one sane should be calling a msvc ffmpeg.exe with a absolute msys-style path [13:46] <nevcairiel> i'll figure otu a way to adjust my fate setup with relative path or something [13:46] <nevcairiel> only remaining problem is the colon parsing [13:49] <ubitux> this one is solved [13:49] <ubitux> but ok, i'll wait for your patch [14:00] <cone-621> ffmpeg.git 03Michael Niedermayer 07da4e4d65f456: aacdec: reorder multiuplications to make code safer against too large input values. [14:00] <cone-621> ffmpeg.git 03Tomas Härdin 075c108092a34e: mxfenc: Write MultipleDescriptor ref in Preface [14:15] <durandal_1707> it is possible to dynamicaly (while reading packets) adding seek index entries? [14:16] <nevcairiel> it is, i believe some demuxers also do that [14:37] <nevcairiel> ubitux: tada! http://fate.ffmpeg.org/report.cgi?time=20121025123604&slot=x86_32-msvc10-wi… [14:37] <nevcairiel> just some path values shuffled around and suddenly it works [14:38] <ubitux> yay [14:38] <ubitux> magic [14:38] <divVerent> I hope this %3a thing is a bug in FATE ;) [14:38] <ubitux> yes [14:39] <ubitux> i have the issue with the shared box as well [14:39] <nevcairiel> its somewhat irritating that i had to set a target path [14:39] <nevcairiel> but if i dont, it uses a msys style path there [14:39] <nevcairiel> and we're back where we started =p [14:40] <nevcairiel> ok updating config for the 64-bit fate instance, and everything is good [14:41] <ubitux> did you have to patch anything in the tests? [14:41] <nevcairiel> no [14:41] <nevcairiel> it was just some terrible interaction between all the different scripts involved [14:42] <nevcairiel> next i should setup a shared msvc box, but the damn aac encoder still doesnt compile [14:43] <nevcairiel> i should use those new fancy fate macros to add a dependency so i can just turn it off [14:44] <ubitux> i don't understand how it can work [14:44] <ubitux> :D [14:45] <ubitux> but ok [14:45] <nevcairiel> what work? [14:45] <ubitux> "movie='d:\foo\bar'" becoming "movie=d:\foo\bar" [14:45] <ubitux> i wonder how you solved that [14:46] <nevcairiel> i have no idea [14:46] <nevcairiel> all i did was make sure its actually d:\foo\bar and not /d/foo/bar [14:47] <ubitux> :/ [14:47] <ubitux> you didn't kept my patch in, right? [14:48] <ubitux> that's insane how can it work.. [14:49] <nevcairiel> clean head, no patch [14:49] <nevcairiel> the only changes i did was to my script that runs fate and the fate config script [14:49] <nevcairiel> just changed the parameters for samples and target_path [14:50] <ubitux> ok& [14:50] <ubitux> then i don't get it [14:50] <ubitux> :D [14:50] <nevcairiel> maybe the sample is passed with a relative path now? [14:50] <nevcairiel> because my SAMPLES is relative [14:50] <ubitux> oh. [14:50] <nevcairiel> "../../samples" to be precise [14:51] <ubitux> that's the reason yes [14:51] <ubitux> right ok [14:51] <ubitux> you're just avoiding the issue then :p [14:51] <nevcairiel> yay me [14:51] <nevcairiel> :P [14:51] <ubitux> :)) [14:52] <nevcairiel> supporting absolute windows style path in single quotes should still work, i guess [14:52] <ubitux> it doesn't work [14:52] <ubitux> except in an external file [14:52] <nevcairiel> i know, but it should [14:52] <nevcairiel> ;) [14:53] <ubitux> it might work with double escaping [14:53] <ubitux> like saste pointed out the other day [14:54] <ubitux> nevcairiel: anyway, thanks for the workaround [14:54] <nevcairiel> couldnt let my fate box being broken all this time [14:57] <ubitux> is there any way to communicate from the encoder to the muxer? [14:58] <nevcairiel> side data? :D [14:58] <nevcairiel> its just a avpacket afterall [14:58] <ubitux> yeah but i don't have the packet in the encoder.. [14:58] <ubitux> well, subtitles& [14:58] <nevcairiel> oh subtitles [14:58] <ubitux> !#$% [14:59] <nevcairiel> a/v got avpackets in their encoders..... :p [15:00] <ubitux> i guess i'll have to create a new function encode function [15:10] <cone-621> ffmpeg.git 03Paul B Mahol 07296f9c2b3bc2: dsicinav: return meaningful error code [15:50] <cone-621> ffmpeg.git 03Paul B Mahol 07d8245c3bcdd1: dsicinav: return proper error code in case of malloc failure [15:50] <kriegerod> In my app the video muxing process is of non-monotonic intensity, but "bursty". I mux portions from one keyframe to next one at once. Output is often transmitted by UDP, and output stream has high peaks of bitrate, which leads to loss. I think about adding functionality of smoothing output bitrate for UDP URLProtocol. Could anybody help with algorithm? [15:54] <Tjoppen> VBV [15:54] <Tjoppen> though I don't know how to use it [15:59] <michaelni> setting maxrate and bufsize enabled VBV [16:12] <mateo`> is there a formula to compute the buffer size for mpeg2video ? [16:13] <kriegerod> VBV must be an algorithm? Could you please say what this abbrev stands for, because i cant google it [16:14] <michaelni> video buffer verifier [16:14] <michaelni> also you can leave the buffer size 0 with latest ffmpeg and it will try to pick something reasonable [16:15] <michaelni> but basically the maxrate specifies the maximum bandwith that should be used the the bufferf size the maximum latency/delay [16:15] <michaelni> s/the the/and the/ [16:27] <cone-621> ffmpeg.git 03Justin Ruggles 07d7de11260bd1: ac3dec: ensure get_buffer() gets a buffer for the correct number of channels [16:27] <cone-621> ffmpeg.git 03Michael Niedermayer 079a76b7375eaa: avsdec: Set dimensions instead of relying on the demuxer. [16:27] <cone-621> ffmpeg.git 03Anton Khirnov 079e575e54a057: dfa: check that the caller set width/height properly. [16:27] <cone-621> ffmpeg.git 03Paul B Mahol 0712941dbe2cc7: dfa: convert to bytestream2 API [16:27] <cone-621> ffmpeg.git 03Kostya Shishkov 072281ac9ffd28: dfa: add some checks to ensure that decoder won't write past frame end [16:27] <cone-621> ffmpeg.git 03Kostya Shishkov 07d0267ecf768b: dfa: use more meaningful return codes [16:27] <cone-621> ffmpeg.git 03Anton Khirnov 070c19855539d7: dfa: improve boundary checks in decode_dds1() [16:27] <cone-621> ffmpeg.git 03Kostya Shishkov 07965302c9f336: indeo: check custom Huffman tables for errors [16:27] <cone-621> ffmpeg.git 03Kostya Shishkov 07911c250aef9f: factor out common decoding code for Indeo 4 and Indeo 5 [16:27] <cone-621> ffmpeg.git 03Kostya Shishkov 07e0daa15a96cf: indeo: track tile macroblock size [16:27] <cone-621> ffmpeg.git 03Kostya Shishkov 07b561618014a2: indeo: clear allocated band buffers [16:27] <cone-621> ffmpeg.git 03Kostya Shishkov 07c5ec19085978: indeo: check for invalid motion vectors [16:27] <cone-621> ffmpeg.git 03Michael Niedermayer 07fe8243d7a9e3: Merge commit 'c5ec1908597824e93bbe20137ac9662f84f3cb07' into release/0.10 [16:38] <cone-621> ffmpeg.git 03Anton Khirnov 07332555f66049: ivi_common: make ff_ivi_process_empty_tile() static. [16:38] <cone-621> ffmpeg.git 03Anton Khirnov 070815d9174c48: indeo4/5: check empty tile size in decode_mb_info(). [16:38] <cone-621> ffmpeg.git 03Michael Niedermayer 07dc8371b2b12f: indeo5dec: Make sure we have had a valid gop header. [16:38] <cone-621> ffmpeg.git 03Janne Grunau 073efe6becc79b: indeo5: prevent null pointer dereference on broken files [16:38] <cone-621> ffmpeg.git 03Michael Niedermayer 075c413648c148: indeo5: check tile size in decode_mb_info(). [16:38] <cone-621> ffmpeg.git 03Anton Khirnov 071c8e2561b489: indeo3: fix out of cell write. [16:38] <cone-621> ffmpeg.git 03Michael Niedermayer 0714bba214fa9a: lagarith: check count before writing zeros. [16:38] <cone-621> ffmpeg.git 03Michael Niedermayer 076744eee1e5bf: wmaprodec: check num_vec_coeffs for validity [16:38] <cone-621> ffmpeg.git 03Anton Khirnov 070582b8e3eabb: avidec: use actually read size instead of requested size [16:38] <cone-621> ffmpeg.git 03Michael Niedermayer 072bc1e4fcb96c: indeo4: update AVCodecContext width/height on size change [16:38] <cone-621> ffmpeg.git 03Michael Niedermayer 072051adbfa008: cavsdec: check for changing w/h. [16:38] <cone-621> ffmpeg.git 03Anton Khirnov 0715c2e8027f48: wav: do not fail on empty INFO tags [16:38] <cone-621> ffmpeg.git 03Michael Niedermayer 0736487066ee04: Merge commit '15c2e8027f4827018608badb1bff1294af1810e4' into release/0.10 [16:42] <j-b> michaelni: if a user wants to use min/max, and no rc-buf-size, well, maybe VLC is not the right too [17:02] <ludde> When muxing a .ts file, are you allowed to split a single h264 frame into two separate PES packets? [17:03] <av500> yes [17:04] <av500> erm [17:04] <av500> not sure [17:04] <av500> :) [17:04] <ludde> ok :) [17:05] <michaelni> ludde, normally anything can be split in multiple PES packets, if you want to be 100% sure check the spec about h264 in ts [17:05] <av500> but I think you dont need to have frames strictle aligned to PES start [17:05] <ludde> michaelni: do you know the number of that spec? [17:06] <ludde> which frame does the PTS in the PES refer to, if the PES is not right before the frame? [17:06] <michaelni> H.222 is one mpeg systems something the other but their content should be the same [17:06] <JEEB> doesn't a recent enough H.222/MPEG-TS spec cover that? I think I saw quite a few references in a recent'ish version of the mpeg-ts spec [17:06] <JEEB> (towards H.264) [17:07] <michaelni> the PTS in PES refers to the first AU that comences in that PES [17:07] <nevcairiel> there is Annex B in 14496 Part 10, which covers how the h264 bitstream should be formed, dunno if it mentions such information as well [17:08] <JEEB> yes, there's that too but that's general info IIRC [17:08] <ludde> would this be valid .tse: [17:08] <ludde> sorry, .ts: [17:09] <ludde> Ts Packet 1: <PES HEADER FOR THE PPS/SPS STUFF> PPS/SPS STUFF> <INITIAL DATA OF FRAME 1> [17:09] <ludde> Ts Packet 2: <PES HEADER FOR FRAME 1> <MORE DATA OF FRAME 1> [17:10] <ludde> or worse: [17:10] <cone-621> ffmpeg.git 03Michael Niedermayer 07592ba6781558: alsdec: Check k used for rice decoder. [17:10] <cone-621> ffmpeg.git 03Michael Niedermayer 070f81057c1251: alsdec: Check that quantized parcor coeffs are within range. [17:10] <cone-621> ffmpeg.git 03Thilo Borgmann 07c5f9c272e9df: alsdec: Fix out of ltp_gain_values read. [17:10] <cone-621> ffmpeg.git 03Mans Rullgard 07c28e1c12adf4: alsdec: remove dead assignments [17:10] <cone-621> ffmpeg.git 03Thilo Borgmann 07dc5283dffcd4: alsdec: fix number of decoded samples in first sub-block in BGMC mode. [17:10] <cone-621> ffmpeg.git 03Michael Niedermayer 073c55bf1201ac: vc1dec: check that coded slice positions and interlacing match. [17:11] <cone-621> ffmpeg.git 03Sean McGovern 07a2d4d9f4fbe1: wmapro: prevent division by zero when sample rate is unspecified [17:11] <cone-621> ffmpeg.git 03Anton Khirnov 075acd1c6561c0: avidec: return 0, not packet size from read_packet(). [17:11] <cone-621> ffmpeg.git 03Anton Khirnov 07141d4ed6c091: cmdutils: avoid setting data pointers to invalid values in alloc_buffer() [17:11] <cone-621> ffmpeg.git 03Ronald S. Bultje 0779fb7bc667dd: h264: fix deadlocks on incomplete reference frame decoding. [17:11] <cone-621> ffmpeg.git 03Justin Ruggles 075920d00d7417: libvorbis: fix use of minrate/maxrate AVOptions [17:11] <cone-621> ffmpeg.git 03Justin Ruggles 0724025cc0b972: libvorbis: use VBR by default, with default quality of 3 [17:11] <cone-621> ffmpeg.git 03Anton Khirnov 07be209bdabb11: vf_pad: don't give up its own reference to the output buffer. [17:11] <cone-621> ffmpeg.git 03Michael Niedermayer 07d6a55ab016f0: Merge commit 'be209bdabb11c59de17220bdbf0bf9c9f7cc16f5' into release/0.10 [17:19] <ludde> can you assume that the whole PES header always fits in a single .ts packet? [17:20] <nevcairiel> isnt a pes header rather short? [17:20] <ludde> yes, but it may contain padding [17:27] <cone-621> ffmpeg.git 03Franz Brauße 07443f1463c0e1: smacker audio: sign-extend the initial 16-bit predicted value [17:27] <cone-621> ffmpeg.git 03Anton Khirnov 07d792be5681b4: yuv4mpeg: return proper error codes. [17:27] <cone-621> ffmpeg.git 03Anton Khirnov 070b923a2b72c1: vf_pad/scale: use double precision for aspect ratios. [17:27] <cone-621> ffmpeg.git 03JindYich Makovi ka 079822e3aa52d1: h264: avoid stuck buffer pointer in decode_nal_units [17:27] <cone-621> ffmpeg.git 03Luca Barbato 070f3381ad5bff: mpegaudiodec: fix short_start calculation [17:27] <cone-621> ffmpeg.git 03Alex Converse 078076d32f3092: tiffenc: Check av_malloc() results. [17:27] <cone-621> ffmpeg.git 03Reinhard Tartler 07ca8c814970a9: Prepare for 0.8.4 Release [17:27] <cone-621> ffmpeg.git 03Anton Khirnov 07a0f6c93f52f8: lavc: remove stats_out from the options table. [17:27] <cone-621> ffmpeg.git 03Reinhard Tartler 072c8ce46250ff: Update Changelog for the 0.8.4 Release [17:27] <cone-621> ffmpeg.git 03Mans Rullgard 076365b43295a0: svq3: replace unsafe pointer casting with intreadwrite macros [17:27] <cone-621> ffmpeg.git 03Michael Niedermayer 07988910a277f1: Merge remote-tracking branch 'qatar/release/0.8' into release/0.10 [17:27] <ubitux> mmh it's missing the branch! [17:46] <burek> Can any developer give an answer to this question: http://ffmpeg.gusari.org/viewtopic.php?f=13&t=705 [17:47] <burek> if you don't want to register on the forum, just answer here, I'll copy/paste it [17:48] <av500> he was here yesterday [17:48] <av500> [17:48:22] <frank_stl> Hi, We are developing one ffmpeg feature. So the update of ffmpeg is important to us. Just wondering does anybody know what is expected date of next ffmpeg version release? [17:48] <av500> [17:49:58] <saste> frank_stl, one release every 6 months more or less, which feature are you implementing? [17:49] <burek> oh I see, ok :) thanks [17:53] <michaelni> and the 6 month is still wrong [17:56] <av500> lets repeat it a few times then :) [19:40] <cone-621> ffmpeg.git 03Martin Ettl 07cc72d52dc1ed: ffserver: fix printf argument type [19:45] <durandal_1707> ubitux: you work on supporting caption tracks? [19:46] <cone-621> ffmpeg.git 03Nicolas George 0777a72d348519: lavfi/vf_fps: allow to set the rounding method. [19:51] <ubitux> durandal_1707: not right now [19:51] <ubitux> why? [19:52] <JEEB> if anyone wants an excersize in masochism you could try implementing ARIB STD-B 24 captions [19:54] <ubitux> i belive the priority would be teletext and the 6xx/7xx (i don't remember the numbers) [19:55] <JEEB> naturally, since USfags use those [19:55] <ubitux> but i have other projects in mind :p [19:55] <ubitux> i'd better design the AST sub rect type [19:56] <ubitux> and then insert that stuff in lavfi [19:56] <JEEB> there are a couple of half-baked ARIB caption implementations, but nothing has gotten to the mainstream things yet [20:00] <cone-621> ffmpeg.git 03Alexis Ballier 07ce028ab9a798: Restore installation of libavutil/lzo.h after 2a91ada8282f18d2807abee5188225bba1b19bda [20:03] <durandal_1707> seems i cant fined pts/dts when transcoding pcm in wav to wav or any other format [20:03] <durandal_1707> s/fined/fix [20:18] <cone-621> ffmpeg.git 03Nicolas George 0748ec8b25a7de: lavfi/af_volumedetect: print stats in uninit(). [21:06] <durandal_1707> what speech codec support 8k,12k and 16k sample rate? [21:12] <gnafu> durandal_1707: Opus does, doesn't it? Also, it looks like Speex does 8 and 16. [21:15] <durandal_1707> zvr files i'm exploring appears to use g729 like codec - which use 8k only AFAIK while file is in 16k [21:29] <cone-621> ffmpeg.git 03Janne Grunau 07cc88dacc1aa3: g722enc: fix size argument in memset [21:29] <cone-621> ffmpeg.git 03Janne Grunau 07c279e37e901e: flashsv: propagate inflateReset() errors [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 070b9d46434894: swscale-test: fix freeing of uninitialized variable [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 07ba10ea845f41: asrc_aevalsrc: Fix use of uninitialized pointer inside av_strtok() [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 077450a0215ae2: ffprobe: fix use of uninitialized pointer in av_strtok() [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 078a525e4d18c5: av_tempfile: fix leak in error case [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 07d12bf6fc9e7e: libvpxenc: fix memleak on error path [21:29] <cone-621> ffmpeg.git 03Paul B Mahol 07c9df500190ee: bmp: unbreak non BMP_RGB compression for v4 and v5 [21:29] <cone-621> ffmpeg.git 03Paul B Mahol 07e6dfaf7bb89a: truemotion2: remove unreachable code [21:29] <cone-621> ffmpeg.git 03Paul B Mahol 0746c1e5de58ac: yop: check return value of avformat_new_stream() [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 07f2d56c2eebe6: motion_est: more complete SAB diamond size check [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 078b64036038e2: aacsbr: change order of operation to prevent out of array read [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 073038e2041e87: ffserver: prevent nb_streams from becoming too large [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 077a0e5a63d01b: vf_fade: fix memleaks of args [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 07db4903f4e4b5: ffv1: avoid checking a double for equality [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 07c09b4dde3776: wtvdec: fix memleak on error [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 07fa73f547a023: jpegls: fix off limit [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 0735b15a0da849: jpegls: increase run_index to 4 [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 0712801f969bf5: trasher: check seek return value. [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 0793a0dd8358ee: ffeval: avoid folding EOF onto a valid char [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 07de4606a5b798: pp: avoid overflow in w*h [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 07cff9f07d391f: ffv1: make sure gob_count is not 0 [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 07400b23beab0a: dnxhddata_ Fix mixup of sizeof() and array elements in ff_dnxhd_find_cid() [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 07e6fa08f14efd: flashsv: check deflateInit() return value [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 0775a11e950fc4: mpegvideo: fix motion_val checks [21:29] <cone-621> ffmpeg.git 03Thilo Borgmann 077f1fb8d2a367: alsdec: fix clipping of weightings for MCC decoding [21:29] <cone-621> ffmpeg.git 03Michael Niedermayer 075b5e61a0bf0d: noise_bsf: fix division by 0 [21:30] Action: michaelni notes that cone does not list the branch commits get pushed to [21:36] <funman> michaelni: you should ask thresh [21:39] <cone-621> ffmpeg.git 03Stefano Sabatini 07b19bfd6c9f42: lavd/lavfi: fix leak in case of failure [21:40] <durandal_1707> what is going on? heads should get ripped of and necks ... [22:10] <ubitux> saste: hey :) [22:10] <ubitux> saste: in the showspectrum patch comment, you were talking about negatic logic [22:11] <ubitux> what were you talking about? [22:12] <saste> if (!expr) { ... } else ... rather than if (expr) { ... } else [22:12] <ubitux> oh, ok. [22:13] <ubitux> saste: do you mind if i keep the local variable to "sliding" and just rename the option to "slide"? [22:13] <michaelni> saste, did you get your coverity account ? [22:14] <saste> ubitux, i prefer a consistent "slide", but do as you want provided you keep an external consistent interface [22:14] <saste> michaelni, no [22:15] <michaelni> did you check both mail addresses ? [22:15] <michaelni> the accounts definitly have been created [22:19] <cone-621> ffmpeg.git 03Clément BSsch 0713d26716fb58: lavfi/showspectrum: add sliding mode. [22:22] <saste> michaelni, nothing, in both accounts [22:25] <michaelni> saste, try the "forgot password button" on http://scan5.coverity.com:8080 [22:25] <michaelni> username should be saste for gmail and saste2 for poste.it [22:26] <michaelni> if that fails too ill mail them [22:26] <michaelni> also keep in mind the coverity mails tend to get marked as spam [22:32] <cone-621> ffmpeg.git 03Michael Niedermayer 0736982b361672: Update for 0.10.6 [22:33] <saste> michaelni, bingo [22:37] <saste> michaelni, 39747d87d09079371b75092ba979ca460536f646 is broken in many ways [22:38] <saste> checking for a literal string in a filtergraph is just wrong [22:38] <saste> also, why was it added? [22:39] <saste> also, a scale algo is not a flag, we should really fix that [22:39] <michaelni> because fate fails on all non x86 platforms without it (see 2d11ee4bfc7cf8c04e40782fc73d72a3d4f0e246) [22:40] <michaelni> you should be able to reproduce the failure with --disable-asm [22:40] <ohsix> hashes are awesome [22:41] <saste> for the lavfi device i suggest to implement a lavfi option, -scale_sws_flags [22:41] <michaelni> ohsix, b6e158edf34b64b2baf154f7a9e6c46a [22:41] <ohsix> i meant the other ones :p you can google them and they pretty much always reference the same thing, too [22:42] <ohsix> as a general principle, that's pretty handy [22:42] <saste> also adding flags to the scale args in the application/filtergraph code is brittle [22:42] <saste> we need a better mechanism [22:43] <michaelni> if we add it to the lavfi device its still missing when something else uses it [22:43] <saste> we could even access the filtergraph field from the scale filter [22:47] <saste> i'd keep generic filtergraph options separated from the filtergraph description [22:47] <saste> if not, embedding the options in the filtergraph text and doing a strstr is not correct, anyway [22:48] <michaelni> you could add a AVDictionary to the graph and a filter that just puts its args there so a setglobal=swsflags;abcd would do the trick [22:49] <saste> michaelni, AVDictionary? for what purposes? [22:49] <saste> what should it store [22:50] <michaelni> flags for autoinserted filters [22:50] <michaelni> for example [22:50] <michaelni> scale is one [22:50] <michaelni> resample another [22:50] <michaelni> there might be more in the future [22:50] <saste> uhm something like scale=sws_flags=... [22:51] <saste> scale is the key, the auto-inserted options may be the values [22:51] <saste> migth work [22:51] <saste> also we may need to specify the instance name of a filter [22:52] <michaelni> i dont see why [22:52] <saste> so that the user can properly send command to a given specific instance [22:52] <saste> suppose you have many instances of the same filter, and you want to send a command at a specific one [22:52] <michaelni> its about global falgs [22:52] <michaelni> flags [22:53] <saste> relying on the automatically assigned name (parsed_scale_ etc) is not robust [22:53] <michaelni> if you want flags for a specific instance one could just put a scale=... there [22:53] <saste> no i mean with *commands* [22:53] <michaelni> also "scale=sws_flags=..." needs thought, that syntax is not good [22:53] <michaelni> double = = i mean [22:53] <saste> i agree [22:54] <saste> also if you want to specify more options, that would require a third level of escaping [22:55] <saste> e.g scale=iw*2:ih*2:sws_opts='sws_flags=+bitexact:sws_algo=lanczos' [22:56] <saste> that's getting easily clunky [22:57] <saste> ubitux: i want to rename asetnsamples to asetframesize [22:57] <saste> never liked that name [22:58] <cone-621> ffmpeg.git 03Paul B Mahol 0743f662d9bf69: lvfdec: cosmetics: fix identation [22:58] <michaelni> sws_opts='sws_flags=+bitexact:sws_algo=lanczos' <--- this is sick [22:59] <michaelni> sws_flags=+bitexact+lanczos [22:59] <saste> algo != flag [22:59] <saste> algo should be set like a constant, it is semantically not a flag [23:00] <michaelni> then its [23:00] <michaelni> sws_flags=+bitexact:sws_algo=lanczos [23:00] <michaelni> but not [23:00] <michaelni> sws_opts='sws_flags=+bitexact:sws_algo=lanczos' [23:01] <saste> how are we doing in aresample? [23:01] <saste> scale takes additional options which are not forwarded to the sws context [23:01] <ubitux> saste: i don't like asetframesize that much, but whatever [23:02] <saste> so you need to embed sws options into a separate "value", but maybe i'm wrong [23:02] <saste> that in the case you need more options to set, right now we only set the flags [23:03] <saste> ubitux: why not? [23:04] <michaelni> theres no need to seperate the 2levels, either the option is handled by vf_scale.c or sws [23:05] <saste> michaelni, i suppose that's possible, if there is no conflict possibility [23:05] <michaelni> well we are controling both vf_scale and sws so i guess we should manage not to make the 5 options in vf_scale conflict with the 5 in sws [23:07] <saste> michaelni, a general namespacing system may be useful, especially for the movie where we may want to specify codec/format/proto options in a transparent way [23:07] <michaelni> the whole escaping thing is mad [23:08] <saste> michaelni, ? [23:08] <michaelni> the only escaping that should be needed should be for the shell when parameters like strings or filenames need excaping [23:10] <michaelni> and for the end character(s) in strings or filenames [23:10] <saste> michaelni, there are special chars in the filtergraph (;,) and in the filter description (: for separating options) [23:10] <saste> right now we have two levels of escaping, that sucks but I can't see how you can improve things [23:11] <saste> note that alternative syntaxes (e.g. avisynth) don't have this problem [23:12] <ubitux> saste: because i find frame size very vague [23:12] <ubitux> size is not a number of samples, it's more like a byte size [23:12] <saste> ubitux, the thing is that i plan to add a size option [23:13] <saste> so you can say: asetframesize=z=10KiB [23:13] <ubitux> aclip? :) [23:13] <saste> it still make sense to say: asetframesize=n=1000 [23:13] <saste> but saying: asetnsamples=z=10KiB doesn't [23:16] <michaelni> saste, if avisynth doesnt have this problem then quite obviosuly it can be improved [23:16] <saste> michaelni, both syntaxes have their strong points ;-) [23:17] <saste> the ffgraph syntax was designed to express *simple* graphs on the command line [23:17] <saste> avisynth syntax was designed to be read from a file, where no shell escaping issues occurr, and where you don't need to keep it short [23:18] <saste> we could add another alternative syntax (ffsynth?) [23:19] <ohsix> ffgst [23:20] <michaelni> its hard to argue about this without seeing examples of ffgraph (with whitespace +\n), avisynth and soem proposal of a new syntax or chnages to ffgraph [23:20] <ohsix> why is the shell escape thing even a consideration when you're trying to make something comprehensible [23:52] <cone-621> ffmpeg.git 03Paul B Mahol 0704a585f054ea: fraps: use meaningful error codes [23:56] <cone-621> ffmpeg.git 03Stefano Sabatini 0715f52e50fe37: tools: add ffescape utility [00:00] --- Fri Oct 26 2012

1 0

ffmpeg.log.20121025
by burek 26 Oct '12

26 Oct '12

[06:16] <mancha> hi folks. i am getting an error when i try to convert to -acodec libmp3lame -ab 96000 -ar 22500 [06:17] <mancha> Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height [06:18] <mancha> is 22.5k not allowed? [07:20] <V1nce> hello [07:21] <V1nce> question: how can i start ffplay fullscreen? i tried -fs but no go [07:21] <V1nce> in fact it was like going into fullscreen for a half second but it got back windowed [07:22] <V1nce> anyone alive? :-O [07:32] <zap0> ffplay --help? [07:33] <V1nce> don't see any full screen on start zap0 [07:33] <V1nce> i want it to play in fullscreen immediately [07:34] <V1nce> i checked docs as well, ofcourse [07:34] <V1nce> but no traces of fullscreen on start, only "f" key while playing [07:36] <zap0> ffplay -fs file.ext [07:36] <V1nce> -fs doesn't work [07:36] <V1nce> tried on 2 boxes [07:38] <V1nce> oh wait, it works, just tried with ordinary pal video [07:38] <mark4o> ffplay -fs works for me [07:38] <V1nce> but on 720p it doesn't work [07:38] <V1nce> with 720p video [07:38] <V1nce> good god [07:39] <V1nce> yes yes it works, just tried with another video [07:39] <V1nce> 720p mp4 doesn't though [07:39] <V1nce> it enters fullscreen for half second and gets to windowed mode [07:41] <mark4o> what os and screen resolution [07:42] <V1nce> win 7 1920 x 1080 [07:43] <V1nce> okay i just found out that ffplay actually switches the monitor resolution to the size of the video [07:43] <V1nce> when played in fullscreen [07:44] <V1nce> no scaling involved [07:45] <V1nce> so something, somehow, is in conflict with this 1280x720 piece [07:45] <V1nce> and it enters windowed [07:45] <V1nce> no probs with lower resolution though [07:45] <V1nce> but shouldn't be the problem with this one either. hmm [07:48] <V1nce> Frame changed from size:0x0 to size:1280x720 [07:49] <V1nce> all seems good, but it starts windowed :-) [07:49] <mark4o> 720p mp4 works in fullscreen for me, although I'm on a mac not win 7. perhaps an issue with the resolution change [07:49] <V1nce> yes [07:49] <V1nce> if i can just tell it to scale it to the native res [07:49] <V1nce> which will be perfect [07:52] <V1nce> weird thing i press "f" and it is all good [07:52] <V1nce> fullscreen plays like there's no tomorrow [07:52] <V1nce> change res, scales, all great [08:07] <V1nce> solved [08:07] <V1nce> very elegantly [08:07] <mark4o> cool [08:08] <mark4o> how did you fix it? [08:08] <V1nce> you know how? i command line VLC, works like charm :-) [08:08] <V1nce> also i could hide mouse pointer [08:09] <mark4o> ok no ffplay then [08:09] <V1nce> no :-/ [08:10] <V1nce> it is desktop thing, fscommand it from flash projector, so all good [08:11] <V1nce> wanted to open video externally, so it plays and when finished gently exits to the menu [08:11] <V1nce> vlc is very discreet with this [08:12] <mark4o> well glad you got something working [08:12] <V1nce> me too [08:12] <V1nce> ffmpeg is the beast with server related things [08:13] <V1nce> batch it, tell it anything and it will do it [08:14] <mark4o> it doesn't rebel too often [08:17] <V1nce> keeps his mouth shut and does the job [08:17] <V1nce> mark4o, are you related to ffmpeg development? [08:17] <mark4o> no [08:19] <V1nce> okay, more time for bourbon, chicks and horses [08:20] <mark4o> :) just a user and fan [08:29] <V1nce> gtg, ciao mark4o [09:54] <zro> on Debian, do i need to install a seperate package for ffmpeg presets? i dont see them in /usr/share... ? I thought they were there as ffmpeg/*.ffpreset files or something [10:55] <brontosaurusrex> zro: what presets do you need? [12:12] <Yulth> join #zfs [12:18] <spaam> Yulth: no thanks [12:19] <jameshowe> there's no details in the current -map documentation about how to map from multiple input files to a single output file [12:20] <jameshowe> e.g. taking a video stream from the first and an audio stream from the second [12:22] <jameshowe> oh wait, maybe it does, except that "the second -map option specifies the source for output stream 1" is highly misleading [12:29] <skifreak> hey guys, is there a way to copy truehd from one video to another? Just trying to do a -c:a copy from one video file to another. I know demuxing truehd is supported, but remuxing isn't. Trying to figure out a way to get around this problem, thanks! [12:34] <skifreak> should I convert the truehd track instead to another format, such as flac, which would carry all of the channels and quality of the original audio encode? [12:35] <divVerent> it depends on what you are trying to do [12:35] <jameshowe> skifreak: that happens with a lot of things, usually you have to find a bsf to stick in [12:35] <divVerent> what is your output format? [12:35] <jameshowe> but then that isn't guarenteed to work [12:36] <jameshowe> the mp4 muxer can't read it's own aac output for example [12:36] <skifreak> my output format is mkv, but I can do m2ts [12:37] <skifreak> here's the info on the track I'm trying to encode from http://dpaste.com/818100/ [12:38] <divVerent> what is the error you get for trying to mux it into mkv? [12:39] <skifreak> [NULL @ 0x16b1d00] Unable to find a suitable output format for 'ashowinfo:fmt' [12:39] <JEEB> <jameshowe> the mp4 muxer can't read it's own aac output for example <- huh? for me all of my AAC-in-"MP4" muxes have gone just nice, and then read with libavformat [12:39] <skifreak> not a bad idea to use -bsf, just trying to preserve as much noticeable audio quality as I can. My objective is always tranparent encodes. [12:39] <divVerent> skifreak: what if you don't use ashowinfo? [12:40] <divVerent> well, one lossless to another lossless is also guaranteed transparent [12:40] <divVerent> skifreak: I mean, you're using an audio filter, this basically isn't really consistent wiht using -codec:a copy [12:40] <jameshowe> JEEB then if you try to make another mp4 with it using -c copy, you get the "malformated aac bitstream" message [12:40] <JEEB> huh... [12:41] <JEEB> lemme try [12:41] <skifreak> divVerent, I didn't think I was using an audio filter if I was just copying the audio stream [12:41] <jameshowe> or maybe it's make a mpegts first (with libfaac encoder) then streamcopy to an mp4 [12:41] <divVerent> well, you got an error from ashowinfo, which is an audio filter [12:42] <jameshowe> anyway, googling that error message shows everyone else that gets that problem [12:42] <divVerent> or is that just a VERY misleading error and your command line doesn't mention ashowinfo? [12:42] <JEEB> jameshowe, lavf mp4 -> mp4 aac WorksForMe, so it's something else [12:43] <skifreak> sorry, wrong line of code, [matroska @ 0x2116d00] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 2 >= 2 [12:43] <divVerent> that tells more :P [12:44] <skifreak> yea....bit late on this side of the sphere [12:44] <divVerent> basically, whatever is your input is not suitable for mkv, not sure if this is caused by mkv rounding to milliseconds [12:44] <divVerent> if your target format is mkv, I'd however transcode to flac anyway [12:44] <jameshowe> JEEB, just checking my logs from yesterday, looks like (mpegts -> mp4 libfaac) and (mp4 -> mpegts libx264) are the streamcopies that needed bsfs [12:44] <divVerent> when going between different multichannel formats, you may have to test the output [12:45] <divVerent> as channel layouts may differ [12:45] <JEEB> if the input is mkv or mpeg-ts he can try using mkvtoolnix [12:45] <JEEB> mkvmerge from there [12:45] <divVerent> and there may be bugs causing bad conversions [12:45] <jameshowe> at least for the data I was playing with [12:45] <skifreak> divVerent that's why I've been going with just copying the audio stream [12:45] <skifreak> I could go with a different container, if you think one might be more suitable [12:45] <divVerent> but other losses shouldn't happen when toggling between lossless formats [12:46] <divVerent> you can try [12:46] <skifreak> what if my input file was a m2ts instead, might I get different results? [12:46] <divVerent> maybe [12:46] <divVerent> does it go wrong with multiple such videos? [12:47] <skifreak> nope, this is the first [12:47] <divVerent> as it MIGHT also be an actually "defect" encode [12:47] <divVerent> as this is a defect decoders would usually cope with, except for maybe a small click [12:47] <divVerent> nobody would notice in the silence at the start anyway [12:47] <divVerent> but muxers really HATE non monotonous dts [12:48] <skifreak> ok, well, I have version 0.10.3 [12:48] <skifreak> why is that? [12:48] <divVerent> as the dts is what they sort packets by [12:49] <JEEB> jameshowe, the latter being a H.264 stream copy? yeah, mp4->mpeg-ts needs the h264_mp4toannexb bitstream filter, because mp4 contains the H.264 stream in another format, and mpeg-ts want it in another [12:49] <JEEB> it wouldn't really make sense with libx264 re-encoding unless ffmpeg is being really dumb [12:49] <divVerent> skifreak: BTW, does ffmpeg continue encoding apart from the error message? [12:50] <JEEB> because libx264 can output either length prefixed /or/ start code prefixed [12:50] <jameshowe> yep, and the other one needs aac_adtstoasc [12:50] <skifreak> nope, it stops, dea in its tracks, divVerent [12:50] <skifreak> *dead [12:50] <divVerent> skifreak: so, no error, it doesn't abort, but also not encode [12:50] <JEEB> ok, that doesn't mean anything is really broken, just that ffmpeg isn't for some reason "smart" enough to insert those bit stream filters in there [12:51] <jameshowe> although I'm yet to determine whether the resulting file actually works properly on my target devices (whereas re-encoding with libfaac does) [12:51] <skifreak> divVerent: when I try to encode, it gave me the above error, and wouldn't encode one frame. I was back to stdout [12:51] <divVerent> can you quote all messages you got? [12:51] <JEEB> not sure about the adts to asc filter, but in general it's the same thing, two ways of a bit stream and for some reason a certain muxer wants it in a certain way [12:51] <divVerent> just asking, as I also once wrote code using ffmpeg, and did get this error due to a bug [12:51] <JEEB> jameshowe, if you're encoding for hardware boxes or whatever I recommend you use a more tested mpeg-ts muxer [12:51] <divVerent> but it continued encoding, with frames missing due to the error [12:51] <JEEB> either kierank's libmpegts or videolan's thingy [12:52] <JEEB> I'm still surprised that ffmpeg hasn't jumped on either [12:52] <jameshowe> JEEB, no LCD android and iOS devices [12:52] <jameshowe> which is "fun" [12:52] <JEEB> well those things in general get fed quite the broken shit [12:52] <skifreak> divVerent: this is my exact output http://dpaste.com/818104/ [12:52] <JEEB> so I guess libavformat's mpeg-ts muxing should be "OK" [12:52] <jameshowe> the mpegts muxing is good [12:52] <jameshowe> the mp4 muxing is good [12:53] <JEEB> hahaha, now you said a good joke [12:53] <JEEB> it's parse'able as mpeg-ts, but it really, really doesn't AFAIK follow the specs [12:53] <kelnoky> I have several video files here and some of them don't play on my android device (but ofc fine on my pc). the only difference I found between them (via mediainfo) was that the working files have 90k maximum fps while the files that don't work have a lower maximum frame rate than they have minimum frame rate [12:53] <divVerent> skifreak: ah, I see, this error is intentionally fatal in ffmpeg.c [12:53] <JEEB> the only ones that I know that are freely available and follow the specs are kierank's libmpegts and videolan's whatever it was [12:53] <skifreak> divVerent why is it intentionally fatal? [12:54] <jameshowe> it's the aac that goes a bit heywire sometimes [12:54] <divVerent> skifreak: because whenever this happens, the output file IS broken [12:54] <JEEB> you should say "the mpegts muxing is 'good enough for my use case'" [12:54] <divVerent> just, it may be not broken enough to not play any more ;) [12:54] <JEEB> because saying it's "good" will lead to incorrect assumptions :) [12:54] <skifreak> wow, ok [12:54] <divVerent> skifreak: if you are fine with changing code, you could try changing the code there [12:54] <divVerent> by removing that exit_program(1) line in ffmpeg.c [12:54] <jameshowe> it's all relative when you're working with android devices, [12:55] <divVerent> and then trying if you get only one, or lots of these errors [12:55] <jameshowe> it's astonishing how many ways you can implement graphics drivers wrong [12:55] <divVerent> if you get lots, then it sure won't work. If you only get one or two lost audio frames, chances are your encode MAY play back fine anyway, because of the first frames typically being silent anyway [12:55] <divVerent> don't know how TrueHD copes with that though [12:55] <skifreak> well, I can play back the video [12:55] <skifreak> mplayer handles it fine [12:56] <divVerent> skifreak: it did write video? [12:56] <divVerent> but the dts is 2, that's very low... [12:56] <divVerent> so your output file should have neither much audio nor video [12:56] <divVerent> when it aborts at this point [12:56] <JEEB> jameshowe, well the idea is that libavformat's mpegts muxer afaik doesn't create a mux that is fully compliant, while there are now at least two contenders for doing that [12:56] <skifreak> yep, you got it [12:57] <skifreak> It wrote the file, but it's extremely small [12:57] <divVerent> ok [12:57] <divVerent> then try that code change [12:57] <divVerent> if you think you can [12:57] <divVerent> if it leads to a successful encode, ffmpeg MAY even get an option for this in the future (strict -3 maybe? :P) [12:57] <skifreak> I should be able to with this Gentoo box, it's been a really long time since I did it last though [12:58] <jameshowe> right, so what I'm actually here for is mapping the video stream from a filter chain and the audio from a second input into the same file [12:58] <jameshowe> "-map [out] -map 1:a" gives only audio [12:59] <jameshowe> "-map 0:v -map 1:a" gives both streams, but the video is just solid grey (it's not supposed to be) [12:59] <jameshowe> this was previously achieved by "-newaudio", but that option has been removed [13:01] <JEEB> I think the map command is nowadays /really/ powerful but hell if I can remember the syntax :P [13:01] <JEEB> and yeah, just tested on a mp4 mux I've done and I need the mp4-to-annexB filter for a mpeg-ts remux [13:01] <JEEB> for H.264 [13:04] <JEEB> it's not broken or incorrect, but it's funny that you have to manually set it, since I would think it's relatively automatic that if you have a "MP4" file that you have the H.264 stream not in annexB format [13:04] <skifreak> divVerent: all of the exit_program(1) lines? [13:04] <skifreak> in ffmpeg.c? [13:11] <divVerent> skifreak: only the one near a call to av_interleaved_write_frame [13:11] <divVerent> you also can remove all and see what happens ;) [13:12] <skifreak> got it, figured it out, compiling [13:15] <jameshowe> and not using any -map gives the correct video, but no audio stream [13:19] <skifreak> http://dpaste.com/818114/ [13:20] <skifreak> divVerent: received a series of errors, as seen above [13:22] <divVerent> skifreak: okay... quite likely it won't work with matroska then [13:23] <divVerent> what if you now try another output format? [13:23] <skifreak> so I should use a different containter? [13:23] <divVerent> yes [13:23] <divVerent> try .nut [13:23] <skifreak> k [13:23] <JEEB> jameshowe, does it show which video track gets selected without the -map ? [13:23] <skifreak> same [13:23] <divVerent> with similar values too? [13:23] <divVerent> increasing in increments of 5? [13:24] <skifreak> nope, increments of 5 [13:24] <jameshowe> so without any map I get Stream #0:0 -> #0:0 (png -> libx264) [13:24] <divVerent> now wondering if there is a way to print ALL pts values... [13:24] <skifreak> 2788 >= 2788 is on one line, the next is 2793 >= 2793 [13:24] <divVerent> *dts [13:24] <skifreak> good point [13:24] <jameshowe> but that's the correct filter output (the filters use some movie=) [13:24] <divVerent> skifreak: try again with -debug_ts [13:25] <divVerent> and again quote the first pages of output errors ;) [13:25] <JEEB> jameshowe, so -map 0:0:0 would be correct for that, m'kay. What about audio? [13:25] <JEEB> (at least I just had -map 0:0:0 work to get the track 0 of file 0) [13:27] <skifreak> divVerent: here http://dpaste.com/818117/ thanks for the help btw [13:27] <JEEB> (and set it to be track 0 of output) [13:27] <divVerent> skifreak: all you changed was -debug_ts? [13:27] <skifreak> yep [13:27] <skifreak> and .nut output [13:27] <divVerent> weird, so debug_ts doesn't work with stream copy at all? [13:27] <jameshowe> JEEB, when I did that it did not do the filterchain [13:28] <jameshowe> hang on, now I'm getting "Output with label 'out' does not exist in any defined filter graph" [13:28] <jameshowe> arrgh [13:28] <divVerent> skifreak: what if you use matroska again? [13:28] <divVerent> you should get the previous errors back :) [13:28] <divVerent> the error here is about the video stream BTW [13:29] <skifreak> hunh, I was able to disable the audio stream before and the video started to encode just fine [13:29] <divVerent> [NULL @ 0x1ae0b20] [Eval @ 0x7fffcf4dcb20] Undefined constant or missing '(' in 'ts' [13:29] <divVerent> [NULL @ 0x1ae0b20] Unable to parse option value "ts" [13:29] <divVerent> [NULL @ 0x1ae0b20] Error setting option debug to value ts. [13:29] <divVerent> ah, there's the problem [13:29] <divVerent> it's -debug_ts, not -debug ts [13:30] <skifreak> ah, sorry, will try again with .nut [13:31] <skifreak> actually, i double checked, I didn't use _ts [13:31] <skifreak> this is what I got for a .mkv output file http://dpaste.com/818120/ [13:32] <divVerent> you still have -debug ts [13:32] <divVerent> not -debug_ts [13:32] <jameshowe> but only when not attempting to add the audio [13:32] <skifreak> well, it doesn't recognize debug_ts [13:32] <JEEB> I just tried with two files [13:33] <divVerent> skifreak: odd [13:33] <skifreak> Unrecognized option 'debug_ts' [13:33] <JEEB> -i derp1.mp4 - derp2.mp4 -map 0:0:0 -map 1:0:0 out.mp4 [13:33] <divVerent> my version has -debug_ts [13:33] <divVerent> try -fdebug ts? [13:33] <JEEB> and that seemingly works [13:33] <JEEB> argh [13:33] <JEEB> 1:0:1 [13:33] <JEEB> I mean [13:33] <jameshowe> JEEB, yes that works, now try with a filter chain [13:33] <skifreak> nope, same error [13:34] <jameshowe> JEEB, you want my full command? [13:34] <JEEB> somehow I feel like I don't, but pastebin it if you want :P [13:34] <divVerent> skifreak: weird [13:34] <skifreak> divVeren: here is what both of them look like http://dpaste.com/818126/ [13:34] <divVerent> for me, this works: [13:34] <divVerent> ffmpeg -debug_ts -i file.mkv file.nut [13:35] <JEEB> divVerent, his ffmpeg is version 0.10.3, wasn't -debug_ts newer than that? [13:35] <divVerent> OH... that is it then [13:35] <divVerent> does 0.10.3 have a similar option? [13:35] <JEEB> I would guess not [13:35] <skifreak> ok, well, which version should I get? [13:36] <divVerent> either 1.0 or git [13:36] <skifreak> ok, I can go with the hardmasked 1.0 [13:36] <divVerent> this MAY actually also fix your issue... stupid me, haven't tried that before [13:36] <divVerent> beware, it has different library names [13:37] <divVerent> if you have programs depending on ffmpeg, these will need recompiling [13:37] <skifreak> deps? [13:37] <divVerent> it may be easier to rather just compile ffmpeg in your home dir for a test [13:37] <divVerent> and not install it, just use the ffmpeg binary you get after compiling [13:37] <JEEB> in which case just doing a git clone is simpler [13:37] <divVerent> sure [13:37] <divVerent> you will need some --enable commands, like for libx264 [13:37] <skifreak> haha, it's no problem, I compile everythign with gentoo as it is, good workout for my 3770k [13:38] <skifreak> so my ffmpeg command will change? [13:38] <JEEB> can you thank gentoo for disabling asm optimizations on x264 for 32bit [13:38] <jameshowe> http://pastebin.com/nYzs9Vnp [13:38] <JEEB> and thus bringing a nice amount of stupid people saying x264 is slow onto #x264 [13:38] <divVerent> jeeb: that's only to force people to FINALLY use 64bit distros on 64bit systems ;) [13:38] <JEEB> heh [13:38] <divVerent> I so hate people using a 32bit OS on 64bit systems [13:38] <divVerent> it's annoying [13:38] <divVerent> especially if they have over 2 GB RAM ;) [13:39] <skifreak> yea....have to agree with you there divVeren [13:39] <jameshowe> JEEB, and other, more complex filters may also be involved with a few more movie= [13:39] <skifreak> ok, going with 1.0 then [13:39] <divVerent> just, this likely isn't why Gentoo does that [13:39] <JEEB> ^ [13:39] <JEEB> I also found that macports did something similar [13:39] <JEEB> thankfully nobody uses macports [13:39] <JEEB> so it didn't create an influx of idiots [13:39] <divVerent> Gentoo probably rather does it because the asm optimizations for 32bit don't work with a certain bleeding edge version of some compiler gentoo users love to install ;) [13:40] <divVerent> like, maybe it was buggy for a week in git checkouts of clang ;) [13:40] <JEEB> heh, looking at the randomness of compiler versions and browsers that are tagged 'stable', I really don't know about that either [13:41] <divVerent> but still, x264 IS slow ;) [13:41] <JEEB> lol [13:41] <divVerent> unless you tell it to be fast [13:41] <JEEB> the default is --preset medium [13:41] <divVerent> but I want it to be slow and yield good quality, don't care if it takes longer [13:41] <JEEB> which is far from slow [13:41] <divVerent> (still encodes way faster than realtime) [13:41] <divVerent> I still call it slow [13:41] <JEEB> sure, it can be slow depending on your machine, but then again placebo is much slower [13:41] <divVerent> "slower than transferring the file to the iPhone is" ;) [13:42] <divVerent> but anyway [13:42] <divVerent> my point here is, your software can be as fast as you want [13:42] <divVerent> there's always people who'd like it even faster [13:42] <divVerent> on a related note: "next iOS version shortened my battery lifetime" [13:42] Action: skifreak not surprised [13:43] <divVerent> basically, I call x264 "slow", but the slow performance is worth the results [13:43] <divVerent> there is faster codecs, but they suck, and if you want worse quality, you can typically outperform these by using a faster x264 preset anyway ;) [13:44] <skifreak> gotta love -preset [13:44] <divVerent> also, I wouldn't complain if x264 from today to tomorrow got 50% slower. I'd either deal with it, or pick a faster preset. [13:44] <JEEB> -preset and -crf are the best things that have happened to using video encoders lately :P [13:45] <divVerent> -crf, mainly [13:45] <JEEB> yeah [13:45] <divVerent> preset, not so much [13:45] <divVerent> or rather, LAME already does this too :P [13:45] <divVerent> but is an audio encoder [13:45] <JEEB> well, preset is nice for easymode speed/compression selection [13:45] <divVerent> LAME's -q option is quite similar to x264 -preset [13:45] <skifreak> speaking of which should I change the ffmpeg command I've been using in the dpastes? I switch could to something along the lines of -crf 18 -preset for tranparent 1080p? [13:46] <JEEB> crf is nice because you don't have to start guessing random bit rate values for clips [13:46] <divVerent> just that LAME named the option so wrong... [13:46] <divVerent> I mean, how many people have once thought that -b is for CBR mode and -q for VBR mode? ;) [13:46] <divVerent> while -V is VBR, and -q is just a general algorithmic quality setting (similar to x264 -preset) [13:47] <divVerent> skifreak: as for the command I don't know [13:47] <divVerent> you sure shzouldn't need that many options [13:47] <divVerent> I'd just go with -crf 18 for a start [13:47] <skifreak> ok [13:47] <skifreak> deblockalpha wasn't recognized by 1.0 [13:47] <JEEB> argh [13:47] <divVerent> it's probably called different [13:47] <JEEB> manual libx264 settings make me cry [13:48] <divVerent> -deblock <string> E..V.. Loop filter parameters, in <alpha:beta> form. [13:48] <divVerent> but you really shouldn't need to manually tune these [13:48] <skifreak> ....and we're up and running! [13:48] <divVerent> you can use all x264 -preset -profile -tune [13:48] <skifreak> 1.0 worked [13:49] <divVerent> skifreak: even the TrueHD stuff? [13:49] <skifreak> well, it's encoding [13:49] <divVerent> nice [13:49] <skifreak> 8.8 fps with -crf and -slower 18, copying the audio [13:49] <divVerent> so all we found out while debugging was that your ffmpeg version was old and broken ;) [13:50] <skifreak> erm, -crf 18 and -preset slower [13:50] <divVerent> however, BTW [13:50] <divVerent> there is no "really" allowed way to put TrueHD in mkv [13:50] <divVerent> it will go through that hack to use quicktime or avi fourccs [13:50] <divVerent> wait, sorry [13:50] <divVerent> no [13:50] <skifreak> the quality and channels will still be preserved though...right? [13:50] <divVerent> A_TRUEHD exists in matroska, so you can consider it officially supported [13:50] <divVerent> if you copy, sure [13:51] <divVerent> I was just saying, ffmpeg allows you to put "anything" into matroska, no matter how insane [13:51] <divVerent> but it then uses weird "compatibility" hacks to encode quicktime or avi codec tags [13:51] <divVerent> which I wouldn't assume to be a great idea to use [13:52] <divVerent> but anything not "too odd" should be fine [13:52] <divVerent> and I thought TrueHD was "too odd", but it's not ;) [13:52] <skifreak> well, does bluray have any formats which are "too odd"? [13:52] <divVerent> no idea [13:52] <skifreak> hahaha, yep, that's dolby [13:53] <divVerent> in "worst case" you may get a file that will work now but may fail in 5 to 10 years [13:54] <divVerent> like, don't put Intel Indeo in mkv ;) [13:54] <divVerent> not that anyone would want to use this codec anyway ;) [13:55] <JEEB> well, at least the AVI compatibility mode is official, so those who have the decoder will be able to decode it [13:55] <divVerent> jeeb: semi-official ;) [13:55] <JEEB> just that if the codec gets a native muxing mode, that's of course better [13:55] <divVerent> the compat mode itself is [13:55] <divVerent> but the fourccs typically not so much [13:55] <divVerent> It'd be new to me if AVI FOURCCs were always officially registered at Microsoft [13:56] <JEEB> now that's a different level of pedanticity :P [13:56] <divVerent> hehe [13:58] <skifreak> divVerent: on using tune and profile: don't see a reason to use profile since I'll only play these files on my computer with mplayer. [13:58] <divVerent> is the avi compatibility mode any more than a simple header info? [13:58] <divVerent> skifreak: profile still kinda sets decoding CPU usage [13:58] <JEEB> the default is high and most people want high for pc playback [13:59] <divVerent> you MAY find out that using baseline profile may keep your fans quiter while playing back [13:59] <divVerent> but you have to find that out yourself [13:59] <divVerent> normally high is good [13:59] <divVerent> jeeb: from what I see in ffmpeg's muxer, the most evil part of the AVI compat mode is that dts, not pts, get written into the timestamp field [14:00] <divVerent> or is there more evil involved? [14:00] <skifreak> divVerent: sound doesn't matter at all to me with my Noctua DH-14, video encoding is the reason I OCed my 3770k [14:01] <iive> avi have timestamp field? [14:01] <divVerent> iive: no [14:01] <divVerent> iive: just seeint ghis in matroskaenc.c that when the AVI compat mode is invoked, write_dts is set [14:01] <divVerent> which means that the matroska muxer will put dts, not pts, in the timestamp fields [14:01] <divVerent> no idea why, of course there are no comments :P [14:02] <iive> oh... i got scared for a moment. [14:03] <jameshowe> so, any ideas for my -newaudio replacement? [14:03] <divVerent> and a similar thing happens on demuxing, when "ms_compat" is set, the demuxer outputs dts, otherwise pts [14:03] <skifreak> divVerent: seems the only tune options I would use is animation, film, and grain. Should I use grain only when the bluray source has grain in the video? [14:04] <divVerent> skifreak: I never tested the latter two... so no idea [14:04] <divVerent> I would assume that too [14:05] <divVerent> grain: for grainy (or really foggy) source material where the grain should be kept and is not filtered out before encoding (see: https://en.wikipedia.org/wiki/Film_grain) the movie 300 is a good example of a grains source [14:05] <divVerent> film: ment for normal film material encoded at a decent datarate (lowers the inloop-deblocking and tweaks the psychovisual settings slightly) [14:05] <skifreak> http://forum.videohelp.com/threads/344451-the-meaning-of-various-tunings-in… yep [14:05] <divVerent> skifreak: that's the URL I just wanted to post as reference [14:06] <skifreak> yea, already found the page before I asked, wanted to see if you had any more insight or could elaborate [14:07] <skifreak> one of the many aspects of bluray I don't understand, is why film grain is present. Why can't better film/masters be made? [14:07] <divVerent> they want to annoy you [14:08] <divVerent> the grain is in fact digital watermarking to track down the movie pirates [14:08] <divVerent> j/k [14:08] <skifreak> lolz [14:08] <divVerent> more seriously: bluray version of an old movie can only be as good as the film material of the movie [14:08] <divVerent> they can't shoot the movie again [14:09] <jacobs> Hello all I am trying to capture an android device screen with this command: /data/local/ffmpeg -y -vcodec rawvideo -f rawvideo -pix_fmt rgb32 -s 320x480 -i /dev/graphics/fb0 /sdcard/output2.avi 2> /sdcard/out.txt . this creates a file with a single ( unclear ) frame and stops. this is the output: http://pastebin.com/LtfZ1Amg . any idea what am I doing wrong ? [14:09] <skifreak> true, but we're talking about movies such as 300 and the Passion [14:09] <skifreak> they're new enough [14:09] <divVerent> and e.g. The Matrix I'd assume to be a bit grainy :P [14:09] <skifreak> nope [14:09] Action: skifreak checks [14:09] <divVerent> or, they may have saved $500 on equipment :P [14:09] <divVerent> because there just was no budget for better stuff [14:10] <divVerent> or, they like grain [14:10] <divVerent> from working in an IT company, I know how it is with these budgets [14:10] <skifreak> The Matrix has a very, very incredibly small amount of grain [14:10] <skifreak> looking at 1.5' from my 55" [14:10] <divVerent> there is always budgets to buy new servers, but never for a single DVD-R for backing up stuff [14:11] <skifreak> I know, but still films such as 300 and the Passion have tons of grain [14:11] <skifreak> 10x more than the Matrix, easily [14:11] <divVerent> nobody knows [14:11] <divVerent> it had a budget of $65 million [14:11] <divVerent> maybe with less grain, it'd have needed $65,050,000 instead [14:11] <divVerent> ;) [14:11] <skifreak> yep, and not enough to buy decent film [14:11] <skifreak> rofl, yep, probably right [14:12] <skifreak> the next bothersome point for me is that there are not any films out at 60fps, and bluray cannot accomedate 60fps 3D [14:12] <divVerent> just like I can't get a single DVD-R for backing up stuff at work here [14:12] <divVerent> there just is no process to get cheap stuff [14:12] <skifreak> wow, makes me glad I left the business world for awhile [14:12] <skifreak> not to say grad school doesn't have its shortcomings... [14:13] <divVerent> luckily, I am external here, we suffer a LOT less from the bureaucrazy[sic] [14:13] <skifreak> nice [14:14] <divVerent> the few "actual employees" mostly have to redeclare operational expenditures (running costs) as capital expenditures (project costs) just to make upper level management happy [14:14] <divVerent> because there is no op-ex budget [14:14] <divVerent> that's why e.g. systems don't get upgraded, they get replaced [14:14] <skifreak> hunh, ok [14:14] <divVerent> in the end it's way more expensive that way, but the upper management can say "we reduced operational costs by 50% again" [14:14] <skifreak> doesn't sound too awfully bad.. [14:15] <skifreak> yea, but what is that to you [14:15] <divVerent> I'm fine, my wages is cap-ex because I am external ;) [14:15] <skifreak> my point :-D [14:15] <divVerent> but it really looks like managing a company into destruction [14:16] <skifreak> true, can't disagree [14:16] <divVerent> and of course this all started since the company was bought by another country [14:16] <divVerent> surprisingly not in USA, but UK [14:17] <divVerent> *company in another country [14:18] <skifreak> yea...definitely wouldn't want to be part of the complicated upper management in that [14:19] <skifreak> the biggest problem I've been trying to get around presently is 3D playback. I am trying to play full3D, don't know if I should go with sbs 3800x1080 or frame sequential. I was thinking frame sequential, unless I could get my tv to recognized my computer as a bluray player and then send framepacked video to it [14:20] <skifreak> I have options for sbs in 3D mode on my tv, but I don't know if it will be able to interpret the different between 1920x1080 video and 3880x1080. Most likely not (or unless it thinks the computer is a bluray player) So I have thought of going with frame sequential [14:21] <skifreak> though how I do make mplayer pick up on the flagged left frames and send these flags to the video card and then the TV? haven't been able to find much of any help on #mplayer or ##hardware [14:22] <skifreak> just thought if anyone would know how to get around the software decoding aspect of the problem, it might be here [14:23] <skifreak> encoding the files doesn't seem to be very hard, it's just the plackback aspect [14:28] <zap0> skifreak, i have pondered the same issues. i have LG 3D TV. [14:28] <zap0> it does SBS and OU, not sure if it does frame seq. [14:29] <zap0> i have heard the TV will not recog 3800x1080 :( [14:29] <skifreak> I have any LG as well zap0, LV9500. It can do frame sequential playback [14:30] <skifreak> ah, yea, I don't know about mine, I would think it should....but I have never connected a bluray player to it [14:30] <zap0> if you did frame seq, how does the TV determine which is left, which is right? [14:30] <skifreak> I think frame sequential would be the best for me, and I can tag the left frames... [14:31] <skifreak> well, there are flags in the 3D file identifying the left frames. These flags should be passed through the video card to the tv [14:31] <skifreak> if the tv wanted the right frames to be tagged instead I can simply change one of the 3d settings on the tv [14:32] <skifreak> same as changing sbs L|R to R|L [14:32] <zap0> back in 5, going out to TV to have a look [14:33] <skifreak> zap0: cool, thanks [14:39] <hendry> anyone have a recipe for shaving the first few seconds from a raw .mkv capture? [14:41] <divVerent> mkvmerge can do this via --split [14:42] <hendry> divVerent: hmm, mkvmerge doesn't seem to be packaged in Archlinux [14:43] <hendry> wondering if there is a general equiv of imagemagick, but for videos, to fade-ins and audio bg tracks programatically [14:43] <divVerent> it is packaged in arch [14:43] <divVerent> the package is called mkvtoolnix-cli [14:44] <divVerent> ffmpeg can use frei0r filters, that's as close as you will get [14:47] <hendry> what does "frei0r" mean? [14:48] <skifreak> zap0: any thoughts [14:52] <jameshowe> re-wrote as a complex filtergraph and it works [14:52] <jameshowe> going to be fun re-writing the ones with even more input... [15:05] <natrixnatrix89> Hi guys I'm struggling with ffmpeg to read mjpeg video stream from an ip camera.. [15:05] <natrixnatrix89> it says "could not find codec parameters" although I've specified -vcodec mjpeg and -s 640x480 http://pastebin.com/vgsQsm2r [15:05] <natrixnatrix89> if I try the same url in VLC, it does play fine.. [15:06] <natrixnatrix89> (although the network connection of camera is very intermittent..) [15:13] <skifreak> zap0: let me know if you find anything, this is the most I've been able to find out http://www.avsforum.com/t/1281356/how-to-create-3d-frame-sequential-video/30 [15:18] <durandal_1707> natrixnatrix89: fetching file with wget and displaying with ffplay works fine [15:19] <jameshowe> does the documentation for alphamerge imply that setpts=PTS-STARTPTS on the inputs is pointless? [15:20] <durandal_1707> natrixnatrix89: use -f mjpeg and not image2 [15:21] <durandal_1707> and you do not need to set vcodec [15:38] <natrixnatrix89> durandal_1707: Thanks! [15:45] <skifreak> seems ffmpeg-1.0 has broken gst-plugins-ffmpeg, for me, can't seem to compile it http://bpaste.net/show/53566/ has anyone else ran into this problem? [15:47] <JEEB> skifreak, most probably it just needs some older API [15:47] <JEEB> can't really do much else but to ask the using application/library to update their usage [15:47] <JEEB> or use the internal copy they seemingly have in there [15:48] <skifreak> JEEB any recommendations? how would I use the internal copy? [15:49] <JEEB> also it seems like that version you're pulling off is made for ffmpeg-0.10.13 ? [15:49] <JEEB> see if your distro has customized it for newer versions [15:50] <JEEB> skifreak, removing --with-system-ffmpeg might do it :P [16:07] <skifreak> JEEB: thanks for the help, it seems this is an oustanding bug (423717) [16:08] <Sashmo_at_work> I could really really use some help. I am at my wits end with my streams going out of sync after 5+ hours. Anyone willing to pick up where I left off, and fix it? I would be willing to pay for the assistance. [16:15] <Silicium> hi there [16:15] <Silicium> i just compiled ffmpeg/ffserver because i had a problem using ffmpeg and h264 [16:16] <Silicium> but i still getting a segfault on: [16:16] <Silicium> ffmpeg -v verbose -r 5 -s 640x480 -r 29.97 -f video4linux2 -i /dev/video0 http://localhost:8080/webcam.ffm [16:16] <Silicium> ffserver is running [16:17] <Silicium> Port 8080 [16:17] <Silicium> BindAddress 0.0.0.0 [16:17] <Silicium> MaxClients 10 [16:17] <Silicium> MaxBandwidth 50000 [16:17] <Silicium> NoDaemon [16:17] <Silicium> <Feed webcam.ffm> [16:17] <Silicium> file /tmp/webcam.ffm [16:17] <Silicium> FileMaxSize 10M [16:17] <Silicium> </Feed> [16:17] <Silicium> <Stream webcam.mp4> [16:17] <Silicium> Feed webcam.ffm [16:17] <Silicium> Format h264 [16:17] <Silicium> #VideoSize 640x480 [16:17] <Silicium> #VideoFrameRate 10 [16:17] <Silicium> VideoBitRate 2000 [16:17] <Silicium> VideoQMin 1 [16:17] <Silicium> VideoQMax 10 [16:17] <Silicium> </Stream> [16:17] <Silicium> oops, sorry [16:17] <Silicium> any idea? [16:17] <Sashmo_at_work> DUDE use pastebin!!!!!!! [16:17] <Silicium> yes i'am sorry [16:40] <tikasukaz> hello every one :) [16:50] <tikasukaz> in presets what is flags2=+wpred+dct8x8+fastpskip [16:50] <tikasukaz> googled can't find any proper documents for "flag2" [16:50] <JEEB> you generally shouldn't care about those, they are minor things and either on or off depending on the preset [16:50] <JEEB> (and profile) [16:52] <tikasukaz> problem is i have two identical server on 1st server encoding is fast (300 fps+ ) on 2nd server encoding doesn't go above 150 fps [16:52] <tikasukaz> 2nd server is new, and much lower cpu load thn 1st server. [16:53] <JEEB> well, use a faster preset in that case, or check for other kinds of things that are limiting the frame rate [16:53] <JEEB> also make sure your libx264 is compiled with assembly optimizations [16:55] <tikasukaz> my friend compiled ffmpeg for ACC3 support he also tweaked few things i guess.. can i use those codes on new server ? [16:55] <tikasukaz> CentOS version are different. [16:56] <tikasukaz> sorry for noob question.. [16:56] <zap0> skifreak, got a bit distracted! [16:57] <zap0> skifreak, only have antenna input, and 3d options would not permit me to move the cursor to the frame-seq icon, only SBS and OU [16:57] <JEEB> uhh, you most probably will want to build your own up-to-date libx264 and ffmpeg [16:57] <JEEB> and probably yasm, since the one that comes with centos will probably be very awfully old [16:57] <zap0> skifreak, maybe frame-seq only works once a compatible HDMI input is active. [16:57] <tikasukaz> JEEB thanks alot.. i will try. [17:01] <JEEB> tikasukaz, http://4.bp.blogspot.com/_D_Z-D2tzi14/S8TRIo4br3I/AAAAAAAACv4/Zh7_GcMlRKo/s… [17:03] <Tjoppen> I like this alot [17:06] <tikasukaz> lol :P [17:06] <tikasukaz> actually it should be "a lot" [17:07] <tikasukaz> i was bad in English classes, mam was too hot :P [20:06] <Yulth> Hi everyone [20:11] <Yulth> I'm encoding HE-AAC and HE-AACv2 to make M4A files, but when I play this files under players like Foobar2000, their properties are the same: AAC audio with SBR for both, but HE-AACv2 doesn't seem to have PS. Anyone have any idea on what is happening? [21:25] <knob> Good afternoon everyone =) [22:06] <zro> im having trouble converting some mp4 files to ogv w/ ffmpeg2theora. it creates files that just blip open in vlc which then reports the message [0x1116ee8] main input error: ES_OUT_RESET_PCR called [22:07] <zro> ffplay open the file and plays the first fraction of a second really fast, then shows a still frame for the duration. [22:07] <zro> any id ea what my issue might be? [22:20] <undercash> hi [22:20] <undercash> it s possible to stream subtitles with ffmpeg? [22:26] <zro> ffmpeg -i input.mp4 -acodec libvorbis -vcodec libtheora output.ogv appears to be making a file that is huge and hiddeous, but plays [22:42] <zro> ya this is ridiculous. i can make it work on certain files, but not tothers, but i cant see what the difference would possibly be [22:46] <zro> here's the diff in poutput from the works and fails streams : http://www.diffb.in/by [23:29] <llogan> zro: ffmpeg2theora is not a ffmpeg project. [23:29] <llogan> also it's hard to see what you're doing without seeing the command and complete console output. [23:30] <llogan> and yes, the defaults aren't always great (excluding libx264). [23:36] <zro> thanks llogan [23:43] <llogan> zro: by "i can make it work on certain files, but not tothers" are you referring to ffmpeg or ffmpeg2theora? [00:00] --- Fri Oct 26 2012

1 0

ffmpeg-devel.log.20121024
by burek 25 Oct '12

25 Oct '12

[00:00] <llogan> lazy ass! that's isn't going to be good, is it? [00:03] <Compn> heh [00:10] <durandal_1707> saste: lol, that patch you pinged [00:11] <saste> durandal_1707, ? [00:13] <durandal_1707> you pinged patch which cannot be applied any more [00:14] <saste> durandal_1707, no i just had to skip an hunk [00:14] <saste> i was lazy to send an updated patch [00:15] <durandal_1707> ok [00:15] <saste> i was tempted to just push, but since that's not code which I "maintain" i preferred to get an explicit approval from someone else [00:34] <cone-265> ffmpeg.git 03Stefano Sabatini 075ffadec30aaa: lavc/utils: propagate or return meaningful error codes in avcodec_open2() [00:34] <cone-265> ffmpeg.git 03Stefano Sabatini 074a227a706d47: lavc/utils: fix a few grammar/style nits in log messages [00:34] <cone-265> ffmpeg.git 03Stefano Sabatini 0733c4535db735: examples/decoding_encoding: add and use the decode_write_frame function [00:51] <cone-265> ffmpeg.git 03Michael Niedermayer 071856162caa91: mpegvideo enc: choose VBV buffer size for mpeg1/2 when none is specified instead of failing [00:51] <cone-265> ffmpeg.git 03Michael Niedermayer 073556e7ce737e: mpegvideo enc: choose VBV buffer size for mpeg4 when none is specified instead of failing [01:33] <llogan> i don't see any commercial stuff on the projects page with my lazy search. are these also allowed to be listed? i don't see why not. [01:33] Action: llogan is going through backlog of requests [01:46] <llogan> and what's the consensus on old/abandoned/homeless stuff? [01:47] <Compn> we generally dont link to commercial stuff much [01:47] <Compn> abandoned stuff, update it to archive.org url [01:49] <llogan> hhmm...i'm not going to be an old link farmer. i'd prefer to dump the shit. [01:51] <llogan> if we don't list commercial stuff "much" the site should say so. [01:56] <llogan> im wondering if the moldering list is even worth "maintaining" (not that it has much maintenance). how much value does it really have? [01:57] <cone-265> ffmpeg.git 03Michael Niedermayer 076c87b2be7257: swscale_unscaled: remove unused variables [01:57] <cone-265> ffmpeg.git 03Michael Niedermayer 07a494792961a0: yuv2rgb: fix integer overflow in fill_table [01:57] <cone-265> ffmpeg.git 03Michael Niedermayer 07cccb45751e93: yuv2rgb: fix integer overflow in fill_gv_table() [02:45] <Compn> llogan : well, the point is, who maintains a list of projects based on ffmpeg ? [02:45] <Compn> its just us [02:45] <Compn> theres no wikipedia list [02:45] <Compn> maybe someone wants an ffmpeg based (whatever) , where should they look ? [02:46] <Compn> i mean, if you dont want to maintain it, thats fine [02:46] <Compn> how about maintaining a list of recommended projects? or what is your idea ? [02:47] <iive> llogan: if the project have vanished, then dump the link. if the project is still accessible, but not changed for a long time, sink it down the list. [02:48] <iive> linking to commercial stuff is sort of advertising, and this is not free. [02:48] <Compn> its alpha betical iive [02:48] Action: Compn wonders if iive seen ffmpeg project page [02:48] Action: iive looks around. Is diego still here? [02:49] <Compn> iive : i dont think he comes here anymore ? [02:50] <iive> he likes to make things in alphabetical order. [03:12] <llogan> Compn: no, maintaing a list of "recommended projects" wasn't my idea, AFAIK. [03:15] <llogan> your points are valid, but i don't think the list is important enough to maintain. [03:17] <llogan> and the little attention the site does get can be better used elsewhere. [03:22] <Compn> i agree [03:22] <Compn> but theres no need to delete it , or any links [03:24] <Compn> oh lawd , 3rd party debate is getting a rocky start [03:24] <Compn> oop wrong window [13:22] <cone-609> ffmpeg.git 03Mans Rullgard 073f65eff408f7: Include sys/time.h before sys/resource.h [13:22] <cone-609> ffmpeg.git 03Mans Rullgard 07da0d0ae9a429: network: add fallbacks for INADDR_LOOPBACK and INET_ADDRSTRLEN [13:22] <cone-609> ffmpeg.git 03Mans Rullgard 07c3e73100af9b: network: use getservbyport() only if available [13:22] <cone-609> ffmpeg.git 03Mans Rullgard 07fab0a8b2c6ca: libm: add fallbacks for various single-precision functions [13:22] <cone-609> ffmpeg.git 03Mans Rullgard 07a805cefd8b81: swscale: avoid pointless use of compound literals [13:22] <cone-609> ffmpeg.git 03Mans Rullgard 0780521c1997a2: build: allow targets to specify extra objects to link with executables [13:22] <cone-609> ffmpeg.git 03Michael Niedermayer 072dbc93455c14: Merge commit '80521c1997a23e148edf89e11b939ab8646297ca' [13:22] <cone-609> ffmpeg.git 03Michael Niedermayer 0742ee9f398197: build: fix detection of math functions with 2 parameters. [13:46] <cone-609> ffmpeg.git 03Mans Rullgard 074ebc6a74108f: build: Plan 9 support [13:46] <cone-609> ffmpeg.git 03Mans Rullgard 072a222a7255af: fate-lavfi: replace sed/grep/cut combos with awk [13:46] <cone-609> ffmpeg.git 03Diego Biurrun 07c2dac8ac7a1f: rtmppkt: Avoid unescaped backslash in Doxygen comment [13:46] <cone-609> ffmpeg.git 03Luca Barbato 070c03cc683864: mp3: exit on parsing error in mp_decode_frame [13:47] <cone-609> ffmpeg.git 03Michael Niedermayer 07f3b8096bc0e7: Merge commit '0c03cc68386443f1e96ab6fb358220faf67cd5ff' [14:08] <cone-609> ffmpeg.git 03Luca Barbato 079ab0874ea8b6: mp3: properly forward mp_decode_frame errors [14:08] <cone-609> ffmpeg.git 03Diego Biurrun 07aa91fe80913f: dxtory: Replace AV_WN16A(AV_RN16A()) combination by AV_COPY16 [14:08] <cone-609> ffmpeg.git 03Diego Biurrun 07af6dd6de08d9: intreadwrite: Add AV_COPYxxU macros for copying to/from unaligned addresses [14:08] <cone-609> ffmpeg.git 03Diego Biurrun 07330b864cda95: dsputil: Replace AV_WNxx(AV_RNxx()) combinations by AV_COPYxxU [14:08] <cone-609> ffmpeg.git 03Reinhard Tartler 07a9f0dce283f8: prepare 9_beta2 release [14:08] <cone-609> ffmpeg.git 03Diego Biurrun 07ceb754d041f5: lzo: Use AV_COPY*U macros where appropriate [14:09] <cone-609> ffmpeg.git 03Michael Niedermayer 075555d2075a26: Merge commit 'ceb754d041f5f6327fd9195a5f43575af9516daa' [14:14] <cone-609> ffmpeg.git 03Mans Rullgard 07d82f18850441: Remove need for padding in av_memcpy_backptr() [14:14] <cone-609> ffmpeg.git 03Michael Niedermayer 075cf794e0c1f7: Merge commit 'd82f188504410fdfa446c5682c128c31bb5851a4' [14:25] <cone-609> ffmpeg.git 03Diego Biurrun 07e831b3b852a2: av_memcpy_backptr: Drop no longer necessary malloc padding [14:25] <cone-609> ffmpeg.git 03Mans Rullgard 07a153e45b953d: dfa: use av_memcpy_backptr() where previously impossible [14:25] <cone-609> ffmpeg.git 03Diego Biurrun 07d7a39b334079: doxygen: Build Doxygen documentation in the doc/ subdirectory [14:25] <cone-609> ffmpeg.git 03Diego Biurrun 075e28e974339e: Move Doxyfile into the doc/ subdirectory [14:25] <cone-609> ffmpeg.git 03Justin Ruggles 07a5ef830b1226: lavc: use the correct API version guard macro for avcodec_encode_audio() [14:25] <cone-609> ffmpeg.git 03Justin Ruggles 0744d854a518f9: atrac3: return an error if extradata_size is not a specific known size [14:25] <cone-609> ffmpeg.git 03Justin Ruggles 07c68317ebbe49: lavc: fix documentation for AVCodecContext.delay [14:25] <cone-609> ffmpeg.git 03Michael Niedermayer 07719fde47ca64: Merge commit 'c68317ebbe4915035df0b08c23eea7a0b80ab881' [14:43] <cone-609> ffmpeg.git 03Anton Khirnov 071bc64c2814d4: lavc: initialize output AVFrame before decoding. [14:43] <cone-609> ffmpeg.git 03Anton Khirnov 078e84f2055c41: avconv: remove now unneeded calls to avcodec_get_frame_defaults(). [14:43] <cone-609> ffmpeg.git 03Anton Khirnov 075a9567631a90: g.723.1: add missing CODEC_CAP_DR1 [14:43] <cone-609> ffmpeg.git 03Anton Khirnov 07292a08cbab33: asfdec: cosmetics, reformat ff_asf_parse_packet() [14:43] <cone-609> ffmpeg.git 03Michael Niedermayer 07d0ab71ed11c5: Merge remote-tracking branch 'qatar/master' [15:30] <cone-609> ffmpeg.git 03Michael Niedermayer 07e377b50b2df9: matroskaenc: avoid implicit truncation in avio_w* [15:30] <cone-609> ffmpeg.git 03Michael Niedermayer 07a3fb3a670e8c: avio: avoid implicit truncation in avio_w* [15:38] <durandal_1707> LOOOOOOL!!!! http://lists.libav.org/pipermail/libav-devel/2012-October/038308.html [15:40] <Compn> durandal_1707 : careful, Daemon404 doesnt like us complaining about libav. :P [15:41] <durandal_1707> well commit is somehow correct, but picked name is not [15:45] <ubitux> i'm not sure that's a good idea to stimulate the flames from other entities with such flag [15:45] <ubitux> but it made me smile :) [15:46] <thegeek> perhaps have a flag for "recommended" codecs instead? :) [15:46] <ubitux> the idea is to stem the generation of bad formats/codecs [15:46] <ubitux> files [15:46] <av500> then IDIOTIC is perfect [15:47] <av500> also, when such codec is used the output filename should have "_made_by_an_idiot_" added [15:47] <durandal_1707> av500: you forgot tags [15:47] <av500> yes [15:47] <av500> and hardcodec SUBS [15:47] <ubitux> av500: yes, but what isn't IMO is that it's completely subjective [15:47] <av500> coded* [15:48] <av500> ubitux: of course it is [15:48] <Compn> why are the tags AV_CODEC_BLAH_BLAH_LOSSY ? cant it just be tags: LOSSY ? [15:48] <av500> how would you objectively define an idiot? [15:48] <ubitux> av500: and so if you start doing that, you will likely end up adding the flag for every existing codec/format [15:48] <Compn> av_our_function_names_are_going_to_get_longer_in_the_future_with_these_ideas [15:48] <ubitux> because everyone has grudge about everything [15:49] <av500> we can maybe spare PCM and YUV :) [15:49] <Compn> they dont compress well [15:49] <ubitux> it's idiot not to compress them [15:49] <Compn> er, what ubitux said [15:49] <ubitux> another problem is that it's not justified [15:49] <durandal_1707> also, i think that IDIOTIC flag is more deserver for containers/formats and less for codecs [15:49] <ubitux> it should at least have a comment with a few arguments [15:49] <ubitux> for each one having the flag [15:53] <durandal_1707> using AV_CODEC_PROP_BAD and AV_CODEC_PROP_UGLY is better [15:54] <ubitux> AV_CODEC_PROP_HATERSGONNAHATE [15:54] <thegeek> michaelni: the bug report for the asm crash I mentioned yesterday can be found here: https://ffmpeg.org/trac/ffmpeg/ticket/1844 [15:55] <michaelni> where do i add AV_FORK_PROPERTY_IDIOTIC ? [15:56] <av500> michaelni: wherever you want, you have commit rights [15:56] <ubitux> AV_PROPERTY_IDIOTIC_FORK? [15:56] <thegeek> :P [15:57] <durandal_1707> michaelni: please feel free to add it wherever it really is, so it can be fixed/rewrited [15:57] <ubitux> i'm not against that AV_CODEC_PROP_IDIOTIC, but the message should be changed [15:57] <ubitux> to something like "this format is generally not recommended" etc [15:58] <ubitux> anyway, let's see how the bikeshed ends up on libav side [15:58] <ubitux> unless they all really hates everything, it might takes a while to get pushed [15:59] <Compn> huh i didnt know google donated a server to libav [16:00] <av500> its only a small data center [16:00] <ubitux> haha [16:00] <Compn> hope that corporate influence does not warp their project [16:01] <av500> mozilla got 300million, libav one server .... [16:01] <Compn> is it me, or is it hard to turn off google 'protection' in firefox now ? [16:02] <Compn> i didnt see an option for it in the prefs, had to go about:config [16:02] <cone-609> ffmpeg.git 03Michael Niedermayer 078a34b85fa8de: avio: detect implicit truncation by assert in avio_w* [16:02] <Compn> previous versions used to ask if you wanted to allow google to know every single website you visited [16:06] <av500> why would you know allow it, they are not evil [16:07] <Compn> just basic computing courtesy [16:07] <Compn> 'dont do things behind the users back without asking' [16:07] <Compn> maybe i'll go ask mozilla whats what [16:08] <Tjoppen> >previous versions used to ask if you wanted to allow google to know every single website you visited [16:08] <Tjoppen> wut? [16:09] <Compn> old firefox used to popup a warning about safe browsing [16:10] <Tjoppen> I see [16:11] <Tjoppen> switching off the auto-google on mistyped URLs took some figuring out [16:11] <Tjoppen> and all other stupid shit firefox does nowdays, like hiding http:// [16:12] <Tjoppen> but it still does when doing auto-complete. no idea how to make it not hide them at all [16:12] <Compn> http://www.mozilla.org/en/firefox/phishing-protection/ [16:12] <Compn> that was the url to explain , back in firefox 2/3 days [16:13] <Tjoppen> ah. I went and disabled it [16:13] <Tjoppen> though it doesn't actually send any addresses to google [16:14] <Tjoppen> time to reply to that idiotic patch [16:14] <av500> Compn: its easy, dont google for Yahoo, just type yahoo.com [16:14] <av500> solved! [16:16] <Compn> you use yahoo as the search engine ? [16:16] <av500> but yahoo has all the itnernet [16:17] <kierank> no bing does [16:17] <kierank> duhg [16:18] <Compn> Tjoppen: heres what it used to do > http://code.google.com/p/google-safe-browsing/wiki/Protocolv2Spec [16:19] <Compn> i turn off autocomplete, autocomplete is for jerks [16:20] <Compn> Tjoppen : er nm, bad url [16:20] <merbanan> I search for porn sometimes just to make google think I'm a pervert [16:21] <av500> google knows [16:21] <Tjoppen> I hear bing is the search engine to use for porn [16:21] <Tjoppen> "google for work, bing for porn" [16:22] <saste> what if you work in porn? [16:22] <av500> then you bing google [16:23] <Compn> i wasnt looking at porn at work, that was for research! [16:23] <av500> Compn: I was only looking at encoding artifacts.... [16:25] <saste> ubitux: I found the culprit of weird pod \, => , conversion [16:25] <ubitux> ! [16:25] <saste> but i think we should avoid "\," in single filter examples [16:26] <saste> well not sure about this [16:26] <saste> if we do foo=ARGS [16:27] <ubitux> :) [16:27] <saste> we're recalling the filtergraph syntax, implying that the snippet can be used literally in a filtergraph [16:27] <saste> anyway it is currently inconsistent [16:27] <ubitux> i have another "escaping" problem if you like them, but not in the documentation [16:27] <ubitux> btw, in the documentation i think we have some '@' issues as well [16:28] <saste> not in the documentation? [16:29] <saste> and today i did cd doc; make clean [16:29] <saste> and discovered it is broken since months [16:33] <michaelni> saste, btw as you memtion docs, i think the example path & patterns in the Doxygen file where forgotten when they where moved to doc/ [16:36] <michaelni> saste, if you want i can look into fixing it or you can ? [16:36] Action: michaelni wonders why doxygen is so slow [16:36] <saste> michaelni, should be trivial change in examples_path and pattern [16:37] <ubitux> <@saste> and today i did cd doc; make clean // was it ever supported? [16:37] <ubitux> we don't support this for libs, tests, etc [16:37] <saste> i never use doxygen (yeah one reason is that it is sloooow to build) [16:37] <saste> ubitux, maybe don't [16:37] <ubitux> <@saste> not in the documentation? // yes, you can't put a '\n' in the option parameter [16:38] <saste> i have no idea what is the "option parameter" [16:38] <ubitux> -foobar $(printf 'hello\n') [16:38] <ubitux> the \n is dropped [16:39] <ubitux> -foobar "$(printf 'hello\n')" [16:39] <ubitux> i don't know what part of the code is doing this [16:39] <ubitux> but it's a problem for some options [16:39] <ubitux> like -headers in the http protocol [16:39] <ubitux> since it requires you to end the string with \r\n [16:40] <ubitux> (try -headers "$(printf 'X-foobar: hello\r\n')") [16:43] <durandal_1707> what subtitles are supported in mp4? [16:43] <ubitux> mov_text [16:44] <ubitux> (that's the name in ffmpeg) [16:45] <durandal_1707> and nothing else? [16:45] <ubitux> i don't think so, at least in ffmpeg [16:45] <durandal_1707> by mp4 spec? [16:48] <JEEB> oh boy [16:48] <JEEB> "MP4" [16:49] <saste> ubitux: texi2html is not escaping @@ [16:49] <ubitux> :( [16:49] <JEEB> there's basically 3gpp subtitles that can be muxed into "MP4" [16:49] <JEEB> you could poke MP4_maniac for details [16:50] <MP4_maniac> j^ [16:50] <MP4_maniac> oRúW_ [16:50] <JEEB> lol [16:51] <JEEB> anyways, it's a mess, but officially'ishly the thing supported is something that is colloquially called "3gpp timed text" or whatever [16:51] <durandal_1707> so subrip is not supported? [16:51] <JEEB> no, not officially [16:52] <JEEB> of course people have made their own proprietary additions to the "MP4" container [16:53] <JEEB> ("MP4" being what is governed by the ISO Media Container and the MPEG-4/3GPP/SMPTE specs, mainly) [16:53] <MP4_maniac> 3gpp timed text is for "3GPP", streaming text format (14496-17) is for "MP4" [16:53] <saste> michaelni, are you already testing Doxygen, or should I do it myself? [16:53] <MP4_maniac> streaming text format != 3gpp timed text [16:53] <michaelni> saste, iam testing but it does not work [16:53] <JEEB> ok [16:54] <JEEB> so there were two types [16:54] <ubitux> <@durandal_1707> so subrip is not supported? // i wonder if the mov text encoder strip them& [16:55] <ubitux> mmh it should actually, since it's based on the ass markup [16:58] <MP4_maniac> JEEB: streaming text format uses timed text formats optimized(?) for streaming. one of formats is 3gpp timed text. others are reserved/undefined yet [17:02] <saste> another problem is that non-ASCII chars are not correctly rendered [17:03] <saste> e.g. the special "x" symbol, both in HTML and MAN output [17:06] <saste> doxygen is taking ages and heating my PC [17:06] <saste> what's the recommended command to run: doxygen Doxyfile? [17:08] <saste> also it seems in loop, always printing the same warnings again and again [17:12] <michaelni> saste, you can set SEARCH_INLCUDES NO to make it run fast [17:14] <saste> now it's not producing any output... [17:15] <saste> enough, I'll post a patch and let's see if someone will be able to test it [17:15] <cone-609> ffmpeg.git 03Michael Niedermayer 077d0ca5b7e436: doxygen: disable header/footer/stylesheet [17:15] <saste> ubitux: texi2html is a monster 23K perl script [17:16] <saste> i managed to fix the @@ escape issue, will send a patch if i'll find the motivation [17:19] <ubitux> have no fear of the monster [17:19] <ubitux> fear is the mind killer [17:22] <cone-609> ffmpeg.git 03Stefano Sabatini 0751467ee86846: doc/filters: remove unnecessary and inconsistent double ',' escaping [17:22] <cone-609> ffmpeg.git 03Stefano Sabatini 0755c77a0ca3ce: doc/filters: escape special Texinfo character @ in overlay example [17:45] <cone-609> ffmpeg.git 03Tomas Härdin 07187630b2449f: mxfdec: Fix CID 732262 [17:45] <cone-609> ffmpeg.git 03Tomas Härdin 071d22d269f54c: mxfdec: Fix a potential DoS vector in mxf_read_pixel_layout() [17:48] <frank_stl> Hi, We are developing one ffmpeg feature. So the update of ffmpeg is important to us. Just wondering does anybody know what is expected date of next ffmpeg version release? [17:49] <saste> frank_stl, one release every 6 months more or less, which feature are you implementing? [17:51] <frank_stl> We are trying to implement the GPU based filter acceleration. Target filters are deshake, scale, sharpening and some other filters. [17:53] <cone-609> ffmpeg.git 03Tomas Härdin 07fbb39f44f1d3: mxfdec: Fix potential leak in mxf_read_local_tags() [17:53] <cone-609> ffmpeg.git 03Tomas Härdin 07e3ba817b95bb: mxfenc: Fix possible integer overflows [17:54] <thegeek> frank_stl: gpl or lgpl? [17:55] <michaelni> saste, iam trying to make a release every 3 month actually :) [17:59] <frank_stl> saste, Thanks. [18:46] <ubitux> frank_stl: deshake could get some nice efficiency improvements before you add gpu acceleration, or it might get meaningless [18:46] <ubitux> (see for instance https://ffmpeg.org/trac/ffmpeg/ticket/1430) [19:12] <michaelni> "<j-b> [mpeg2video @ 0177ab80] a vbv buffer size is needed, for encoding with a maximum bitrate" // <--- should be fixed with latest ffmpeg [19:12] <j-b> michaelni: regression from 0.11.1 to 1.0.0 according to user [19:13] <michaelni> with same vlc ? that would be odd [19:14] <j-b> of FFmpeg [19:14] <j-b> " Maybe it would be worth mentioning that downgrading ffmpeg from 1.0 to 0.11.1 solves the issue. " [19:17] <michaelni> j-b, this cant be, the error message quoted does not exist in ffmpeg 1.0 nor in 0.11.1 [19:18] <j-b> michaelni: user moronic? :) [19:18] <michaelni> it does exist in 0.10 though [19:18] <Compn> is user using packaged ffmpeg ? [19:19] <j-b> Compn: your guess is as good as mine [19:20] <michaelni> j-b, the bug was real i think but the error messages dont match up, 1.0 should have shown "Either both buffer size and max rate or neither must be specified" [19:21] <j-b> who|what introduced the bug? [19:23] <michaelni> j-b, i do not know for sure, but possibly 1d981d16b744b000414cd5a2cc38db90d26d3903 in vlc, there may be other changes in vlc and or ffmpeg that affected it [19:24] <j-b> is the VLC change wrong? [19:24] <j-b> I can revert [19:24] <michaelni> setting a max rate and no buffer size does not work with older ffmpeg nor libav AFAIK [19:25] <michaelni> latest ffmpeg will fill in something reasonable for mpeg1/2/4 instead of failing [19:26] <michaelni> if you dont set a max rate you also dont need to set a buffer size no matter if old or new ffmpeg [19:29] <j-b> so a match up of ffmpeg before 1.0 and the 1d981d16b744b000414cd5a2cc38db90d26d3903 change to 0 [19:30] <michaelni> the fix is not in 1.0 [19:30] <michaelni> just master [19:30] <j-b> but somehow the user says it is broken in 1.0 and not in 0.11.1 [19:31] <michaelni> did he say he was using the same vlc for both ? [19:31] <j-b> yes [19:31] <j-b> 2.0.4 [19:57] <cone-609> ffmpeg.git 03Paul B Mahol 07e12cfd044c2b: lavf: const correctness for avformat_new_stream [20:15] <michaelni> j-b, after several tries, i manages to build 2.0.4 with 0.11.1 and with [20:15] <michaelni> ./vlc -I dummy --ffmpeg-hurry-up --no-loop --no-repeat --ffmpeg-skiploopfilter 4 --audio-language=any --start-time 0 --sout "#transcode{venc=ffmpeg,vcodec=mp2v,vb=7000,acodec=a52,ab=384,samplerate=48000,deinterlace}:std{access=file,mux=ts,dst="test.mpg"}" --sout-transcode-width 480 --sout-transcode-height 320 --sout-ffmpeg-keyint=3 "sample.avi" [20:15] <michaelni> i get [20:16] <michaelni> Either both buffer size and max rate or neither must be specified [20:16] <michaelni> am i doing something wrong ? [20:17] <michaelni> and i needed a few tries to get vlc link to 0.11.1, it prefers to link to the system installed libs it seems [20:19] <michaelni> I think it would be quite usefull if vlc would print the libavcodec/libavformat versions it uses [20:24] Action: michaelni is using: "strings ./modules/codec/avcodec/.libs/libavcodec_plugin.so | grep Lav.5" to find out which version actually gets used [20:25] <michaelni> or ldd for shared ones ... [20:35] <durandal_1707> pgmyuv should have IDIOTIC flag too [20:38] <durandal_1707> or it should be changet to store real width and height [20:45] <durandal_1707> ffprobe seems to ignore return value from decoder/ causing to report values which are not actually correct [20:56] <durandal_1707> what happened to new coverage results? [21:02] <cone-609> ffmpeg.git 03Paul B Mahol 076ddb03caf002: pgmyuvenc: do not create files which are not supported [21:26] <durandal_1707> michaelni: how to get working seeking in voc? [21:28] <llogan> durandal_1707: why say, "Backdoor.PHP.Rst.ap [21:28] <llogan> damn it [21:28] <llogan> This bug report is likely to be ignored. [21:28] <llogan> doesn't give people much confidence to report bugs [21:28] <llogan> (and no, i'm not installing any backdoors) [21:30] <llogan> and i'm assuming you are richardpl [21:35] <durandal_1707> what? [21:36] <llogan> yeah, i obviously failed at communicating. [21:36] <llogan> are you richardpl on trac? [21:36] <durandal_1707> so? [21:36] <llogan> don't tell people their bug reports are going to be ignored [21:37] <durandal_1707> people use bug reports for patches - that is bad [21:38] <nevcairiel> you should thank people for trying to contribute instead of telling them what they all did wrong [21:39] <durandal_1707> so your solution is: post thank on bug report - result bug report is ignored [21:39] <llogan> even better: you should tell that they are ignorant and shouldn't be wasting your precious time. [21:41] <durandal_1707> exactly - they are wasting time sending patches to trac [21:42] <nevcairiel> if any developer wants to fix the bug, he will find the patch. I don't see why it would be ignored just because tehre is a patch in it [21:44] <nevcairiel> its like saying that any bug reports on trac are going to be ignored [21:45] <av500> patches go to the ML, no? [21:46] <av500> who is going to repost them from trac to the ML? [21:47] <nevcairiel> noone is debating that posting a patch to trac reduces the likelyhood of it being applied, we're just pointing out that there are different ways to repond to people that post them on trac, and not all of them have to be all out hostile [21:47] <Daemon404> ceheyos does it properly [21:47] <Daemon404> explains the ysend to the ML [21:47] <Daemon404> in a non-douchey way [21:48] <llogan> michaelni: i recommend adding a note in the default ticket text (summary of the bug, etc) that patches should go to ffmpeg-devel, not trac [21:48] <Daemon404> that sounds like a good idea [21:48] <Daemon404> in giant red text, if possible [21:48] <Daemon404> mayeb blinking [21:49] <JEEB> lol <blink> [21:49] <llogan> either i don't have permisison to change it or i'm too ignorant to see how. [21:49] <llogan> either is highly likely [21:49] <llogan> ...also a page counter with a monkey and some MIDI music would be nice. [21:50] <Daemon404> diablo gif tiled background, you say? [21:52] <llogan> http://wonder-tonic.com/geocitiesizer/content.php?theme=3&music=10&url=ffmp… [21:53] <nevcairiel> thinking back, you really have to wonder how anyone could ever make such pages without their eyes bleeding [21:54] <Genia> i have a user question .. i'm trying to pass a single option into avformat_open_input, for a custom demux to read. in avformat_open_input, there is a bit of code that wants to see s->iformat->priv_class set, in order to pass those options to the demux. where/what should it be set to? [21:54] <Daemon404> nevcairiel, by being 12 [21:54] <Daemon404> green text on black is cool, man. [21:55] <nevcairiel> the only thing back from those days i did was add page counters and use the odd under construction gifs, never a full blown crazy-person site [21:55] <Daemon404> you were clearly never 10-12 with a geocities account [21:55] <JEEB> I remember using some editor with DHTML presets [21:55] <JEEB> for fancy menus and stuff [21:55] <JEEB> back in 1999-2000 [21:56] <llogan> i remember using AOLpress [21:56] <JEEB> and no editor supported UTF-8 [21:56] <JEEB> so I had to either make a page with cyrillics, or one with öäå [21:58] <llogan> my first page comprised of a starfield background and a photo of Inigo Montoya [21:58] <michaelni> llogan, what where should be changed to what ? [22:00] <llogan> michaelni: http://pastebin.com/HgVhVxTW [22:03] <llogan> s/Note that p/P [22:05] <michaelni> llogan, fixed [22:06] <llogan> thanks [22:10] <gnafu> "<@llogan> my first page comprised of a starfield background and a photo of Inigo Montoya" <-- I see nothing wrong with this. [22:14] <cone-609> ffmpeg.git 03Clément BSsch 07c01d1d4ddf4d: http: add -content_type user option. [22:15] <michaelni> nevcairiel, 953a3dcc4e28995072a1148e9d533e8d7b32248c breaks make tools/fourcc2pixfmt.o [22:17] <cbsrobot> it's a fake ! [22:17] <cbsrobot> too much left aligned [22:17] <ubitux> mmh i need to fix fate now [22:18] <ubitux> with the new option saste added [22:19] <michaelni> nevcairiel, simply adding "#undef printf" fixes it [22:29] <cone-609> ffmpeg.git 03Michael Niedermayer 0795d07f33dc5c: libavfilter: include needed header for AVDictionary [22:29] <cone-609> ffmpeg.git 03Michael Niedermayer 07f68ff2e0e79c: fourcc2pixfmt: fix compilation [22:52] <ubitux> http://pastie.org/5111260 [22:53] <ubitux> nevcairiel: do you think this will make the substitution with msys? [22:53] <ubitux> (see the printf) [22:58] <ubitux> nevcairiel: if you have the possibility to try http://b.pkh.me/0001-WIP-use-the-lavfi-graph_file-option-to-workaround-pa.p… i'd be interested [22:59] <ubitux> heh, i can even drop one escape [23:00] <ubitux> mmh actually no [23:03] <j-b> michaelni: http://git.videolan.org/?p=vlc.git;a=commit;h=50a0ceae0ffa535512c5fd8e0952f… something like this? [23:10] <michaelni> j-b, that should fix the failure but if the user intentionally sets min/max rate he might wonder why they have no effect [23:10] <j-b> well, no magic solution [23:18] <ubitux> ok patch updated, should be better [23:18] <ubitux> i hope it works :( [23:28] <ubitux> saste: i used your option [23:29] <ubitux> i might have a patch to fix fate [23:29] <saste> nice [23:29] <saste> but? [23:29] <ubitux> http://b.pkh.me/0001-fate-use-the-lavfi-graph_file-option-to-workaround-p.p… [23:29] <ubitux> but i have no mean to test [23:30] <saste> Daemon404, ^^ [23:30] <ubitux> i'm going to send the patch anyway [23:37] <ubitux> saste: the issue is kind of fun actually [23:37] <ubitux> funny puzzle. [23:41] <cone-609> ffmpeg.git 03Stefano Sabatini 07ce875a613a16: doc/texi2pod.pl: drop unnecessary rule, which was converting "\," to "," [23:41] <cone-609> ffmpeg.git 03Stefano Sabatini 07a161def1e4a5: Doxyfile: update examples path and patterns [23:41] <cone-609> ffmpeg.git 03Stefano Sabatini 07a945607a78ce: ffprobe: add an array of section print buffers to the WriterContext [23:45] <ubitux> ffprobe is getting pretty cute [23:59] <ubitux> saste: thx for the escaping fix :) [23:59] <saste> i'm the escaping man [23:59] <ubitux> :D [00:00] --- Thu Oct 25 2012

1 0

ffmpeg.log.20121024
by burek 25 Oct '12

25 Oct '12

[00:13] <klaxa> Endorgh: http://www.icecast.org/3rdparty.php [00:13] <klaxa> that's the list of icecast source clients [00:13] <klaxa> Endorgh: this is probably what you want: http://www.icecast.org/ices.php [00:14] <Endorgh> but [00:14] <Endorgh> is ices able to strean aac and acc+? [00:14] <klaxa> hmmm... doubt it [00:15] <klaxa> do you have to use icecast or do you just want an http/tcp stream? if you just want an http/tcp stream ffserver could do the job [00:18] <Endorgh> the purpose is to set up a complete and reliable radio station. Icecast does it very well, Can ffmpeg do the same? [00:19] <klaxa> i'm actually not sure [00:20] <klaxa> i rarely use ffserver and stopped using icecast [00:20] <klaxa> i bet there is a way to use ffmpeg as a source client for icecast [00:20] <llogan> perhaps you can pipe from ffmpeg to oggfwd to icecast [00:21] <klaxa> would that work with aac and aac+ as he wishes [00:21] <klaxa> ? [00:21] <llogan> as input, yes, but i was assuming vorbis output [00:21] <Endorgh> I'm confused because major radio stations use icecast and stream aac+, but I don't know what kind of source client they use to stream aac [00:23] <klaxa> Endorgh: this might be of interest: http://stackoverflow.com/questions/5215019/icecast-2-protocol-description-s… [00:23] <Endorgh> llogan: this process implies reencoding? [00:23] <klaxa> as well as this: http://stackoverflow.com/questions/12885597/icecast2-two-sources-same-strea… (although the question is a bit different) [00:24] <llogan> Endorgh: probably since i would guess oggfwd only works with ogg friendly stuff [00:24] <llogan> you could always ask the radio stations what they use [00:25] <klaxa> you could hack something together with shell scripting and netcat (here seems to be a quite good documentation of the icecast protocol: http://forums.radiotoolbox.com/viewtopic.php?t=74 ) [00:25] <klaxa> asking the stations how they do it would be the first thing i'd do before writing something myself... wait actually i'd try to write it myself before asking others :X [00:26] <Endorgh> llogan: I don't think want tell to me this kind of things hehehe [00:26] <Endorgh> *they want [00:27] <llogan> then don't even try [00:27] <klaxa> you could try asking in the icecast channel what source client they would recommend for your usecase [00:28] <Endorgh> okok, thanks for the suggestions! :) [00:29] <Endorgh> klaxa: they say that only can offer support for non-commercial audio formats [00:29] <klaxa> mmh... understandable [00:44] <buhman> ooh [00:44] <buhman> it's tuesday [00:49] <cbsrobot> Endorgh: what do you wanna do exactly ? [00:53] <Endorgh> cbsrobot: I want to stream aac+ audio to an Icecast server [00:53] <cbsrobot> one file ? [00:53] <cbsrobot> playlist ? [00:53] <cbsrobot> dynamic playlist ? [00:54] <Endorgh> yes, directly from a file with no resampling (or playlist, perhaps) [00:54] <cbsrobot> no microphone ? [00:55] <Endorgh> no [00:55] <Endorgh> only files [00:55] <Endorgh> previously converted to aac or m4a container [00:55] <cbsrobot> I'd go with liquidsoap [00:56] <cbsrobot> but in my expereience it's better to use mp3 in the backend [00:56] <cbsrobot> and convert the stream to aac+ at the end [00:57] <Endorgh> exactly [00:57] <Endorgh> but liquidsoap works with raw data, and it implies to reencode using CPU resources [00:57] <Endorgh> for me, this is a problem [00:58] <cbsrobot> not sure [00:58] <cbsrobot> maybe there is a way to do it without reencoding [01:00] <Endorgh> I haven't found out the way to do it. Any suggestions? [01:04] <Kireji> I was using a voicerecorder to record a meeting at work and my HTC G2 ran out of power and shut down [01:04] <Kireji> the resulting 3gp file is not playable with VLC or mac Quicktime - how can I go about repairing it and getting what audio I can from the file? [01:04] <Kireji> http://www.3gprecovery.com/ has some videos and forum posts, but I have low confidence they are reputable [01:04] <Kireji> when I open it with "ffmpeg -i broken.3gp out.mp3" I get broken.3gp: Error while opening file [01:05] <Kireji> specifically "[mov,mp4,m4a,3gp,3g2,mj2 @ 0xdff260]moov atom not found" [01:05] <Kireji> what can I do? it's got 8.5MB of data we really want [01:06] <cbsrobot> Endorgh: no it doesn't seem so [01:07] <durandal_1707> Kireji: recreate moov atom [01:08] <Kireji> ok, that sounds great, can ffmpeg do that? [01:08] <Kireji> man'ing [01:09] <durandal_1707> Kireji: no, what audio codec was used? [01:17] <Kireji> I'm not sure, I think it's mp4 [01:17] <Kireji> it's whatever is default on an Android HTC G2 [01:18] <Kireji> I'm playing now with AtomicParsley and it's saying there are 0 tracks, I have no idae what I'm doing [01:19] <durandal_1707> Kireji: it does not output anything instead of that? [01:20] <Kireji> I'm looking now at working recordings made with the same program, and it outputs a bunch [01:21] <Kireji> http://i.imgur.com/TGkoj.png [01:21] <Kireji> working and broken .3gp files [01:22] <Kireji> durandal_1707: I may have misunderstood your question [01:24] <Kireji> a few different places (like http://blog.alwayshere.info/2011/01/ffmpeg-moov-atom-not-found.html) says this program may be able to add or fix moov atoms [01:24] <dislo> Hey Everyone, I am trying to take a RTSP stream in and serve a RTSP stream back out for android phones and the apple http streaming protocol for iwhatevers. Can I do this using ffconv and ffserver? [01:25] <llogan> what's ffconv? [01:25] <dislo> llogan, ffmpeg sorry [01:32] <durandal_1707> Kireji: Kireji only soluton i see it to learn mov format open hex editor and try to find and save audio data, to make use of saved data you will also need to know bitstream of audio codec and you will need to manualy mux audio packets into containter that support such codec [01:33] <Kireji> I have a bunch of other examples of correctly recorded files from work, that should all have the correct headers [01:33] <Kireji> like the other example in that picture [01:34] <durandal_1707> muxing should not be hard... [01:34] <Kireji> would it work to take another recording, rip out all the data and paste in the longer data from the broken file? [01:34] <Kireji> I don't know what muxing is [01:35] <durandal_1707> Kireji: moov atom is located at end of file [01:36] <Kireji> ah [01:36] <Kireji> crap [01:37] <durandal_1707> you need to locate each audio packet in file and remux it [01:37] <Kireji> this something a program can do, something like a repair program, or something that takes human effort? [01:37] <durandal_1707> thats why it is imortant what audio codec was used [01:38] <Kireji> if human work, not worth it, it was an important meeting, 8 of us were talking for 1.5h. not going to spend sig $ on it [01:38] <durandal_1707> it is probably amrwb... [01:38] <Kireji> ah ok, let me see if I can get codec from another successful recording from same program [01:38] <Kireji> amrwb? [01:39] <durandal_1707> Adaptive Multi-Rate WideBand [01:39] <durandal_1707> or Adaptive Multi-Rate NarrowBand [01:40] <durandal_1707> probaby some trivail parser could be written taking several assumptions .... [01:40] <Kireji> can ffmpeg tell me the codec? [01:40] <Kireji> man'ing [01:40] <durandal_1707> ffmpeg -i input.file [01:41] <Kireji> Stream #0.0(eng): Audio: samr / 0x726D6173, 8000 Hz, mono, s16 [01:42] <Kireji> is that it? [01:42] <Kireji> or [01:42] <durandal_1707> samr is fourcc [01:43] <durandal_1707> for AMRNB [01:44] <durandal_1707> and looking at code it is not hard to parse....., but that exercise is left to the reader [01:44] <Kireji> ok [01:44] <Kireji> thank you [02:08] <RunawayDevil> which codec should I use to make a DVD HD (29.4 Mbps)? [02:09] <RunawayDevil> using mpeg2video the maximum is 9.4mb (to make a vob) [02:09] <klaxa> DVD standard is mpeg2 if i'm informed correctly [02:09] <klaxa> hmm... [02:09] <RunawayDevil> standard is that :/ [02:10] <RunawayDevil> it's because I have here many DVD DL, and it's very bad when I see just 3 gb full and 5 empty, it's like wasting [02:10] <RunawayDevil> so i've decided to up [02:10] <klaxa> i can imagine [02:12] <RunawayDevil> anyone else? [02:12] <klaxa> what is the source's bitrate? [02:15] <RunawayDevil> https://www.mercadolibre.com/jms/mlb/lgz/login/otp?otp=044bcc791a300ccbb240… [02:15] <RunawayDevil> ops [02:15] <RunawayDevil> sorry lol [02:16] <RunawayDevil> klaxa it's 14mb [02:16] <klaxa> so you want to push it up to 29.4 mbps? [02:16] <klaxa> just to fill the dvd? [02:16] <RunawayDevil> maybe yes, obviously i will have some gain in quality, no [02:16] <RunawayDevil> ? [02:17] <klaxa> you could gain some quality with post processing i guess... but just pushing up the bitrate usually doesn't increase quality [02:17] <klaxa> but wastes space [02:18] <klaxa> do you want to play it with your dvd player or put the dvd's in a computer? [02:18] <RunawayDevil> dvd player in 1080p [02:19] <klaxa> is that even possible? [02:20] <RunawayDevil> im trying near as possible [02:20] <klaxa> i haven't seen many dvd players, but none had a port that supports that resolution [02:22] <klaxa> according to wikipedia (yay best source) the DVD format is a restricted mpeg2video definition as you already seem to know [02:22] <klaxa> http://en.wikipedia.org/wiki/DVD-Video#Frame_size_and_frame_rate [02:23] <klaxa> it's capped at 9.8 mbit/s [02:23] <klaxa> probably (i'm almost certain) because of DVD player's read-speed [02:26] <klaxa> http://en.wikipedia.org/wiki/DVD-Video#Data_rate <-- this appears to support my statement [02:26] <cbsrobot> RunawayDevil: were you able to hardsub ? [02:26] <klaxa> again wikipedia, probably not the most reliable source [02:27] <RunawayDevil> cbsrobot: now yes, I'm to make a .vob first I convert .srt to .ass in jubler and so I hardsub with iFFmpeg [02:28] <RunawayDevil> I use jubler to edit stuff too [02:28] <RunawayDevil> cbsrobot http://s11.postimage.org/r51ucn1pv/Captura_de_Tela_2012_10_23_s_22_25_38.png [02:28] <RunawayDevil> that's my config, what do you think? [02:29] <cbsrobot> can you watch hd dvd ? [02:29] <klaxa> what cable do you use to connect your dvd player to your tv? [02:29] <cbsrobot> it's like eol for hd dvd: http://en.wikipedia.org/wiki/HD_DVD [02:29] <cbsrobot> most use bluray nowadays [02:30] <RunawayDevil> HDMI [02:30] <cbsrobot> but you have a hd dvd player ? [02:30] <RunawayDevil> yes [02:30] <cbsrobot> ah ok [02:31] <RunawayDevil> but for a simple DVD with good quality [02:31] <RunawayDevil> do you think is that good? http://s11.postimage.org/r51ucn1pv/Captura_de_Tela_2012_10_23_s_22_25_38.png [02:32] <cbsrobot> hmmm [02:33] <cbsrobot> nah [02:33] <cbsrobot> if you have a "HD DVD" Player [02:33] <cbsrobot> I'd encode in avc [02:33] <cbsrobot> not in mpeg2 [02:34] <RunawayDevil> hum [02:35] <klaxa> was about to suggest that before my router decided to fuck up my wifi [05:40] <dragunov_11> Hi, am having problem in transcoding a mjpeg stream from a camera to h264 stream over rtsp. http://pastebin.com/jnefyUiA [05:41] <dragunov_11> says tcp connection failed, connection refused. [09:34] <elkng> how to extract frames from 400 to 500 from video using frames number as parameter ? [11:07] <jalalsfs> Hi all, [11:08] <jalalsfs> I tried to record my desktop with ffmpeg but it doesn't work http://fpaste.org/BoLP/ [11:12] <elkng> jalalsfs: try this one: "ffmpeg -f x11grab -r 25 -s 1024x768 -i :0.0+100,200 output.flv" [11:26] <jalalsfs> elkng, I looks the same look http://fpaste.org/Tm39/ [11:29] <elkng> jalalsfs: do you have radeon video card ? [11:30] <Yulth> Hi everyone [11:31] <gazzwi86> I have an issue with ffserver [11:31] <gazzwi86> I'm trying to stream my webcam feed and can only get a download in chrome, one which doesn't download video [11:32] <gazzwi86> from status.html it says its transferred 66bytes but no more [11:34] <jalalsfs> elkng, no, it's Intel. [11:44] <Yulth> How I can specify an output format when writing to stdout? This command line is giving me an error: ffmpeg -i input.m4a -codec copy -f m4a pipe:1 [11:53] <Yulth> any ideas? [12:06] <Sashmo> hey guys, anyone have an idea on why I begin to get so much audio drift in my live encodings from transport stream? I find that I have to restart my encoding every 5+ hours to fix the audio sync? [12:21] <Yulth> any ideas? How I can to write to stdout m4a format? [12:33] <brontosaurusrex> Yulth: perhaps ffmpeg -i input.m4a -acodec copy -f mp4 pipe:1 [12:38] <Yulth> Just at this moment I found out the solution: the format must be specified as "-f astd", because audio was aac [12:38] <Yulth> sorry, "adts" [12:39] <JEEB> that would be raw aac [12:39] <JEEB> naturally, mp4 without fragments would not be open'able straight away, so it kind of makes sense [12:42] <Yulth> mmm [12:43] <Yulth> and is there any way to make a copy of m4a input file, without reencoding, to m4a stdout? [12:43] <Yulth> preserving metdata, etc... [12:51] <klaxa> Yulth: cat input.m4a | <next command to use the m4a from stdin> [12:55] <Yulth> ok, here the pastebin: [12:55] <Yulth> http://pastebin.com/FE1XdXWy [12:58] <burek> pipe:: Invalid data found when processing input [12:59] <burek> what are you trying to do? [12:59] <burek> remux m4a to aac? [13:00] <burek> try: ffmpeg -i Fisher_40k.m4a -acodec copy -f adts output.aac [13:47] <jameshowe> ffmpeg 1.0 CLI - is there a way to pass the same encoder output to multiple muxers in a single command? [14:18] <jalalsfs> I can't record from my desktop look at this http://fpaste.org/ZAOc/ [14:45] <burek> jameshowe http://ffmpeg.org/trac/ffmpeg/wiki/Creating%20multiple%20outputs [14:45] <burek> maybe that can help [14:56] <divVerent> burek: basically, the site says "no, ffmpeg can not do it, but the shell can" ;) [14:57] <divVerent> one extra hint for jameshowe: try -f nut as format to be piped, because nut can contain "anything" unlike mpegts [15:17] <teolicy> Hi. I'm trying to convert an mp4 stream to a raw YUV stream with pipes. [15:17] <teolicy> As you can see here (http://pastebin.com/EHAHHK1H) when I read the input file and write to an output file, my command line works. [15:18] <teolicy> However, when I try to use pipes, my command line fails. [15:18] <teolicy> The command line I used (with output) is in the paste, but I'll also copy it here: [15:18] <teolicy> ffmpeg -i - -an -r 25 -f rawvideo -vcodec rawvideo -pix_fmt yuv420p - < input.mp4 > output.mp4 [15:18] <teolicy> And the input file (just a sample) can be retrieved here: https://s3.amazonaws.com/audish/input.mp4 [15:19] <teolicy> I tried playing a bit with analyzeduration and probesize, but not sure I used good numbers and didn't get better results. [15:19] <teolicy> I suspect I should specify a -f and a -vcodec for my input, but not sure what to specify. [15:19] <teolicy> Any help will be appreciated. [15:52] <jameshowe> the trouble with doing it in multiple piped commands is that the MOV/MPG muxer can't deal with adts aac data unless it's just come out of the encoder [16:21] <teolicy> I ran qt-faststart on my source and now it works fine through the pipe. Great! [16:21] <ubitux> teolicy: you can remux with -movflags +faststart as well [16:22] <teolicy> ubitux: you mean, when creating the original source, right? [16:22] <ubitux> ffmpeg in-nofaststart.mp4 -c copy -map 0 -movflags +faststart out-faststart.mp4 [16:22] <ubitux> something like this [16:22] <teolicy> aye, thanks. [16:25] <jameshowe> just -movflags faststart, no + [16:26] <ubitux> +faststart adds to the default flags [16:26] <marlinc_> Any one in here who would like to try my live streaming application while it isn't fully ready yet [16:26] <marlinc_> Need some suggestions and stuff [16:30] <Simex> Someone? [16:30] <dTal> you didn't exactly sell it :p [16:31] <Simex> I know but it is true :p [16:31] <Simex> Would you like try it? [16:31] <Simex> It is open-source [16:47] <rickbol> when transcoding a DV stream to mpeg4, can ffmpeg produce a "streamable" mpeg4? i.e. move the MOOV attribute (and what others) to the file header, etc? [16:47] <rickbol> where can I find out how to construct the necessary parms to make this happen? [16:47] <ubitux> -movflags +faststart [16:48] <JEEB> of course that only means that it will become like that /after/ the encoding has finished [16:48] <JEEB> if you need to have access to output ASAP [16:48] <JEEB> you need the fragments feature [16:48] <JEEB> I think libavformat's muxer should support that? at least the parser does. [16:48] <ubitux> fragment? yes we have such thing [16:48] <ubitux> and we also have a segmenter [16:49] <rickbol> I would like to be able to view the trancode result (confidence monitoring) more-or-less in real-time. [16:50] <JEEB> yeah [16:50] <JEEB> fragments needed [16:52] <rickbol> ultimately the resultant file will be posted for streaming from a website. Will fragments "complicate" that? [17:04] <JEEB> rickbol, I wish I remembered the details of that (aka if the main index is generated or not) -- I know that some parsers probably are going to fail with just fragments [17:46] <ogodon> total noob here, is using ffmpeg for converting .mkv HD into some SD format a bad idea? [17:46] <ogodon> or is it like using a flamethrower to light a candle? [17:51] <rickbol> ogodon: ffmpeg is among the most comprehensive tools for trancoding (converting) one format into another. [17:52] <ogodon> thanks rickbol, I'll give it a shot then, I tried it some time ago and got frustrated, but I'm hoping for better results now [17:54] <rickbol> the syntax isn't easy... and you have to know what your goal is, but if you work with audio\video, ffmpeg is worth the effort it takes to learn it. [17:57] <ogodon> thanks, any clues as to where to catch up with the syntax? [18:07] <dragunov11> Hi, I needed help in transcoding a mjpeg stream to x264 stream, tried following loads of tutorials from google search, but doesn't work. can you guys help please? [18:09] <rickbol> ogodon: google for your specific case. example... ffmpeg mkv to avi [18:10] <rickbol> is ffmpeg changing name to avconv? [18:16] <durandal_1707> rickbol: nope [18:16] <ogodon> I did, didn't find it, but I think I worked it out... it's going to take a while to see if it worked though [18:17] <rickbol> durandal_1707: ubuntu (ubuntustudio) is deprecating ffmpeg for avconv? wtf? [18:20] <ubitux> rickbol: it's just propaganda, see above [18:28] <ogodon> it works!!!!! thank you rickbol, you were right... :-) [18:39] <Yulth> Hi everyone! [19:01] <Yulth> It's known that HE-AAC is a non-free codec, and in all forums, chats, etc people say that one have to pay or adquire a valid license to use HE-AAC (or any variant) legally. It's all right? [19:02] <durandal_1707> yes [19:02] <Yulth> ok [19:06] <Yulth> but reading the license requirements, etc, I've found out this webpage, on supposedly is shown all information about this regard. http://www.vialicensing.com/licensing/aac-faq.aspx [19:06] <Yulth> and curiously, it shows literally: [19:06] <Yulth> "License fees are due on the sale of encoders and/or decoders only. There are no patent license fees due on the distribution of bit-stream encoded in AAC, whether such bit-streams are broadcast, streamed over a network, or provided on physical media. " [19:07] <Yulth> Does it mean that online radio stations are free to stream AAC channels without paying fees? [19:07] <Yulth> Or how it works? [19:08] <JEEB> IANAL and I don't know the full license, but it seems that if you don't distribute the decoder or encoder, you should be OK..? [19:09] <JEEB> but you probably will need to poke them about getting the license itself [19:11] <Yulth> So, Can an online radio station stream HE-AAC audio without problems? [19:11] <Yulth> for me is very confusing... [19:15] <Yulth> anybody have some knowledge on this matter? [19:19] <JEEB> Yulth, ask via licensing if you want a proper answer [19:19] <JEEB> but I would say that in most cases they really wouldn't give two derps, and IANAL [19:20] <JEEB> I mean, it's funny how you actually ask that while everyone else just assumes that because H.264 can be used on the web for free if you distro it for free [19:23] <Yulth> ok, I understand [19:26] <durandal_1707> JEEB: so he can stream only if it stream from free OS? [19:28] <JEEB> durandal_1707, uhh what [19:28] <JEEB> what does that have to do with anything [19:29] <durandal_1707> JEEB: nvm, i misread it [19:29] <JEEB> the pasted point just notes that distro'ing the dec/encoders is payware, and otherwise you just have to get the paperwork done if you only distro the bit stream [19:30] <JEEB> (I will guess they have the same thing as MPEG-LA, which means you have to get the paperwork done even if you don't pay) [19:37] <k1ng__> hello [19:38] <k1ng__> i am trying to convert mov to x264 mp4 but i am only getting audio [19:38] <k1ng__> http://codepad.org/kAwwtI9P [19:38] <dericed> k1ng__: a guess, but try adding -pix_fmt yuv420p [19:41] <k1ng__> ok let me try [19:41] <k1ng__> cool [19:41] <k1ng__> thanks a lot mate, its worked :D [19:41] <k1ng__> dericed, you rock [19:53] <dericed> k1ng__: no problem [19:53] <k1ng__> :) [21:36] <jalalsfs> Hello there. [21:37] <llogan> jalalsfs: hello [21:38] <jalalsfs> Hello llogan, hwo are you? [21:41] <jalalsfs> When I try to record desktop by FFmpeg it doesn't work look at this http://fpaste.org/CHfx/ [21:42] <llogan> ffmpeg tells you that when your -s exceeds the actual screen size [21:42] <llogan> also, don't use sameq [21:42] <llogan> http://superuser.com/a/478550/110524 [21:43] <llogan> and here is a good guide: http://ubuntuforums.org/showthread.php?p=8746719#post8746719 [22:07] <tr0m> could someone kindly have a look at http://pastebin.com/yjGRPHNE please, i cant script... [22:08] <tr0m> maybe im missing something, the script works on fc systems but not so far on centos [22:09] <llogan> 4mv is now mv4. [22:10] <llogan> if you're running the same script on different ffmpeg versions you will likely experience syntax differences [22:10] <tr0m> yes [22:10] <tr0m> makes sense [22:11] <wm4> what are libswscale's semantics about memory between end and start of a line? (i.e. that space that is used for alignment or when the image is a cropped view of a larger image) [22:11] <wm4> i.e. the area between the pixels (w,y) and (0,y+1) [22:13] <tr0m> llogan: any ideas on the (+aic) part [22:13] <llogan> do you get a new error now? [22:14] <tr0m> yah [22:14] <wm4> it seems libswscale generally likes to stomp over some bytes past the last pixel of each scanline because of SIMD operations [22:15] <tr0m> [msmpeg4 @ 0x89d3700] 4MV not supported by codec [22:16] <durandal_1707> wm4: such bytes should be 0? [22:16] <llogan> tr0m: remove it [22:16] <tr0m> k [22:16] <wm4> durandal_1707: it does write black to bytes it stomps over [22:17] <wm4> durandal_1707: however, I want libswscale to write into a cropped image, without destroying the outside areas in the outer image [22:17] <tr0m> [wmav2 @ 0x8517da0] output buffer size is too small [22:17] <tr0m> audio encoding failed [22:18] <durandal_1707> wm4: i doubt that is possilbe, because of simd stuff, but ask michaelni anyway [22:18] <wm4> michaelni: see above [22:19] <llogan> tr0m: i don't know. i've never encoded with msmpeg4/wmav2. use pastebin to show the command and the complete output. [22:20] <tr0m> alright will inabit coffee & cig time [22:20] <tr0m> thx [22:23] <michaelni> wm4, i aggree it would be ideal if swscale didnt write into the alignment padding. a patch improving this is welcome if it doesnt slow thngs down [22:49] <tr0m> llogan: http://pastebin.com/mQuE6epw [22:50] <wm4> michaelni: I suppose ideally, libswscale would use the slow path for the last pixels in a scanline, but I certainly don't know swscale well enough to know if that's even possible [22:53] <wm4> also, why does libswscale require a minimum image size? [23:01] <mancha> hi folks. i am building mplayer svn@today with embedded ffmpeg git@today. i get lots of warning like this: ffmpeg/libavutil/libm.h:36:6: warning: "HAVE_ATANF" is not defined [23:01] <mancha> any ideas? [23:02] <mancha> seem i get 8 warning scroll by on most files (8 math functions, atanf atan2f powf, etc) [23:03] <mancha> did a recent commit maybe mess up the headers a bit? [23:08] <michaelni> wm4, honestly i dont remember, but you can remove the size check and debug the problems that occur with smaller sizes [23:13] <mancha> i think 80521c1997a23e might be the culprit [23:16] <llogan> burek: hows it going with the pi? [23:20] <llogan> damn, tr0m left just as i had an answer for him [23:20] <llogan> (probably bug 1104, try git master) [00:00] --- Thu Oct 25 2012

1 0

ffmpeg-devel.log.20121023
by burek 24 Oct '12

24 Oct '12

[00:00] <ubitux> "movie=f='c:\foo\bar'" doesn't actually receive "f='c:\foo\bar'" as arg, but "f=c:\foo\bar" [00:00] <saste> but it won't work in the general case, e.g. if the filename contains "'" [00:00] <saste> ubitux, yes because of the double escaping [00:00] <ubitux> ...so when running a av_get_token in the src movie filter... [00:00] <ubitux> it will no honor the non-existing quotes [00:00] <ubitux> and so it fails :( [00:00] <ubitux> not* [00:00] <saste> keep in mind that we run av_get_token two times [00:01] <saste> first when reading the filter description in the filtergraph [00:01] <saste> so we read until the next "," or ";" [00:01] <ubitux> yeah right [00:01] <saste> then we use again av_get_token to read the various arguments, separated by ":" [00:01] <ubitux> also there is a second issue, which we can't do much about in the mingw/msvc env [00:02] <saste> that is? [00:02] <ubitux> basically msys seems to do some sed to translate the paths [00:02] <ubitux> but it doesn't detect the path when mixed in the filtergraph in the case of the sws flags [00:02] <ubitux> http://fate.ffmpeg.org/report.cgi?time=20121022212821&slot=x86_32-msvc10-wi… [00:02] <saste> ouch [00:02] <ubitux> look how the two tests fail differently [00:03] <ubitux> Error initializing filter 'movie' with args '/d/Dev/ffmpeg/fate/samples/svq3/Vertical400kbit.sorenson3.mov' [00:03] <ubitux> this one is because the path was not transformed [00:03] <ubitux> (not detected with the "sws_flags=+accurate_rnd+bitexact;", adding a space after may or not help) [00:03] <ubitux> and the silencedetect failure is because of the issue we were just discussing [00:04] <saste> we can workaround that [00:04] <ubitux> ah? [00:05] <saste> we just need to read the filtergraph from a file [00:05] <ubitux> :) [00:05] <ubitux> the silencedetect failure, right? [00:05] <ubitux> (that's what you meant by "that") [00:06] <saste> why it is not finding "/d/Dev/ffmpeg/fate/samples/svq3/Vertical400kbit.sorenson3.mov"? [00:06] <ubitux> because that path doesn't exist [00:06] <ubitux> msys is supposed to "sed" [00:06] <saste> what's the real filename? [00:06] <ubitux> but it doesn't recognize it when mangled with the sws_flags=... [00:06] <ubitux> /Dev/ffmpeg/fate/samples/amrwb/seed-12k65.awb i guess [00:07] <ubitux> oups no [00:07] <ubitux> d:/Dev/ffmpeg/fate/samples/amrwb/seed-12k65.awb [00:07] <ubitux> so.. [00:07] <ubitux> d:/Dev/ffmpeg/fate/samples/svq3/Vertical400kbit.sorenson3.mov [00:07] <cone-571> ffmpeg.git 03Michael Niedermayer 0745ae9a8fc903: j2k: reorder multiplications to avoid interger overflows related to code blocks [00:07] <cone-571> ffmpeg.git 03Michael Niedermayer 07da317efd9256: j2k: reorder multiplications to avoid interger overflows related to precincts [00:08] <saste> if you read the arguments from a file, it shouldn't mangle the filenames [00:08] <saste> something like: [00:09] <saste> ffprobe -f lavfi -lavfi_graph_file=ffprobe_scenedetect_lavfi_graph ... [00:10] <saste> for the silencedetect problem: "amovie='$(SRC)',silencedetect=d=.1" [00:10] <saste> for the silencedetect problem: "amovie=\\'$(SRC)\\',silencedetect=d=.1" [00:10] <ubitux> how do you put the SAMPLES path into the lavfi graph? [00:10] <ubitux> (file) [00:10] <ubitux> generated at runtime? [00:11] <saste> it's complicated, but you can generate the file with a cat command [00:12] <ubitux> the \\ seems to work [00:13] <ubitux> amovie=\'c:\foo\bar\':sp=3,silencedetect: Protocol not found [00:13] <ubitux> haha [00:14] <ubitux> i guess it will behave differently in a win environment? :) [00:14] <saste> yes it is the cervellotic heuristic used for protocol recognition [00:14] <ubitux> (i just made a symlink c:\foo\bar -> test.mp3) [00:15] <ubitux> do you want to commit the \\ workaround? [00:15] <saste> me? [00:15] <ubitux> sure [00:15] <saste> no i don't want to take the blame if it borks [00:15] <ubitux> :D [00:16] <saste> well i'd need to verify that it works as intended in windows [00:16] <ubitux> i don't have any to test [00:17] <ubitux> maybe it's not worth the effort if we're going to use the file method [00:17] <saste> well it shouldn't broke *nix, and windows is already broken so it can't make things worse [00:17] <saste> possible [00:17] <ubitux> so yes we can fix the silencedetect with this [00:18] <ubitux> now we need something for the other one :p [00:18] <saste> do you want me to implement -lavfi_graph_file? [00:18] <ubitux> if it solves the issue, could be nice yes [00:18] <saste> i don't know if i'll do it today or even tomorrow [00:19] <ubitux> how do you plan to do that btw? [00:19] <saste> but if you can wait up to two days... [00:19] <ubitux> -f lavfi -i "graph_file=foobar.lavfi" ? [00:19] <saste> i'm too tired to do it now [00:20] <ubitux> i would have done it if i had time, but it's getting late... [00:20] <saste> no, -i takes the title, if lavfi_graph is not defined it use the title as a filtergraph description [00:21] <ubitux> mmh [00:21] <saste> so it is -lavfi_graph_file -f lavfi -i "foobar" [00:21] <saste> * -lavfi_graph_file FILE -f lavfi -i "foobar" [00:22] <ubitux> this will be a lavd/lavfi.c AVOption, right? [00:22] <saste> yes [00:22] <ubitux> what kind of dummy thing will you put for -i? [00:23] <saste> -i "saste rulez" [00:23] <ubitux> sounds nice [00:23] <ubitux> will you quote 'saste'? [00:24] <saste> seriously -i "silencedetect test" [00:24] <saste> or something like that [00:24] <ubitux> ah it will act as a filtergraph name, ok :p [00:29] <cone-571> ffmpeg.git 03Michael Niedermayer 073318d6b94098: ffmpeg: check avpicture_get_size() retuen value [01:19] <ubitux> saste yay! [01:20] <ubitux> thanks a lot :) [01:24] <cone-571> ffmpeg.git 03Paul B Mahol 07a57d560db3a1: LVF demuxer [02:33] <cone-571> ffmpeg.git 03Michael Niedermayer 077770c0da0731: fate/mapchan: add bitexact flag [03:04] <ohsix> is the point of the bot to badger people or what, it doesn't seem to get much reaction [03:12] <cone-571> ffmpeg.git 03Michael Niedermayer 07e9b61fffbe43: g722dec: s/Libav/FFmpeg/ The copyright years predate the existence of Libav [03:12] <cone-571> ffmpeg.git 03Michael Niedermayer 0734ccb94796b7: g723_1: remove unneeded cliping [03:16] <michaelni> ohsix, it got you to react, i admit i fixed it already before the next bunch of boxes tested the old pre fix version [03:40] <creep> h [05:03] <ohsix> michaelni: well, it didn't make me react in a manner that fixed the failure :] [09:22] <ubitux> is it possible from the URLContext to acccess the AVFormatContext? [09:22] <ubitux> i guess no, but well.. [11:00] <ubitux> mpf, headers option in http protocol doesn't add up, it replaces [11:01] <ubitux> :( [12:40] <cone-265> ffmpeg.git 03Anton Khirnov 07b691135d0c6a: lavc: remove stats_out from the options table. [12:40] <cone-265> ffmpeg.git 03Anton Khirnov 075957aefc7c75: libxvid: remove useless doxy comments. [12:40] <cone-265> ffmpeg.git 03Mans Rullgard 07abdee95224fe: pcm: define AVCodec instances only for enabled codecs [12:40] <cone-265> ffmpeg.git 03Justin Ruggles 07e55d53905f34: atrac3: cosmetics: pretty-printing and renaming [12:40] <cone-265> ffmpeg.git 03Justin Ruggles 07aefdb735c3df: atrac3: simplify some loop indexing [12:40] <cone-265> ffmpeg.git 03Justin Ruggles 075ac673b5531d: atrac3: use AVCodecContext.channels instead of keeping a private copy [12:40] <cone-265> ffmpeg.git 03Michael Niedermayer 07dcb0d1193a3d: Merge commit '5ac673b5531d846b79a3d77e3e932e0cb1234c45' [12:49] <cone-265> ffmpeg.git 03Justin Ruggles 0756a9d2b44e97: atrac3: move the 'version' field from ATRAC3Context to where it is used [12:49] <cone-265> ffmpeg.git 03Justin Ruggles 0764ebbb8f89d1: atrac3: move the 'delay' field from ATRAC3Context to where it is used [12:49] <cone-265> ffmpeg.git 03Justin Ruggles 07cdd0e0de816d: atrac3: use AVCodecContext.block_align instead of keeping a private copy [12:49] <cone-265> ffmpeg.git 03Justin Ruggles 077c1f93afe6b0: atrac3: remove unused ATRAC3Context field, samples_per_channel [12:49] <cone-265> ffmpeg.git 03Justin Ruggles 07a2664c91fba1: atrac3: move the 'samples_per_frame' field from ATRAC3Context to where it is used [12:49] <cone-265> ffmpeg.git 03Michael Niedermayer 078748472f3996: Merge commit 'a2664c91fba15a1307f676ffad511f8f86fb3a27' [12:58] <cone-265> ffmpeg.git 03Justin Ruggles 077e76f270819e: atrac3: remove unused ATRAC3Context field, bit_rate [12:58] <cone-265> ffmpeg.git 03Justin Ruggles 07c51311b907e0: atrac3: move the 'frame_factor' field from ATRAC3Context to where it is used [12:58] <cone-265> ffmpeg.git 03Justin Ruggles 0778edce3f1973: atrac3: separate window initialization from IMDCT initialization [12:58] <cone-265> ffmpeg.git 03Justin Ruggles 075d1007f74dd4: atrac3: initialize static tables in AVCodec.init_static_data() [12:58] <cone-265> ffmpeg.git 03Michael Niedermayer 077bc6631dd027: Merge commit '5d1007f74dd496d54b932242004382f44e3b22b4' [13:12] <cone-265> ffmpeg.git 03Justin Ruggles 07327747de15be: atrac3: simplify MDCT window calculation [13:12] <cone-265> ffmpeg.git 03Justin Ruggles 0789a6c32bc1bc: atrac3: use sizeof(variable) instead of sizeof(type) [13:12] <cone-265> ffmpeg.git 03Justin Ruggles 07808686fc1ea6: atrac3: remove unused ATRAC3Context field, sample_rate [13:12] <cone-265> ffmpeg.git 03Justin Ruggles 07a1f4cd371ac7: atrac3: replace a calculation with FFALIGN() [13:12] <cone-265> ffmpeg.git 03Anton Khirnov 074e61a38aa038: avconv: only apply presets when we have an encoder. [13:12] <cone-265> ffmpeg.git 03Anton Khirnov 079cac8a519980: APIchanges: update lavr bump date [13:12] <cone-265> ffmpeg.git 03Anton Khirnov 072b8dd371e4d2: lavu: postpone recent deprecations until the next major bump [13:12] <cone-265> ffmpeg.git 03Michael Niedermayer 07af3fe43d841c: Merge commit '2b8dd371e4d276ca0d342e82b8b4cc281be0630a' [13:26] <cone-265> ffmpeg.git 03Anton Khirnov 07add3a6902ea4: lavu: bump major to 52 [13:26] <cone-265> ffmpeg.git 03Michael Niedermayer 07c2a25e141e01: Merge commit 'add3a6902ea4d459f36655656a1f8daa409118a7' [13:46] <cone-265> ffmpeg.git 03Anton Khirnov 07c6b25d1d5e6a: Revert "avutil: Add a copy of ff_sqrt_tab back into avutil to restore ABI compatibility" [13:46] <cone-265> ffmpeg.git 03Anton Khirnov 07468ea9d5b14f: Revert "avutil: make some tables visible again" [13:46] <cone-265> ffmpeg.git 03Michael Niedermayer 07a65bc2712db2: Merge commit '468ea9d5b14f92fe61f47f034e67066f65163f5f' [14:56] <cone-265> ffmpeg.git 03Anton Khirnov 078207c67a2417: lavu: remove disabled FF_API_GET_BITS_PER_SAMPLE_FMT cruft [14:56] <cone-265> ffmpeg.git 03Anton Khirnov 07c692957c4edd: lavu: remove disabled avoptions cruft [14:56] <cone-265> ffmpeg.git 03Michael Niedermayer 07d0ac26a264e5: Merge commit 'c692957c4edd119920cd7f2121628dd183589e6a' [15:04] <cone-265> ffmpeg.git 03Anton Khirnov 0700ed7f5958d0: lavu: remove disabled FF_API_AV_FIFO_PEEK cruft [15:04] <cone-265> ffmpeg.git 03Diego Biurrun 078f4c414df6b8: Improve wording and spelling of av_log_missing_feature messages. [15:04] <cone-265> ffmpeg.git 03Diego Biurrun 07e273a1a54291: fate: Add dependencies for WMA and WavPack tests [15:04] <cone-265> ffmpeg.git 03Diego Biurrun 078b8899ac3233: fate: Declare avcodec/avformat deps in the respective Makefile snippets [15:04] <cone-265> ffmpeg.git 03Michael Niedermayer 0735ce42e07029: Merge commit '8b8899ac3233b4f7af83ded0dc032fad8902d714' [15:12] <cone-265> ffmpeg.git 03Diego Biurrun 077d420a4294dc: fate: More fine-grained dependencies for libavcodec test programs [15:12] <cone-265> ffmpeg.git 03Diego Biurrun 0771f3eb4536d9: fate: cosmetics: Group idct8x8 test together with all other libavcodec tests [15:12] <cone-265> ffmpeg.git 03Diego Biurrun 0788bdec03b986: fate: Add dependencies for audio tests [15:12] <cone-265> ffmpeg.git 03Diego Biurrun 072f17f265ed79: mpegaudio: Refactor mp3on4 flush function [15:12] <cone-265> ffmpeg.git 03Michael Niedermayer 07ec444c84cfd3: mpegaudiodec: Fix buffer handling on random access [15:12] <cone-265> ffmpeg.git 03Michael Niedermayer 0779ec524ff52a: Merge commit 'ec444c84cfd30ef34a4e7b4ec7ee77d7e5250d25' [15:15] <cone-265> ffmpeg.git 03Mans Rullgard 076f9018fc5ecd: fate: fix ENCMUX macro [15:15] <cone-265> ffmpeg.git 03Mans Rullgard 07a1f6ad69c779: fate: check that dependencies actually exist [15:16] <cone-265> ffmpeg.git 03Michael Niedermayer 074d54ecf4b8e8: Merge remote-tracking branch 'qatar/master' [15:16] <ubitux> @_@ [15:18] <durandal_1707> that happens when stuff are not written right from the start [15:51] <cone-265> ffmpeg.git 03Peter Ross 078469fc17e289: wtvdec: display warning for encrypted wtv files (instead of 'unsupported chunk' messages) [16:18] <cone-265> ffmpeg.git 03Michael Niedermayer 07d8f27eceaabf: asfdec: ignore stored duration for truncated files [17:32] <cone-265> ffmpeg.git 03Michael Niedermayer 0731cdf6002ae5: lavf: increase retries to guess duration from pts by 1 [18:19] <cone-265> ffmpeg.git 03Michael Niedermayer 075442c6cce33d: lavf: fix duration estimation for multi program TS [19:28] <cone-265> ffmpeg.git 03Paul B Mahol 07e2820d99f25d: pnmdec: use more meaningful error codes [20:54] <kolbyjack> Hi, I'm trying to use libavformat to write an mp4 file, and vlc will only play the first second with its default demuxer. If I reprocess the file with ffmpeg -i myfile.mp4 -vcodec copy ffmpeg.mp4, it plays fine. Is there an easy way I can track down what I'm doing differently from ffmpeg? I'm finding it kind of difficult to follow. [21:04] <funman> kolbyjack: you can try vlc --demux avformat [21:04] <kolbyjack> Yeah, that plays fine, but I need to play it on an ipad, unfortunately. [21:05] <kolbyjack> I just need to figure out what I'm doing wrong when I'm muxing, since the video is apparently okay [21:49] <cone-265> ffmpeg.git 03Isaac Dooley 07469a65b15025: x11: Fix x11grab BadCursor [21:54] <thegeek> I'm getting a crash in swresample_0!ff_mix_2_1_a_int16_sse2 in my mingw-compiled ffmpeg 1.0 (x64), any tips for things to double-check in the toolchain before I start debugging etc? [21:55] <thegeek> I'm using mingw-w64 4.7.2 and I can not reproduce it with zeranoe's builds, so it's probably something I'm doing wrong, but it's wierd because a lot of other stuff works fine;P [21:58] <thegeek> aand if I disable sse/sse2 it happens in the mmx version of the function [21:58] <michaelni> that excludes SSE alignemnt issues [21:58] <michaelni> what happens without asm ? [21:59] <thegeek> hang on [21:59] <iive> i hit an alignment bug today, in intrinsic code inside mesa3d v9.0. using gcc-4.7.1 [22:01] <thegeek> I was using mingw-w64 4.7.0 and also some older deps of gcc, updating to the latest did not make a difference [22:07] <thegeek> michaelni: disabling asm fixed it [22:09] <thegeek> I'll do some more research on it and support a bug report [22:10] <thegeek> as far as I can tell I trigger it by using the concat filter two join two video files [22:11] <iive> thegeek: is there valgrind (like) program for windows? [22:11] <thegeek> I have used some in the past [22:11] <thegeek> though I can't remember the names of any of them;P [22:11] <thegeek> the big problem though is the lack of "windows" debug symbols [22:11] <thegeek> makes it hard to use any windows tools [22:12] <thegeek> I'll use gdb etc tomorrow [22:14] <cone-265> ffmpeg.git 03Stefano Sabatini 07c7065f1f8953: lavfi/sendcmd: release file in case of failed allocation [22:14] <cone-265> ffmpeg.git 03Stefano Sabatini 07bd6240e7e9f3: lavfi/sendcmd: move buf init() variable to internal scope where it is used [22:14] <cone-265> ffmpeg.git 03Stefano Sabatini 078b03cd3cd7a3: lavd/lavfi: add graph_file option [22:15] <thegeek> and hey, it seems you can use valgrind on windows as well through mingw [22:54] <cone-265> ffmpeg.git 03Carl Eugen Hoyos 072fdc6f355cfc: Do not detect mov with maximum score if the atom size is too small. [23:25] <saste> michaelni, any: ping on lavc/utils: propagate or return meaningful error codes in avcodec_open2() [23:54] <llogan> saste: "enabled, add" is a comma splice. I think. [23:55] <saste> llogan: i like comma splices [23:56] <llogan> everybody likes comma splices. [23:57] <saste> everybody likes comma splices, they give spice to boring single sentences. [23:58] <llogan> He who controls the Splice, controls the universe! [23:59] <llogan> why did my server turn off by itself? [23:59] <saste> he's tired [23:59] <saste> to serve [00:00] --- Wed Oct 24 2012

1 0

ffmpeg.log.20121023
by burek 24 Oct '12

24 Oct '12

[00:45] <njbair_> what's the "standard" WxH for mobile video? [01:41] <mooglenorph> Hi, where can I find the docs for the drawbox filter? [01:42] <mooglenorph> Alternately, is there an easier way to accomplish drawing a constant-position black box in my video? [01:42] <mooglenorph> I have where it should be in pixel offsets and it doesn't change all video file long. [01:44] <saste> mooglenorph, http://ffmpeg.org/ffmpeg.html#drawbox [01:44] <saste> or man ffmpeg [01:46] <mooglenorph> and would I specify it with -filter drawbox=x:y:width:height:color ? [01:46] <durandal_1707> P3 ppm is supported (decoding) [01:57] <saste> durandal_1707, good [01:57] <saste> are you planning to add encoding support, or should i add it to my todo? [01:57] <saste> mooglenorph, -vf drawbox=... [01:58] <durandal_1707> saste: no, you have nothing better to do? [01:59] <saste> durandal_1707, so many things [03:44] <kode54> damn [03:44] <kode54> is 2.4Mbps that low for 720x480 at 29.97 fps? [03:44] <kode54> (MJPEG) [03:48] <Exatrive> I think so.. just a guess tho. [03:55] <kode54> that's what I get for trusting ffdshow with 1-pass quality based encoding [03:55] <kode54> I guess the default of 85 sucks [03:56] <Exatrive> 85 is pretty low in jpeg when it come to motion jpeg [03:57] <Exatrive> from my experience it take about 93 or more to get it less artifact in the motion [03:57] <kode54> next time I'll use 100, since space isn't an issue [03:57] <Exatrive> but that was with my experienc in virtualdub [03:58] <Exatrive> yeah if you can get it to do 100% that would be best [03:58] <kode54> must shoot somebody for making 85 the default [03:59] <Exatrive> even at 100% tho I'm not sure if that maximazes subsampling quality [04:00] <kode54> I'm capturing from composite NTSC [04:00] <kode54> in not so ideal conditions [04:00] <Exatrive> yeah [04:00] <Exatrive> ? [04:01] <kode54> I'm using a USB powered capture device that only has drivers for Windows XP, so I'm running the capture software inside of a virtual machine and feeding it the device through USB passthrough [04:01] <Exatrive> I would try to capture to lossless format then till your final edit is ready to be reencoded [04:01] <kode54> assuming that it won't munch the disk space [04:01] <Exatrive> really? [04:01] <kode54> 32GB left on that virtual drive [04:01] <Exatrive> what capture device is that? [04:02] <kode54> Adaptec Gamebridge [04:02] <kode54> last I checked, the linux developers who even remotely approach work on the chipset it's based on are uninterested in it because it's a USB powered low latency device, so apparently not designed for serious video capture [04:04] <Exatrive> yeah I just had a google and linux support is none exsistanct.. and its a pretty old device.. [04:04] <Exatrive> ebay is full of USB devices tho that would work in linux [04:05] <Exatrive> but unless your archiving a bunch of vhs or something .. prolly not worth it. [04:05] <kode54> it's only interesting for the occasion when I want to pull videos off my DVR box [04:06] <kode54> hurray for proprietary black box video recorders [04:06] <Exatrive> lol [04:06] <kode54> it even encrypts the videos to the user's account so they can't be replayed when they refurbish the box [04:08] <kode54> and they're even clumsy enough to not wipe the drives when they refurbish, so you end up with a machine with hours of random shows you can only view the titles and information blocks for [04:08] <kode54> which you then have to erase one by one [04:09] <kode54> the only way to mass erase videos is to load up their web interface on a desktop pc (which goes out through the public internet) and check all the videos and click delete [04:10] <Exatrive> what version is your gamebridge? [04:10] <kode54> revision a [04:10] <Exatrive> model/ [04:10] <Exatrive> ? [04:10] <kode54> avc-1400/1500 [04:11] <kode54> yes, the label says that [04:11] <kode54> pn asm-0438-0a-a [04:11] <kode54> *00438 [04:11] <kode54> *01-a [04:11] <kode54> damn, I screwed that up badly [04:11] <kode54> asm-00438-01-a [04:11] <Exatrive> and you have had no luck getting it to install? [04:12] <kode54> only installs in Windows XP, or from what I've heard, Vista/7 32 bit [04:12] <kode54> it has no 64 bit drivers [04:12] <kode54> in fact, Adaptec doesn't even provide a driver download, the only way to get the drivers is from the bundled CD [04:13] <kode54> another wonderful piece of proprietary hardware [04:13] <kode54> seems to capture nicely [04:14] <kode54> if only I had used huffyuv or similar [04:15] <Exatrive> lol [04:17] <Exatrive> I capture to lossless MPEG 4 but... almost no editor will even open the .mp4 without crashing hard... [04:18] <Exatrive> if I capture the same in ffdshow as AVI using similar lossless functions tho.. other programs seem to open it just fine.. not sure what all ffmpeg does in the stream to not allow editors to open it. [04:20] <Exatrive> as for that device.. yeah not much in the way of x64bit support at all.. [05:07] <svm_invictvs> Heya [05:08] <svm_invictvs> I was curious, given a raw byte stream, if it was possible to use ffmpeg (or one of its related libraries) to read the SEI data from a raw byte stream, without actually decoding the whole frame. [05:08] <svm_invictvs> Is that possible, or am I barking up the wrong tree? [05:47] Action: Exatrive ** WinSys ** Client: HexChat 2.9.3 (x64) ** OS: Microsoft Windows 7 Ultimate ** CPU: Intel(R) Core(TM) i5-2500K CPU @ 3.30GHz (3.00 GHz) ** RAM: 16364 MB Total (4863 MB Free) ** VGA: NVIDIA GeForce GTX 570 ** Uptime: 279.45 Hours ** [05:47] <Exatrive> lol opps [06:33] <magn3ts> ergh, wtf do I have to do to get print_format to work? [06:33] <magn3ts> I'm building ffmpeg from source, surely it would be in there, no? It looks like it was added back in January [06:35] <magn3ts> ffmpeg version git-2012-10-23 and yet I get "unrecognized option 'print_format'" [06:36] <magn3ts> oh jesus [07:23] <ubitux> magn3ts: this is a ffprobe option [07:29] <magn3ts> ubitux: yeah, my idiot self found that out about an hour ago, my apologies. [07:29] <ubitux> no worry :p [08:46] <elkng> I can use "ffmpeg -ss 0:0:33 -t 12 -i file.avi -c:v png %d.png" to get frames in png format of part of video from 33 to 45 second how can I get exact frames, if I want to get frames only from 400 to 500 ? [09:16] <stclaws> Ah, again: Trying to compile ffmpeg with --enable-librtmp and getting that dreaded ERROR: librtmp not found when doing configure. I have pkg-config installed and PKG_CONFIG_PATH set to just about anything I can think of, but still. Any clue anyone? [09:16] <elkng> stclaws: do you have librtmp installed ? [09:18] <stclaws> yes, just installed rtmpdump, and it is there: /usr/local/include/librtmp [09:18] <stclaws> which is a dir containing amf.h http.h log.h rtmp.h [09:19] <elkng> try to reboot [09:19] <stclaws> I ran pkg-config librtmp librtmp/rtmp.h RTMP_Socket directly and no complaints [09:20] <elkng> "is it possible to use ffmpeg as a video editing library to embed its functionality in other programs, like if I want to get 500th frame and I could use something like ffmpeg_get_frame("/tmp/file.avi", 500, "png"); and it will return pointer to png file structure is it possible to use ffmpeg or one of its libraries this way ? [09:21] <stclaws> elkng: Reboot? Why would that help? [09:24] <elkng> if librtmp was freshly installed [09:25] <stclaws> Is that ever needed on a LInux system? [09:30] <stclaws> Did anyone here install ffmpeg with rtmp support on a Centos system? [09:30] <stclaws> or something like that [09:32] <ubitux> "try to reboot" wtf :) [09:35] <stclaws> are u on Windows, elkng? :) [09:37] <stclaws> pkg-config --list-all | grep rtmp [09:37] <stclaws> librtmp librtmp - RTMP implementation [09:38] <stclaws> How come then that the log says: check_pkg_config librtmp librtmp/rtmp.h RTMP_Socket ERROR: librtmp not found [09:39] <stclaws> I am mystified [09:51] <elkng> stclaws: was librtmp freshly installed right before compilling ffmpeg ? what the packet manager ? [10:05] <stclaws> elkng: compiled from source right before compiling ffmpeg [10:06] <elkng> try to reboot [10:51] <Exatrive> Anyone who has played with MP4 and sony vegas know what settings my be causing vegas to not be able to render the video stream, but only the audio? I have tried to reporduce the effect in handbreak with verious settings and the only thing that kinda looks like it might effect sony vegas is Subpixle me and Mode decision. cause with certain settings vegas will still render the video but it will be garbage. [10:52] <Exatrive> in regards to mp4 this is something straight from ffmpeg and having to reincode in handbreak before sony will not barf on it. [10:52] <Exatrive> in power director another editor it failes to open it also [10:53] <Exatrive> I have even tried Avidemux 2.6 but that hardcrashes even attempting to open the video. [10:54] <Exatrive> but.. MPC and VLC can play it fine.. just can't bring into any editor. [11:37] <brontosaurusrex> Exatrive: convert to some fat I frame format, then edit < perhaps [11:37] <brontosaurusrex> prores or something [11:37] <brontosaurusrex> ffmpeg should do it [14:58] <Aziroshin> Hello. [15:00] <Aziroshin> I am trying to remux content from mk4 to mp4 (something I unfortunately have to do, as it seems). Command: ffmpeg -i input.mkv -acodec copy -vcodec copy output.mp4, Error: track 1: could not find tag, codec not currently supported in container, Full log: http://pastebin.com/ahYSpdwX [15:01] <cbsrobot> Aziroshin: you cannot put flac into mp4 [15:01] <ubitux> [mp4 @ 0x63d7b0] track 1: could not find tag, codec not currently supported in container [15:01] <cbsrobot> you need to transcode it to aac f.ex. [15:01] <cbsrobot> hi ubitux [15:01] <ubitux> hey cbsrobot :) [15:01] <Aziroshin> I should have read the log better. [15:01] <Aziroshin> Thanks. [15:01] <ubitux> no lossless codec possible in mp4? [15:02] <cbsrobot> not that I know of [15:02] <ubitux> mmh we lack an ALS encoder it seems [15:02] <divVerent> 15:01:50 @ubitux | no lossless codec possible in mp4? [15:02] <divVerent> 15:02:03 cbsrobot | not that I know of [15:02] <divVerent> apple lossless? [15:02] <divVerent> ah, you meant that [15:02] <divVerent> isn't it normally called ALAC? [15:02] <ubitux> mpeg-4 als is something else afaik [15:03] <ubitux> maybe you can put some alac in mp4 indeed [15:03] <Aziroshin> I hope gentoo offers a USE flag to make ffmpeg disable the --strict stuff by default. [15:03] <ubitux> (or maybe just in the mov variant) [15:03] <divVerent> 15:03:02 @ubitux | maybe you can put some alac in mp4 indeed [15:03] <ubitux> Aziroshin: i'm not sure you can remove this at built time [15:03] <divVerent> I am pretty sure you can [15:03] <divVerent> given ALAC audio files are called .m4a [15:03] <divVerent> and use MP4 containers [15:04] <Aziroshin> It's transcoding now. :) [15:04] <divVerent> 15:03:09 Aziroshin | I hope gentoo offers a USE flag to make ffmpeg disable the --strict stuff by default. [15:04] <divVerent> Aziroshin: why? [15:04] <divVerent> alias ffmpeg="ffmpeg -strict -2" [15:04] <divVerent> ;) [15:04] <divVerent> does such stuff really HAVE to be compile time? [15:04] <Aziroshin> divVerent: Or that. XD [15:05] <ubitux> divVerent: it seems i can with mp4 [15:05] <ubitux> but i can with mov [15:05] <divVerent> you can but you can? [15:05] <ubitux> can't with mp4 sorry [15:05] <ubitux> ./ffmpeg -f lavfi -i 'testsrc[out0]; aevalsrc=sin(2*PI*t*440)[out1]' -map 0 -t 5 -c:a alac out.mp4 no ok [15:05] <ubitux> ./ffmpeg -f lavfi -i 'testsrc[out0]; aevalsrc=sin(2*PI*t*440)[out1]' -map 0 -t 5 -c:a alac out.mov ok [15:05] <divVerent> ah, then ffmpeg probably lacks it :P [15:05] <divVerent> because, Apple does produce such m4a files [15:05] <divVerent> on iTunes e.g. [15:05] <ubitux> are you sure they haven't not the mov signature? [15:05] <Aziroshin> Do you think transcoding from flac to aac will cause recognizable quality loss? [15:06] <divVerent> I can check [15:06] <ubitux> i've a problem with negation today. [15:06] <divVerent> don't have any here, though.. damn [15:06] <divVerent> maybe on the server... [15:06] <Aziroshin> Because if that's the case, I'll encode from the original wav file and mux the streams together from scratch. [15:07] <divVerent> how do I see what signature it has? [15:07] <divVerent> compatible_brands: M4A mp42isom [15:07] <divVerent> is this enough? [15:07] <ubitux> hexdump the top of the file [15:07] <divVerent> 0000000: 0000 0020 6674 7970 4d34 4120 0000 0000 ... ftypM4A .... [15:07] <divVerent> 0000010: 4d34 4120 6d70 3432 6973 6f6d 0000 0000 M4A mp42isom.... [15:07] <divVerent> 0000020: 0009 e2c0 6d6f 6f76 0000 006c 6d76 6864 ....moov...lmvhd [15:07] <divVerent> 0000030: 0000 0000 c8b4 3d9a c8b4 43bd 0000 ac44 ......=...C....D [15:07] <ubitux> ftypM4A ok [15:08] <ubitux> "m4a " is audio only [15:08] <divVerent> well, it is an audio only file [15:08] <ubitux> but well, mp42 isom is in as well.. [15:09] <divVerent> wonder if MP4Box will allow muxing it with video... [15:10] <divVerent> hm... [15:10] <divVerent> IsoMedia import - skipping track ID 1 (unknown type 'alac') [15:10] <divVerent> that is a problem [15:16] <divVerent> ubitux: we don't need no non-negations [15:16] <ubitux> :D [15:16] <divVerent> we ain't not need no non-negations [15:17] <divVerent> ain't is my favorite, as it may or may not be negative :P [15:17] <Aziroshin> Somehow, youtube doesn't eat my mp4 file. [15:17] <Aziroshin> I transmuxed it from mkv to mp4, and it still doesn't eat it. [15:18] <Aziroshin> It's h264 and aac. [15:19] <divVerent> ubitux: is there a "secret hack" to enable a buffer for seeking in ffplay? [15:19] <divVerent> like what mplayer does when streaming from network [15:19] <divVerent> it then keeps some stuff buffered so you can seek back for about 20 sec [15:20] <ubitux> no idea [15:20] <divVerent> asking, as I wanted to use a complex filter (multiple sources) with ffplay [15:20] <divVerent> but that seems to simply not be possible [15:20] <Aziroshin> When I do ffmpeg -i <file> and it shows me: "Aziroshin's Mhykol Mines "Temp" Server Diary, Episode V: The Woodhenge.mp4: Invalid data found when processing input", something's odd, yes? [15:20] <divVerent> so I wanted to do the filtering in ffmpeg, and ffplay from stdin [15:21] <divVerent> ffplay only connects [in] and [out] [15:22] <divVerent> alternative would be using the movie= source [15:23] <divVerent> catch with that source is that it sucks :P [15:23] <divVerent> seriously, how does one even specify a file name with it if it contains evil characters [15:30] <dTal> divVerent: you escape it [15:31] <dTal> put a backslash in front of every asphanumeric character [15:31] <dTal> *non-alphanumeric [15:34] <dTal> that's bash by the way not ffmpeg [15:43] <thegeek> I'm getting sse2 crashes in libresample in my own mingw build of ffmpeg 1.0 but I can not reproduce in zeranoe's builds; has anyone seen similar problems? [15:44] <thegeek> I noticed that I used gcc 4.7.0 while zeranoe use 4.7.1 so I'm updating my gcc [15:44] <thegeek> anything else I should try? [15:46] <divVerent> dTal: ah, ok [15:46] <divVerent> but that's useless, can't seek when using the lavfi source [17:50] <hendry> is there a more efficient codec than ffvhuff for creating a "lossless raw" ? [17:50] <durandal_1707> efficient in what sense? [17:51] <Tjoppen> ffv1 [17:52] <hendry> durandal_1707: hd space? i only have a 2G /tmp [17:52] <hendry> and it fills up in 20 seconds or so [17:59] <RunawayDevil> My God, why is so fucking hard to make a damn dvd on mac [17:59] <RunawayDevil> whyyyy whyyyyyyyyyyyyyyyyyy [18:01] <Tjoppen> hendry: HDDs are $.05 c/GB.. [18:01] <Tjoppen> -c [18:06] <RunawayDevil> ahhh [18:06] <Tjoppen> or you a lossy codec at high enough quality [18:07] <RunawayDevil> why when I convert anyfile to .vob, using mpeg2video codec in extremaly high definition is fast and when i do the same to .mp4 using h.264 it takes almost 10 hours? [18:07] <RunawayDevil> someone can explain me this? [18:08] <Tjoppen> because mpeg-2 is a simpler codec [18:08] <Tjoppen> you can tell x264 to be faster [18:08] <RunawayDevil> but there is too much difference in the gain of quality that justify my loss of time making a h.264? [18:09] <Tjoppen> try one of the faster x264 presets and judge for yourself [18:09] <RunawayDevil> i wanna make a simple dvd with good resolution to watch [18:09] <RunawayDevil> tell me your opinion please [18:09] <Tjoppen> DVD = MPEG-2 [18:09] <RunawayDevil> hum [18:09] <RunawayDevil> h.264 = blu ray [18:09] <Tjoppen> yes [18:10] <hendry> hmmm. new problem: x264 [error]: baseline profile doesn't support 4:4:4 [18:10] <durandal_1707> it is self explanatory [18:11] <RunawayDevil> Tjoppen dvd hd uses which codec? [18:11] <Tjoppen> hendry: -pix_fmt yuv422p [18:11] <RunawayDevil> which what ahhh im losting my mind lol [18:13] <RunawayDevil> I had some bad issues with conversion and burning in macOS, the first one was with the subtitles. I used iFFmepg instead ffmpegx because I could change the size of the fonts and stuff like that... I read in some forums maybe teh problem is with mencoder [18:15] <RunawayDevil> well, I'm very fedup, now using iFFmpeg (that i bought) I still can change the size of subtitle without shit in the grammar and accents, the app works but not so good too, when I convert to .vob the app automaticaly convert my .srt to .ass and thats the problem... after this better do not edit, or it will be a shit! [18:18] <thegeek> Is it possible to get a "crossfade" effect when joining two video clips ? (I'm using the concat video filter atm) [18:18] <RunawayDevil> Tjoppen I will burn a DVD DL, suggest me the codec to make some DVD HD and fill this dvd, pls [18:22] <Tjoppen> "To record moving pictures, DVD-Video uses either MPEG-2 compression at up to 9.8 Mbit/s (9,800 kbit/s) or MPEG-1 compression at up to 1.856 Mbit/s (1,856 kbit/s)." [18:23] <Tjoppen> I don't see how you have any choice (unless you want to use uber-crap mpeg-1) [18:24] <hendry> why is mp4 so much smaller than webm? e.g. http://r2d2.webconverger.org/2012-10-24/ [18:24] <hendry> also why is mp4 preferred by chrome over webm... #confused [18:29] <Tjoppen> that depends on your encoder settings [18:29] <Tjoppen> (hashtags don't work on IRC) [18:30] <hendry> Tjoppen: do these settings look OK? https://github.com/kaihendry/recordmydesktop2.0/blob/master/r2d2.sh#L46 [18:31] <Tjoppen> the webm file has crappy I-frames too [18:31] <Tjoppen> anyway, libx264 is still a much better encoder than libvpx [18:32] <Tjoppen> try using -crf with libx264 if you can [18:33] <hendry> Tjoppen: why? and which value should I use? [18:33] <RunawayDevil> "ScriptType: v4.00+" this is fucking me? what is this? is this something from libass? [18:33] <hendry> ffmpeg manpage sez crf "VPX_CQ", "VP8E_SET_CQ_LEVEL" [18:33] <Tjoppen> why? because [18:34] <Tjoppen> vpx has -crf too now? hum [18:34] <Tjoppen> I usually use something between 20 and 24 [18:34] <Tjoppen> lower is better [18:34] <hendry> i'm pretty happy with mp4 quality and size [18:35] <Tjoppen> fair enough. I suppose give -crf 24 a try with libvpx and see what happens? I have no idea [18:36] <RunawayDevil> "ScriptType: v4.00+" this is fucking me? what is this? is this something from libass? [18:41] Action: hendry wonders why the ffmpeg manpage only refers to -crf in context of VP8 [18:45] <hendry> Tjoppen: -crf 24 on the webm line has made the end file size closer to the mp4! [18:49] <cbsrobot> RunawayDevil: what are you ranting ? [18:49] <cbsrobot> what do you want to do ? [18:50] <cbsrobot> (please the simple answer) [18:50] <RunawayDevil> cbsrobot I want to burn a DVD with subtitles bigger because the default I can't read [18:50] <cbsrobot> hardsubs ? [18:50] <RunawayDevil> yes [18:51] <RunawayDevil> because soft is not supported in .vob [18:51] <cbsrobot> so you have a srt file ? [18:51] <RunawayDevil> yes [18:51] <cbsrobot> so: [18:51] <cbsrobot> ffmpeg -i file.srt file.ass [18:51] <cbsrobot> and then [18:52] <cbsrobot> what tool do you use for authoring ? [18:52] <RunawayDevil> iFFmpeg [18:52] <RunawayDevil> it converts automaticaly the .srt to .ass [18:52] <cbsrobot> nah [18:52] <cbsrobot> dvd authoring ? [18:53] <RunawayDevil> iSkysoft iMedia [18:53] <RunawayDevil> I noob, kind lost lol [18:54] <cbsrobot> what tool do you use to create the dvd structure you burn to a dvd ? [18:54] <cbsrobot> DVD Studio Pro ? [18:54] <cbsrobot> dvd-author ? [18:54] <cbsrobot> iDVD ? [18:54] <Tjoppen> hendry: woo [18:55] <RunawayDevil> cbsrobot: http://www.iskysoft.com/imedia-converter-deluxe-mac.html [18:56] <cbsrobot> ah sorry [18:56] <cbsrobot> I just read ripping [18:56] <RunawayDevil> cbsrobot: before I used the iFFmpeg I convert to .vob and so .iso and so burn [18:57] <cbsrobot> what is your input file ? [18:57] <cbsrobot> vob ? [18:58] <RunawayDevil> cbsrobot: yes, if you can suggest me something better [18:58] <RunawayDevil> anyway keep telling [18:58] <cbsrobot> nah - take the native one [18:58] <cbsrobot> so now with the ass filei nplace [18:58] <cbsrobot> can you try: [18:59] <RunawayDevil> hum [18:59] <cbsrobot> ffmpeg -i input.vob -vf ass=sub.ass -c:v mpeg2video -sameq -c:a copy output.vob [19:01] Action: cbsrobot points JEEB to a (maybe) sane usage of sameq &. ^ [19:01] <relaxed> it doesn't work [19:01] <RunawayDevil> cbsrobot what does it do? [19:01] <cbsrobot> relaxed: what doesnt work ? [19:01] <relaxed> -sameq [19:02] <durandal_1707> sameq does nothing with latest version [19:02] <cbsrobot> in his version of ffmpeg maybe it still does [19:02] <relaxed> it was completely removed from libav for that very reason. [19:02] <durandal_1707> cbsrobot: sameq is broken several years [19:02] <cbsrobot> and its from mpeg2 -> mpeg2 [19:02] <cbsrobot> so it should work [19:02] <cbsrobot> well RunawayDevil then try: [19:03] <durandal_1707> just set very small quant to keep some quality [19:03] <Ganymede> General question about video (not ffmpeg-specific), does anyone know a tool to graphical tool to losslessly cut H.264 MP4s? mkvmerge does it (into MKV, which is fine) but is not graphical. I tried using avidemux2 to get the timestamps visually and then use mkvmerge but the timestamps from avidemux2 appear to be inaccurate for H.264 (the reported timestamp is x seconds slow for every minute of video and I [19:03] <Ganymede> don't know the exact formula). [19:04] <relaxed> MP4Box [19:04] <RunawayDevil> just a second [19:04] <cbsrobot> ffmpeg -i input.vob -vf ass=sub.ass -target pal-dvd -quant 3 output.vob [19:04] <relaxed> Ganymede: oh, with a gui I don't know. [19:06] <Ganymede> Or the GUI tool can be separate from the actual cutting tool, I just need accurate timestamps. My current process is to do a first draft cut using avidemux2's timestamps and then watch the resulting videos and get an offset for each and cut again. [19:06] <Ganymede> Very prone to error. [19:06] <relaxed> I believe you're only going to be able to cut at keyframes. [19:07] <Ganymede> That's fine. But when the video lags by 40 seconds after an hour, finding the timestamp for the keyframe is still hard (and mkvmerge as far as I can tell, only accepts timestamps for cutting, not frame numbers). [19:08] <relaxed> You could encode using a lossless, intra only codec [19:09] <RunawayDevil> cbsrobot what size of subtitle do you consider the best one? [19:10] <relaxed> That would allow frame accurate cutting using a gui. (if it supports the lossless codec) [19:10] <RunawayDevil> or anyone else if can tell me [19:13] <Ganymede> Maybe if I encode the H.264 to DivX or anything, I can get avidemux2 to give me accurate timestamps... (I had a feeling the issue was variable frame rate in H.264? Does that exist?) [19:17] <brontosaurusrex> Ganymede: it does, handbrake is happy defaulting to that i think [19:27] <Ganymede> Damn...various factors are preventing me from doing an intermediate encode...guess I'll be stuck with my old workflow. [20:27] <mbradshaw> Hi all, I have a question with using avfilter through the C API. I'm using avfilter to rescale video (and convert it to the right pixel format) and convert audio (to the right sample format, sample rate, and channel_layout) (I'm also buffering audio using asetnsamples to fill frames with a consistent number of samples) [20:29] <mbradshaw> If the source audio stream changes (i.e. sample rate, sample format, channel_layout), or if the source video stream changes (width, height, or pixel format), do I have to completely destroy and rebuild the filter graph? Or is there a way to "update" the source filters to the new formats and update (without destroying) the filter graph? [20:46] <llogan> mbradshaw: we don't get many API answers here. [20:49] <mbradshaw> llogan: Well, I guess I'll try the mailing list then. Thanks. [20:49] <cbsrobot> RunawayDevil: still lurking around ? [20:49] <cbsrobot> sorry I was afk [20:50] <cbsrobot> did you finish your dvd ? [20:51] <cbsrobot> RunawayDevil: for DVD I'd say 20px + for reading [23:09] <VooDooNOFX_> Is is possible to extract a sequence of frames from a prores QT and rewrap it into a new QT of the same params without any conversions using ffmpeg-0.[10|11]? [23:13] <Tjoppen> ffmpeg -i foo.mov -ss <start time> -t <duration> -vcodec copy bar.mov [23:13] <Tjoppen> possibly -i after -t [23:14] <VooDooNOFX_> of course, the copy vcodec. I'll give that a try. thanks [23:23] <buhman> so if one was to take some arbitrary audio samples, and encode them non-losslessly, take the result, and encode it again, I wonder how many recursions would be required to lose 100% fidelity. [23:26] <Tjoppen> that depends on the codec. you might just hit a fixed point [23:26] <Tjoppen> it'd be fairly easy to test. just write a small shell script that uses say libmp3lame to encode one file to 1.mp3, that to 2.mp3, to 3.mp3 etc [23:27] <Tjoppen> do the same with mp2, aac, vorbis and so on [23:43] <Endorgh> hi everyone [23:46] <Endorgh> Is there any way to feed Icecast server with ffmpeg? Or in other words, Is there any way to use ffmpeg to stream audio to icecast? [00:00] --- Wed Oct 24 2012

1 0

ffmpeg-devel.log.20121022
by burek 23 Oct '12

23 Oct '12

[00:12] <cone-618> ffmpeg.git 03Michael Niedermayer 07248b1ff26b7c: Fix various uses of av_log_missing_feature() * 03http://tinyurl.com/9ttzpsb03 [00:12] <cone-618> ffmpeg.git 03Michael Niedermayer 0739747d87d090: avfilter_graph_parse: add support for parsing sws_flags * 03http://tinyurl.com/8eevlbz03 [00:12] <cone-618> ffmpeg.git 03Michael Niedermayer 072d11ee4bfc7c: fate: fix fate-filter-metadata-scenedetect * 03http://tinyurl.com/9rorhfu03 [07:35] <ubitux> michaelni: thx for the fix! [09:08] <ubitux> michaelni: "fix used variable" // "unused", right? [10:07] <cone-571> ffmpeg.git 03Paul B Mahol 07cb0add3ce961: lavf/flacenc: disallow creation of invalid files with -c copy * 03http://tinyurl.com/9rwjvbk03 [10:42] <ubitux> saste: you forgot to remove gcd in cdea54b4c [10:42] <ubitux> btw, looking forward tools/ffescape :)) [10:43] <ubitux> oh and i sent a pending patch on the ml for the SAMPLES issue [10:44] <cjhmdm> hello, can I find libfdk devs here? [10:44] <ubitux> i don't think so [10:47] <saste> wbs^^ [10:48] <ubitux> ah, wbs indeed :) [10:50] <ubitux> btw, just curious, why working on an external encoder? [10:51] <cone-571> ffmpeg.git 03Stefano Sabatini 07b6e36a424447: lavu/parseutils: remove unused gcd variable in av_parse_ratio() * 03http://tinyurl.com/8cnkneh03 [10:51] <cone-571> ffmpeg.git 03Stefano Sabatini 07ccd6def9b3dc: lavfi/aspect: extend syntax for the setdar and setsar filters * 03http://tinyurl.com/8nserd403 [10:51] <cone-571> ffmpeg.git 03Stefano Sabatini 076752aac6bca7: lavfi/aspect: add max option * 03http://tinyurl.com/962my2b03 [10:52] <ubitux> thx saste [10:52] <cjhmdm> ubitux: were you asking me? [10:52] <ubitux> more wbs i believe :p [10:53] <cjhmdm> dunno what that is, but ok heh [10:53] <ubitux> a nickname on this channel :) [10:57] <saste> ubitux: what is $(FATE_SAMPLES_FFPROBE)? [10:59] <saste> tested the patch, and seems to work, but i'll leave review to someone more qualified than me [10:59] <saste> as for ffescape, i can hardly work on it before this night [11:00] <saste> given a filename, you need to perform this: [11:00] <ubitux> saste: FATE_SAMPLES_FFPROBE is a different list than FATE_FFPROBE, the same way FATE_SAMPLES_FFMPEG is a different list than FATE_FFMPEG [11:01] <saste> ubitux, ok bad naming but nothing we can do about [11:01] <ubitux> and these FATE_SAMPLES_<tool> list are in the FATE_EXTERN list, which is only used when SAMPLES is set [11:01] <ubitux> do you want me to add this in the comment description? [11:02] <ubitux> <@saste> given a filename, you need to perform this: // this? [11:05] <saste> shescape(ffescape(ffescape(filename, ":") + ":$other_options", ",;")) [11:06] <saste> you need to escape ":" in the filename, then special graph chars in the resulting string, and finally shell escape [11:06] <saste> we can improve this mess in two ways [11:07] <saste> 1. by putting the filtergraph description in a file [11:07] <saste> i proposed that sometimes ago and it was rejected with the argument that sh can do it [11:07] <saste> (include a file in the command) [11:07] <ubitux> for the fate failure i believe we will just need to ffescape ':' in $SRC [11:08] <saste> 2. by creating an alternative syntax (more avisynth like) [11:08] <ubitux> filtergraph in a description file agree. [11:08] <ubitux> -f lavfi script=myfiltergraph [11:08] <saste> 2. is much work, so we can ignore it for the moment [11:09] <saste> ubitux, yes that's a possibility [11:09] <ubitux> -f lavfi evalscript=myfiltergraph even [11:09] <saste> ubitux, looks, maybe it is *aready* implemented [11:09] <ubitux> so we can add a script= later with a scripting thing like you just mentioned [11:10] <saste> ummhno [11:10] <ubitux> haha [11:10] <saste> but is easy to add [11:10] <ubitux> no way it is @_@ [11:10] <ubitux> ah no, it's the "str" [11:10] <ubitux> not a file [11:10] <saste> check for example how I did it in sendcmd [11:11] <saste> then I wouldn't be against -{a,v}f_file -filter_complex_file [11:11] <saste> but it is not required here [11:12] <saste> gottago, good luck ;-) [11:12] <ubitux> heh [11:13] <ubitux> will see later [11:13] <ubitux> still, generating a file for this in fate looks overkill [11:13] <ubitux> i'd better just ffescape the ':' [12:35] <j-b> [mpeg2video @ 0177ab80] a vbv buffer size is needed, for encoding with a maximum bitrate [12:35] <j-b> any idea? [12:40] <Compn> -rc_max_vbv_use 1 [12:40] <Compn> j-b : according to this http://www.itbroadcastanddigitalcinema.com/ffmpeg_howto.html anyhow [12:41] <j-b> weird, this worked before [12:41] <michaelni> which commit broke it ? [12:42] <j-b> no clue [12:42] <j-b> like yesterday, users complaining [12:45] <michaelni> j-b, how can the problem be reproduced ? [12:46] <j-b> according to user: "C:\Program Files\VideoLAN\VLC\vlc.exe" -I dummy --dummy-quiet --ffmpeg-hurry-up --no-loop --no-repeat --ffmpeg-skiploopfilter 4 --audio-language=any --start-time 0 --sout "#transcode{venc=ffmpeg,vcodec=mp2v,vb=7000,acodec=a52,ab=384,samplerate=48000,deinterlace}:std{access=file,mux=ts,dst="test.mpg"}" --sout-transcode-width 480 --sout-transcode-height 320 --sout-ffmpeg-keyint=3 "C:\Media\sample.avi" "vlc://quit" [12:46] <j-b> let it drop [12:46] <j-b> I will find a fix [13:21] <cone-571> ffmpeg.git 03Michael Niedermayer 076182e0a6f640: vf_aspect: unbreak avoption system * 03http://tinyurl.com/9escc9y03 [13:21] <saste> damn [13:24] <Compn> j-b : vlc doesnt have an automated testing system like fate ? [13:24] <j-b> no [13:24] <saste> Failed to avformat_open_input ... <= what's this ffspeak? [13:24] <Compn> to detect regressions like this [13:31] <saste> j-b: you could call it "VATE" [13:31] <saste> http://www.urbandictionary.com/define.php?term=vate [13:33] <saste> in Italian/Latin it is a word (with a positive connotation, it is the attribution given to a high poet/priest) [13:38] <ubitux> it's hard to make test suites for video players [13:41] <av500> for jerkiness and tearing :) [13:41] <av500> and audio microcuts [13:44] <durandal_1707> ubitux: give job to several folks which will watch various clips all the time [13:45] <ubitux> it won't be called "VATE" then, it will be "CHINA" (i let you find an acronym for the joke) [14:07] <av500> durandal_1707: I watch clips all the time at work :) [14:07] <durandal_1707> av500: talk with j-b [14:08] <durandal_1707> av500: lucky you [14:08] <cone-571> ffmpeg.git 03Michael Niedermayer 07c3778df2d4c0: ffmpeg: fix negative array index * 03http://tinyurl.com/8sa9zdv03 [14:33] <Compn> ubitux : whys that? it has command line interface, and i'm guessing some kind of md5output [14:33] <Compn> if you mean , just for sheer amounts of tests, that would be hard to test every option on every build [14:34] <ubitux> Compn: you would be able to test decoders/demuxers [14:34] <ubitux> not what actually makes the video player [14:35] <ubitux> (the user experience, smooth playback, GUI interations etc) [14:35] <Compn> smooth playback by timing the test [14:38] <ubitux> i didn't say it was impossible, but it's generally way more complex than just diffing outputs [14:44] <michaelni> Compn, can you add regression tests to ffplay :) ? [14:44] <michaelni> i wanted these for a while already ... [14:45] <ubitux> -vo file! [14:49] <cone-571> ffmpeg.git 03Nathan Caldwell 07a4aa20fbdbe8: avcodec: prefer decoders without CODEC_CAP_EXPERIMENTAL * 03http://tinyurl.com/8n42naf03 [14:49] <cone-571> ffmpeg.git 03Nathan Caldwell 07a893655bdaa7: avutil: Add AVERROR_EXPERIMENTAL * 03http://tinyurl.com/8se4wgf03 [14:49] <cone-571> ffmpeg.git 03Nathan Caldwell 07c854102da773: avcodec: handle AVERROR_EXPERIMENTAL * 03http://tinyurl.com/8tnjhnu03 [14:50] <cone-571> ffmpeg.git 03Martin Storsjö 07e0d5ac6ae3a8: rtsp: Update a comment to the current filename scheme * 03http://tinyurl.com/96qyhga03 [14:50] <cone-571> ffmpeg.git 03Martin Storsjö 07c3e15f7b39aa: rtpdec: Don't pass a non-AVClass pointer as log context * 03http://tinyurl.com/9mlgx9803 [14:50] <cone-571> ffmpeg.git 03Michael Niedermayer 07e3a91c51f713: Merge commit 'c3e15f7b39aac2012f09ee4ca86d2bc674ffdbd4' * 03http://tinyurl.com/98d379403 [15:01] <cone-571> ffmpeg.git 03Justin Ruggles 0746a86c61940e: alacdec: set bits_per_raw_sample * 03http://tinyurl.com/9zy555a03 [15:01] <cone-571> ffmpeg.git 03Anton Khirnov 07d6f4fe68c8e8: lavc: extend frame_size doxy. * 03http://tinyurl.com/9aolmt203 [15:01] <cone-571> ffmpeg.git 03Anton Khirnov 076173a8fe5d2c: riff: remove a write-only variable * 03http://tinyurl.com/8eovuy903 [15:01] <cone-571> ffmpeg.git 03Michael Niedermayer 07f9cf14c8da2a: Merge remote-tracking branch 'qatar/master' * 03http://tinyurl.com/9mfuo6b03 [15:04] <saste> someone can recommend an image formats which allows to read the values in a human readable way? [15:04] <saste> sng is a possibility, so i was contemplating the possibility to add native support to ffmpeg [15:04] <saste> XPM may be another alternative [15:05] <ubitux> ppm ? [15:05] <saste> ffmpeg -i tests/lena.pnm lena.ppm [15:06] <saste> vim lena.ppm... not very much [15:06] <saste> IIRC PPM has an ASCII option which we don't support [15:06] <ubitux> https://en.wikipedia.org/wiki/Netpbm_format [15:06] <ubitux> this is what i call "ppm" [15:06] <av500> yes [15:07] <ubitux> https://en.wikipedia.org/wiki/Netpbm_format#PPM_example [15:07] <saste> we don't support the ASCII format [15:07] <ubitux> saste: nit++: 's' is before 'v' [15:07] <ubitux> (ffescape, ffeval -- gitignore) [15:08] <saste> ubitux, that list is not sorted anyway [15:08] <saste> locally fixed [15:08] <ubitux> and yes xpm are indeed another one [15:09] <saste> so why esr implemented sng? [15:09] <ubitux> (vim even renders directly xpm :p) [15:17] <saste> replying to myself: SNG supports palette data, so it's ideal for storing/viewing/editing paletted images (like PNGs) [15:54] <cbsrobot> durandal_1707: what happend to http://ffmpeg.org/pipermail/ffmpeg-devel/2012-June/125668.html ? [15:55] <durandal_1707> rot, because nobody REing prores alpha part [15:56] <durandal_1707> it can be added, but nothing will use it, because there is no other codecs that use this format, and anyway it is not really imprtant that much because there is yuva444p16 [15:56] <durandal_1707> and switch can be done at any time once prorse alpha code comes [16:40] <merbanan> durandal_1707: dnxhd also has an alpha mode that isn't re'd [16:42] <durandal_1707> isn't that raw? [16:43] <durandal_1707> but that is 422 [16:43] <merbanan> it is vc3- based [18:12] <cone-571> ffmpeg.git 03Michael Niedermayer 0722793d7bb36e: ffmpeg/lavc: move experimental warnings to libavcodec. * 03http://tinyurl.com/8uw866r03 [19:44] <cone-571> ffmpeg.git 03Matthieu Bouron 07d8173f264f73: dv: fix a check on dv_extract_timecode return value * 03http://tinyurl.com/8jldf9x03 [19:44] <cone-571> ffmpeg.git 03Matthieu Bouron 07c68a8a1340e7: lavf/aiffdec: don't stop parsing after SSND chunk * 03http://tinyurl.com/8wgqwhz03 [20:46] <durandal_1707> what to use to set jpeg encoder quality/compression? [20:51] <michaelni> -qscale X [20:53] <durandal_1707> but from lavc user point of view [22:14] <ubitux> anyone to have a look to http://ffmpeg.org/pipermail/ffmpeg-devel/2012-October/132854.html ? [22:15] <ubitux> i added the following description: [22:15] <ubitux> FATE_SAMPLES_FFPROBE is a different list than FATE_FFPROBE, the same way [22:15] <ubitux> FATE_SAMPLES_FFMPEG is a different list than FATE_FFMPEG; these [22:15] <ubitux> FATE_SAMPLES_<tool> lists are in the FATE_EXTERN list, which is only [22:15] <ubitux> used when SAMPLES is set. [22:21] <ubitux> mmh i'm wondering [22:22] <ubitux> what's wrong actually with the msvc fate instance? [22:22] <ubitux> the configuration is --samples=/d/Dev/ffmpeg/fate/samples [22:23] <ubitux> is the ':' also missing on fate display? [22:23] <ubitux> Daemon404: how is your --samples configured? [22:24] <Daemon404> my what now? [22:24] <Daemon404> context? [22:25] <ubitux> Daemon404: the msvc fate instances are yours, right? [22:25] <ubitux> i'd like to know you set the samples directory [22:25] <Daemon404> theyre nev's [22:25] <ubitux> and if the path is indeed "/d/Dev/ffmpeg/fate/samples" [22:25] <ubitux> oh, ok, sorry [22:25] <Daemon404> i have one msvc instance of a shared build, and its for libav [22:25] <Daemon404> since ffmpeg's doesnt work yet [22:25] <Daemon404> iirc [22:26] <ubitux> nevcairiel: ping :) [22:26] <ubitux> Daemon404: ok :( [22:26] <nevcairiel> wut [22:26] <ubitux> nevcairiel: what's your samples path in the msvc instances? [22:26] <Daemon404> im working on some exotic setups atm... [22:26] <Daemon404> nerbsd/armv5 [22:26] <Daemon404> netbsd* [22:26] <nevcairiel> technically the path is D:\Dev\ffmpeg\... however because its running in a msys bash shell, it gets the msys-style path [22:26] <llogan> nerdsb [22:26] <nevcairiel> msys should translate that to the real path before ffmpeg ever sees it [22:27] <nevcairiel> probably d:/dev/... [22:27] <Daemon404> yes [22:27] <Daemon404> is this that : problem again? [22:27] <ubitux> yes likely [22:28] <ubitux> right now ffmpeg sees '/d/Dev/ffmpeg/fate/samples/svq3/Vertical400kbit.sorenson3.mov' [22:28] <nevcairiel> i doubt that [22:29] <ubitux> well: http://fate.ffmpeg.org/report.cgi?time=20121022182926&slot=x86_64-msvc10-wi… [22:29] <Daemon404> expecting users to escape : in path names is dumb. [22:29] <Daemon404> just sayin. [22:29] <nevcairiel> the script is called with that [22:29] <nevcairiel> but its translated before ffmpeg ever gets it [22:29] <Daemon404> yup [22:29] <nevcairiel> or should be at least [22:29] <nevcairiel> msys is sometimes funny [22:30] <cone-571> ffmpeg.git 03Michael Niedermayer 07250fe6eeb4a1: cmdutils: apply option to codec and format contexts if possible. [22:30] <ubitux> this is a particular case :p [22:30] <ubitux> i wonder if i shouldn't just define another command in place of "run" [22:31] <ubitux> ah well that's stupid [22:31] <ubitux> it wouldn't help either [22:31] Action: Daemon404 notes a 15 gb image takes forever to dd to an sdhc card [22:32] <nevcairiel> hm how did you trick msys into not translating the path? [22:32] <nevcairiel> it did translate it when i looked at the error yesterday [22:32] <ubitux> i hid it in a string [22:34] <nevcairiel> i can probably specify a windows-style path in the fate config [22:35] <Daemon404> [16:32] < nevcairiel> hm how did you trick msys into not translating the path? [22:35] <Daemon404> this is the wrong "solution" [22:36] <ubitux> what solution do you see within a filtergraph Daemon404? [22:36] <Daemon404> ubitux, fix the dang parser [22:36] <Daemon404> so it handles VALID patsh correctly [22:36] <ubitux> well, it's just opening the passed path [22:36] <nevcairiel> in the recent error messages, ffmpeg is actually getting a msys path [22:36] <ubitux> which is '/d/Dev/ffmpeg/fate/samples/svq3/Vertical400kbit.sorenson3.mov' [22:37] <nevcairiel> because msys failed to identify the path in the middle of a argument [22:37] <Daemon404> ah [22:37] <Daemon404> yes [22:37] <Daemon404> msys wont tl that [22:37] <Daemon404> it has specific rules about what it will tl [22:37] <nevcairiel> so i just change it to windows path with wrong-way-slashes [22:37] <nevcairiel> work both for win32 and msys apps [22:38] <ubitux> the problem will then be the ':' [22:38] <Daemon404> if it cant handle the : [22:38] <ubitux> (i guess) [22:38] <Daemon404> thats a bug that should be fixed [22:38] <Daemon404> period [22:38] <Daemon404> or rather [22:38] <Daemon404> colon. [22:38] <ubitux> how? [22:38] <Daemon404> ubitux, why cant it handle : in the first place [22:38] <nevcairiel> because its the key-value pair seperator in filtergraph arguments [22:38] <ubitux> :) [22:39] <Daemon404> well you choose that delim poorly. [22:39] <ubitux> right, but we can't change it now [22:39] <nevcairiel> i think the whole idea of the movie source is a poor one [22:39] <Daemon404> me too [22:39] <nevcairiel> but thats just me [22:39] <ubitux> again, do you see another solution? [22:39] <Daemon404> if it was sane [22:40] <ubitux> having sources in the filtergraph is nice IMO [22:40] <Daemon404> hwy? [22:40] <Daemon404> i dont see ANY benefit [22:40] <Daemon404> from doing -i [22:40] <ubitux> you can describe completely your filtergraph with a string, which simplifies the API a lot [22:40] <ubitux> not all tools want to reproduce the filter complex thing from ffmpeg [22:41] <ubitux> and that's indeed not done in ffprobe, and ffplay [22:41] <ubitux> as the name suggest, it's "complex" [22:41] <Daemon404> yes, one horribly ugly string [22:41] <Daemon404> :| [22:41] <Daemon404> i dont see a benefit to having the src specified in it [22:41] <ubitux> i just told you... :/ [22:41] <Daemon404> [16:40] <@ubitux> not all tools want to reproduce the filter complex thing from ffmpeg [22:42] <Daemon404> thats called design failure [22:42] <Daemon404> but not making that an api [22:42] <ubitux> well it really is simpler [22:42] <ubitux> you only need to send a string to your app [22:42] <Daemon404> there is nothing simpel about lavfi's api. [22:42] <ubitux> to describe a filtergraph [22:42] <Daemon404> or even its input strings [22:43] <Daemon404> its akin to reading lisp [22:43] <Daemon404> for me [22:43] <Daemon404> ;) [22:43] <ubitux> i wonder if we shouldn't just add a CONFIG_SANE_ENV and put SANE_ENV in the deps of these tests [22:45] <nevcairiel> the inability to be able to deal with windows path is something that should be fixed [22:45] <ubitux> anyway, if the path is transformed to a "d:\foo\bar" syntax, it should be possible to handle it without much trouble when ffescape is in [22:45] <ubitux> nevcairiel: it is able to deal with windows path [22:45] <Daemon404> if it cant handle C:/depr.mov [22:45] <Daemon404> no [22:45] <Daemon404> no it cannot. [22:45] <ubitux> it can, you just have to escape the : afaict [22:45] <Daemon404> thats not handling it [22:45] <Daemon404> at all [22:45] <Daemon404> >_> [22:45] <ubitux> please [22:46] <Daemon404> im serious [22:46] <Daemon404> that is insanely dumb [22:46] <ubitux> we can't just change the whole syntax of filtergraph just because you don't want to add a '\' [22:46] <Daemon404> or add some basic checks? [22:46] <ubitux> or maybe that's actually a solution mmh [22:46] <Daemon404> for e.g. /,.:/ [22:46] <Daemon404> or something [22:46] <ubitux> like setting a global separator [22:46] <Daemon404> yes [22:46] <Daemon404> like sed :P [22:47] <Daemon404> s#a#b# [22:47] <Daemon404> or s/a/b/ [22:50] Action: ubitux wonders about what other character to use [22:50] <ubitux> mmh actually [22:51] <ubitux> would "movie='d:\foo\bar':xx=..." not working? [22:51] <ubitux> it's using av_get_token so it might work [22:52] <michaelni> hmm [22:52] <michaelni> why is this failing anyway ? [22:52] <nevcairiel> damnit, there is some bash magic in the fate test that checks if the samples path starts with a slash [22:52] <nevcairiel> unportable shit ftl [22:52] <michaelni> there is surely no "d" option in the filter [22:52] <michaelni> so d: could easily be handled [22:53] <ubitux> michaelni: right now because it's not passing "d:\foo\bar" but the other path [22:53] <ubitux> and then it will fail because of ':' escaping which IMO can be avoiding by just adding '' around $(SRC) [22:54] <michaelni> that should work indeed [22:55] <michaelni> something quite different, i think filter graph descriptions in genera need more whitespace and newlines, that would make them more readable [22:56] <ubitux> https://github.com/ubitux/FFmpeg/compare/master...fate [22:56] <Daemon404> inb4 add a \ [22:56] <Daemon404> and hit return [22:56] <ubitux> spaces should be skipped yes [22:56] <cone-571> ffmpeg.git 03Michael Niedermayer 070de41ead6f90: qt-faststart: check fseeko() return codes [22:57] <ubitux> < nevcairiel> damnit, there is some bash magic in the fate test that checks if the samples path starts with a slash // huh? [22:57] <nevcairiel> well there is [22:58] <nevcairiel> it uses that to try to check if its a relative path [22:58] <ubitux> ah, ok [22:58] <nevcairiel> and something like d:/... looks relative to it :P [22:58] <ubitux> haha [22:59] <nevcairiel> arent there bash-builtins to do this stuff [22:59] <nevcairiel> why the manual hackery [22:59] <nevcairiel> oh well [22:59] <nevcairiel> no fixing the path then [22:59] <ubitux> ok so, how are we supposed to handle that '/d/...' thing? [23:00] <nevcairiel> you're not, fix fate :p [23:00] <ubitux> so much trolling just to realize that there is nothing wrong with lavfi :( [23:01] <nevcairiel> it still failed on the : path before you added those sws flags [23:01] <ubitux> what ':'? [23:01] <nevcairiel> http://fate.ffmpeg.org/report.cgi?time=20121021161916&slot=x86_64-msvc10-wi… [23:01] <ubitux> if we add the '' quoting it will work just fine [23:02] <ubitux> i can push my branch right now to fix that [23:02] <ubitux> i'd just need a ok for 01f2386 and maybe the others [23:02] <nevcairiel> i should figure out a more portable way to figure out if a path is abolute in bash [23:02] <nevcairiel> i'm sure there is a real function for that [23:03] <nevcairiel> all the quick solutions on google check for damn slashes [23:04] <ubitux> it's simple [23:04] <ubitux> we should just check for damn ':' [23:06] <nevcairiel> i could just leave out the drive from the path, its all on the same one anyway, so /Dev/... would work [23:06] <nevcairiel> but thats not a universal solution [23:06] <nevcairiel> and it probably wouldnt work with any msys tools [23:09] <ubitux> wait. [23:09] <ubitux> Error initializing filter 'movie' with args 'd:/Dev/ffmpeg/fate/samples/svq3/Vertical400kbit.sorenson3.mov' [23:09] <ubitux> so this is what you actually pass to the filter [23:09] <nevcairiel> if msys manages to translate the path [23:09] <nevcairiel> or the native path is passed [23:10] <ubitux> ah and the sws_flags thing broke that detection? [23:10] <nevcairiel> yeah, it works if its the first argument [23:10] <nevcairiel> but the sws flags appeared before it [23:11] <ubitux> how does this detection work? [23:11] <nevcairiel> a lot of magic, i guess [23:11] <ubitux> i'm not even sure it will detect it if we enclose it within '' [23:11] <nevcairiel> they might expect quoting [23:12] <ubitux> well we can see it with the silencedetect one [23:12] <ubitux> since it hasn't sws flag thing [23:13] <ubitux> michaelni: do you mind if i push my branch? [23:13] <michaelni> ubitux, if it fixes things sure push it [23:13] <ubitux> ok [23:14] <cone-571> ffmpeg.git 03Clément BSsch 077be9c0c10fc5: fate: fix SAMPLES dependency for ffprobe. [23:14] <cone-571> ffmpeg.git 03Clément BSsch 07ae69c683e58c: fate: add scale filter to the scene detect dependencies. [23:14] <cone-571> ffmpeg.git 03Clément BSsch 072649b78384a7: fate: improve metadata filter deps readability. [23:14] <cone-571> ffmpeg.git 03Clément BSsch 077c2d5eec29e0: fate: quote file paths in movie/amovie filtergraphs. [23:14] <ubitux> let's see now. [23:15] <cbsrobot> so now cone wont give an url anymore ? [23:15] <ubitux> haha :) [23:15] <ubitux> better than a tiny url! [23:16] Action: cbsrobot stares at gnafu [23:17] <ubitux> nevcairiel: i see you're doing some experiments with one msvc instance :D [23:18] <nevcairiel> i tried to change the path, but i found that stupid absolute-path check :p [23:19] <ubitux> i just pushed the '' btw, if you want to give it a try [23:23] <ubitux> nevcairiel: i'd like to have a look to the code doing the path translation; what part of what project is responsible for this? [23:23] <nevcairiel> in msys? i have no clue [23:23] <ubitux> msys from mingw? [23:24] <nevcairiel> yeah its somewhere in one of the msys runtime libs that get linked into the executables [23:24] <nevcairiel> no idea where [23:24] <ubitux> ok, thanks [23:25] <ubitux> git clone git://mingw.git.sourceforge.net/gitroot/mingw/mingw [23:25] <ubitux> Cloning into 'mingw'... [23:25] <ubitux> warning: You appear to have cloned an empty repository. [23:25] <ubitux> :( [23:25] <Daemon404> mingw uses svn [23:25] <Daemon404> why are you cloning mingw [23:25] <Daemon404> >____> [23:25] <Daemon404> nothing good can come of this [23:26] <ubitux> http://sourceforge.net/projects/mingw/develop [23:26] <ubitux> i see cvs & git :p [23:26] <nevcairiel> you probably want http://mingw.git.sourceforge.net/git/gitweb.cgi?p=mingw/msys-runtime;a=summ… [23:26] <ubitux> thx [23:26] <Daemon404> again [23:26] <Daemon404> why [23:26] <Daemon404> <_< [23:26] <nevcairiel> he wants to find the path translation code :p [23:26] <Daemon404> .. [23:26] <Daemon404> theres an entire damn page [23:26] <Daemon404> dedicated to it [23:26] <Daemon404> on their site [23:27] <Daemon404> http://www.mingw.org/wiki/Posix_path_conversion [23:27] <ubitux> oh great [23:27] <ubitux> thx [23:27] <ubitux> http://mingw.cvs.sourceforge.net/viewvc/mingw/msys/rt/src/winsup/cygwin/pat… *_* [23:27] <Daemon404> viewvc is horrible [23:27] <Daemon404> end of story [23:28] <Daemon404> :3 [23:29] <nevcairiel> my fate finished building, nothing changed [23:31] <nevcairiel> anyway, time for sleep [23:31] <ubitux> mmh this looks weird [23:31] <ubitux> night :) [23:33] <ubitux> Missing key or no key/value separator found after key '/Dev/ffmpeg/fate/samples/amrwb/seed-12k65.awb' [23:33] <ubitux> this is pretty weird [23:37] <ubitux> as if the msys escaping was just breaking everything [23:43] <ubitux> mmh something will need to be changed in src movie maybe [23:45] <ubitux> ok get it [23:59] <ubitux> "movie=f='c:\foo\bar'" doesn't actually receive "f='c:\foo\bar'" as arg, but "f=c:\foo\bar" [23:59] <ubitux> that's what causing the issue [23:59] <durandal_1707> heh sws cant handle 4x4 images [23:59] <ubitux> hey saste, perfect time! :) [23:59] <saste> ubitux, i want to tell that the ' should work [23:59] <ubitux> i commited [23:59] <ubitux> but it doesn't [23:59] <ubitux> because of what i just said [00:00] --- Tue Oct 23 2012

1 0

ffmpeg.log.20121022
by burek 23 Oct '12

23 Oct '12

[00:34] <_stclaws> I am trying to compile ffmpeg with rtmp support. I first build rtmpdump. But when doing configure of ffmpeg I get: ERROR: librtmp not found. Any idea why? [01:05] <iive> _stclaws: check config.log [01:12] <_stclaws> iive: I tried everything I could think of. [01:12] <iive> _stclaws: the config.log should contain the failed check, together with the reason for the failure. [01:13] <_stclaws> It just ends with : check_pkg_config librtmp librtmp/rtmp.h RTMP_Socket [01:13] <_stclaws> ERROR: librtmp not found [01:14] <iive> looks like you installed only the library and the -dev with the headers is missing [01:15] <iive> can you find rtmp.h somewhere in your filesystem? /usr/include/librtmp/rtmp.h ? [01:15] <_stclaws> librtmp is here: /usr/local/include/librtmp [01:16] <_stclaws> I thought all would be included with rtmpdump [01:16] <iive> redhad doesn't include /usr/local in their defaults, so you may try specifying it as additional include path. [01:18] <iive> try: --extra-cflags=-I/usrl/local/include [01:21] <iive> oops, /usr/local/include .... [01:22] <_stclaws> iive: tried that path but nope (yes usr) [01:23] <_stclaws> I read somewhere that pkg-config had something to do with it [01:23] <iive> same error in the config.log? [01:23] <iive> can you pastebin a little bit more from the end? [01:25] <_stclaws> http://pastebin.com/U9tEbivS [01:26] <_stclaws> and my config line: http://pastebin.com/3L6j6Kn6 [01:49] <_stclaws> I did add PKG_CONFIG_PATH=/usr/local/lib/pkgconfig because that's where librtmp.pc is [01:51] <_stclaws> Am I supposed to install librtmp-dev also although I did install rtmpdump? [02:06] <hypnocat> when i try to ffplay an mp3, instead of playing it, ffplay displays a new window with a spectrogram of the mp3 [02:07] <hypnocat> why does it do that? and how can i get it to just play the file? [02:07] <sacarasc_> It should also be playing it. [02:08] <hypnocat> ah.. i see that it was.. but just through alsa, not jack [02:09] <hypnocat> how can i turn off the spectrogram display? [02:11] <sacarasc_> Why would you want to? [02:11] <hypnocat> i don't want an extra window to appear, or for ffplay to waste extra processing power displaying it [02:11] <hypnocat> it's distracting, annoying, and wasteful [02:12] <sacarasc_> Minimise it, and if it's wasteful, your computer is either really, really old or you're overly sensitive to waste... [02:13] <hypnocat> my computer is old [02:16] <_stclaws> Ok, I am stumped, it is getting late, and I guess I soon give up... [02:22] <iive> _stclaws: sorry, I got to do other things. [02:22] <iive> installing in /usr/local usually indicates you compiled yourself, so -dev package may not work [02:23] <iive> good call with the pkg config path, but i assume it didn't work either? [02:23] <iive> try to call it directly. [02:29] <iive> e.g. pkg-config --cflags --libs librtmp [02:29] <iive> the configure actually uses pkg-config --exists librtmp [02:29] <iive> but that returns true/false to the script. [02:30] <iive> i must try rtmp myself. maybe tomorrow. [02:36] <_stclaws> when I try to call it directly it complains that libssl is not found [02:37] <_stclaws> it is in /usr/lib/pkgconfig/libssl.pc . So I add that path to PKG_CONFIG_PATH. But still saying Package 'libssl', required by 'librtmp', not found [02:55] <_stclaws> Crap, I just forgot to do export of PKG_CONFIG_PATH :) [03:10] <_stclaws> thanks for the help, iive! [03:15] <_stclaws> Does anyone know what the proper syntax is for the "listen" and "rtmp_live" parameters of the rtmp protocol? [04:39] <zelozelos> i have another question about codec and ffmpg. if i was to output to mp4 then change to wmv, will i save some file size? [06:56] <mark4o> _stclaws: ffmpeg -rtmp_live 1 ... / ffmpeg -rtmp_listen 1 ... [06:57] <mark4o> _stclaws: I think rtmp://...?listen&timeout=... may also work [07:02] <smjm> what's a good lossless video format? [10:13] <_stclaws> I am receiving a rtmp stream from a fms server, but the stream is now and then "renewed" since the fms source is switched. I want ffmpeg to just go on but it thinks the stream ended and exits. Does anyone know how to just keep it up listening and receiving indefinitely? [10:13] Last message repeated 1 time(s). [10:13] <_stclaws> sorry twice [10:17] <Exatrive> Hello.. have a question.. is there any way to have ffmpeg streams not enter a container. I'm capturing audio/video but mp4 doesn't support pcm_s16le audio so I will get a error if I try to capture. Instead I would like to have video go into the mp4 but audio go to another .wav in realtime. [10:18] <Tjoppen> Exatrive: look at the documentation for -map [10:19] <Exatrive> I did look at the map function.. but its not too clear if this will allow it to map the channel to bypass the first container. I will look again and see what I can do... [10:27] <cjhmdm> hello, I'm trying to build the latest ffmpeg from source following the ubuntu compilation guide and am having trouble during the make step for fdk-aac [10:27] <cjhmdm> this is the error I'm getting on make: libtool: link: more than one -exported-symbols argument is not allowed [10:27] <cjhmdm> make: *** [libfdk-aac.la] Error 1 [10:41] <saste> cjhmdm, looks like an autohell libfdk issue [10:41] <cjhmdm> saste: autohell? [10:42] <saste> cjhmdm, autohell = autotools (autoconf, automake, libtool, etc.) [10:43] <cjhmdm> saste: any way I can resolve it? [10:43] <saste> cjhmdm, ask to libfdk devs, they will be more qualified to help [10:46] <cbsrobot> cjhmdm: did you get the source from git ? [10:47] <cjhmdm> cbsrobot: yes, I'm following the guide, line for line here: https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [10:47] <cbsrobot> these are the exported symbols: https://github.com/mstorsjo/fdk-aac/blob/master/fdk-aac.sym [10:48] <cbsrobot> campare it to your fdk-aac.sym [10:49] <cjhmdm> https://gist.github.com/3930424 [10:49] <cbsrobot> maybe you can debug it in the makefile [10:49] <cbsrobot> after the autoreconf step [10:49] <cjhmdm> it's telling me I can't have more than 1 exported-symbol [10:50] <cjhmdm> my fdk-aac.sym file is the same as the one you linked [10:50] <cjhmdm> unless I'm blind heh [10:51] <cbsrobot> just before this line https://github.com/mstorsjo/fdk-aac/blob/master/Makefile.am#L32 [10:51] <cbsrobot> see if you can find it in the Makefile [10:52] <cjhmdm> libfdk_aac_la_LDFLAGS = -version-info 0:1:0 -no-undefined \ [10:52] <cjhmdm> -export-symbols $(top_srcdir)/fdk-aac.sym [10:53] <cbsrobot> what does your shell say about "echo $LDFLAGS" [10:53] <cjhmdm> nothing [10:53] <cjhmdm> no return [10:54] <cbsrobot> see line 384 in your Makefile [10:54] <cjhmdm> what I don't get is why the trouble all of a sudden.. I've built, rebuilt and updated ffmpeg on this machine around 6 times over the past year with no problem [10:55] <cbsrobot> maybe you can print the export-symbols jut before that [10:55] <cjhmdm> although, I do blieve the fdk-aac step is new [10:55] <cjhmdm> line 384 in my makefile is: $(MPEGTPDEC_DIR)/tpdec_latm.cpp \ [10:56] <Exatrive> I'm still not having any luck with the map option.. the audio stream still want to be saved in the mp4 file... with a copy of the audio in the .wav [10:56] <cbsrobot> cjhmdm: search for export-symbols [10:56] <Exatrive> I'm trying to capture video to mp4, and audio to .wav seperatly with no luck. [10:57] <cjhmdm> cbsrobot: line 297: -export-symbols $(top_srcdir)/fdk-aac.sym [10:57] <cjhmdm> the whole line: [10:57] <cjhmdm> libfdk_aac_la_LDFLAGS = -version-info 0:1:0 -no-undefined \ [10:57] <cjhmdm> -export-symbols $(top_srcdir)/fdk-aac.sym [11:01] <cjhmdm> go figure.. compiles just fine on a fresh machine with the same hardware and os as the other machine (but with only base system and necessary build tools).. but all dev tool versions are the same, so can I deduce that it's conflicting with another package on the system? [11:03] <cbsrobot> cjhmdm: search for "all:" in the makefile [11:04] <cbsrobot> and put "@echo $(libfdk_aac_la_LDFLAGS)" after that line [11:04] <cbsrobot> run make again [11:04] <cjhmdm> cbsrobot: all: all-am [11:04] <cbsrobot> see what it prints [11:04] <cbsrobot> yes [11:04] <cjhmdm> Makefile:484: *** target pattern contains no `%'. Stop. [11:04] <cbsrobot> without the quotes [11:04] <cbsrobot> @echo $(libfdk_aac_la_LDFLAGS) [11:05] <cjhmdm> I didn't put quotes :P [11:05] <cjhmdm> all: all-am [11:05] <cjhmdm> @echo $(libfdk_aac_la_LDFLAGS) [11:05] <cbsrobot> with a tab in fron [11:05] <cjhmdm> ahh [11:05] <cbsrobot> *front [11:06] <cjhmdm> https://gist.github.com/3930510 [11:07] <cbsrobot> in your shell [11:07] <cbsrobot> wait [11:07] <cjhmdm> that was the output after adding that line to the makefile and running make [11:09] <cbsrobot> add "@echo $(LINK)" [11:09] <cbsrobot> after the other line [11:09] <cbsrobot> and show me the output [11:09] <mkozjak> hello [11:09] <cjhmdm> so the above line then @echo $(LINK) below it? tabbed also? [11:09] <cbsrobot> yes [11:09] <cjhmdm> same output [11:10] <cjhmdm> wait, I think there's a bit more on there [11:10] <mkozjak> i have a problem with building the latest snapshot or git versions of ffmpeg where i get http://pastebin.com/raw.php?i=AGvrTcBq [11:10] <mkozjak> echo $LD_LIBRARY_PATH says /usr/local/lib: [11:10] <cjhmdm> cbsrobot: https://gist.github.com/3930534 [11:11] <cbsrobot> mkozjak: make clean [11:11] <mkozjak> x264 built with Command line options: "--enable-pic" "--enable-shared" "--extra-cflags=-march=native" [11:11] <cjhmdm> cbsrobot: nevermind.. it is the same output [11:11] <cbsrobot> make clean && make [11:11] <mkozjak> cbsrobot: yeah, doesn't help [11:11] <cbsrobot> cjhmdm: swap the two lines [11:11] <mkozjak> make clean && make distclean && make -j20 [11:11] <cjhmdm> cbsrobot: all: all-am [11:11] <cjhmdm> @echo $(LINK) [11:11] <cjhmdm> @echo $(libfdk_aac_la_LDFLAGS) [11:12] <cjhmdm> same output [11:13] <_stclaws> Could anyone point me to a good tutorial for how to do HTTP Live Streaming with ffmpeg? From an incoming live rtmp stream. [11:14] <cbsrobot> cjhmdm: hmmm [11:14] <cjhmdm> cbsrobot: [11:14] <cjhmdm> err.. yeah, hmm heh [11:14] <cbsrobot> I show you my output [11:14] <cjhmdm> the weirdest thing is, when running make on the new machine, the output is entirely different [11:14] <cbsrobot> -version-info 0:1:0 -no-undefined -export-symbols ./fdk-aac.sym [11:14] <cjhmdm> and it compiles fine [11:15] <cbsrobot> /bin/sh ./libtool --silent --tag=CC --mode=link gcc -g -O2 -o all [11:15] <cjhmdm> but the only other thing on the 'old' machine is wowza media server [11:15] <cbsrobot> firts line is @echo $(libfdk_aac_la_LDFLAGS) [11:15] <cjhmdm> and the needed libs for building ffmpeg [11:15] <cbsrobot> second $(LINK) [11:16] <cbsrobot> try to figure out where in the makefile it adds the second export-symbols [11:16] <mkozjak> can i somehow define to configure where the system should look for libx264.so? [11:17] <cbsrobot> cjhmdm: search for libfdk_aac_la_LDFLAGS and see how it is created [11:18] <wyuka> hi [11:18] <cjhmdm> cbsrobot: ok [11:18] <wyuka> any of the ffvp8 devs around? [11:20] <cjhmdm> cbsrobot: what I don't get is this also... this is the output on the old machine: https://gist.github.com/3930568 --- this is the output on the new machine: https://gist.github.com/3930561 --- the 2 machines are identical in every way. The only difference is the old machine has other packages (unrelated to building ffmpeg) installed [11:21] <cbsrobot> ~/ffmpeg/libvpx ? [11:21] <cjhmdm> cbsrobot: libfdk_aac_la_LDFLAGS = -version-info 0:1:0 -no-undefined \ [11:21] <cjhmdm> -export-symbols $(top_srcdir)/fdk-aac.sym [11:21] <cbsrobot> cjhmdm: wrong copy paste ? [11:22] <cjhmdm> no, that's where libfdk_aac_la_LDFLAGS is created [11:22] <cbsrobot> see https://gist.github.com/3930561 [11:22] <cbsrobot> wrong paste [11:22] <cjhmdm> err yeah [11:22] <cjhmdm> lol [11:22] <cjhmdm> one sec [11:24] <cjhmdm> this is the proper output on the new machine: https://gist.github.com/3930586 [11:24] <cbsrobot> you need to make clean on this machine first [11:25] <cjhmdm> the old or new? [11:25] <cbsrobot> new [11:25] <wyuka> michaelni: ping [11:25] <cjhmdm> I just followed the guide heh [11:26] <cjhmdm> same output after running make clean heh [11:26] <wyuka> i am working on an algorithm that generates intermediate frames using the motion compensation information during decoding [11:26] <wyuka> i need some help with the source code for h264.c [11:27] <cbsrobot> wyuka: check #ffmpeg-devel [11:28] <wyuka> oh okay [11:30] <cjhmdm> cbsrobot: oh well.. I'll try not to stress it heh.. just forces me to set up the new processing server anyway. I just hate not being able to figure out why something is broken heh [11:32] <cjhmdm> cbsrobot: thanks for trying to help either way [11:32] <cbsrobot> np [11:59] <_stclaws> I need to set up ffmpeg to receive multiple consecutive rtmp live streams and re-encoding them into one single stream. Can anyone suggest the best way for doing that? [12:05] <Element9> What do I have to enable while configuring for building to get WMV2 encoding? [12:05] <Element9> --enable-wmv2 alone apparently isn't enought. here are all of the options I'm using if it's relevant: http://pastebin.com/u7XV393g [12:06] <Exatrive> I figured out the mapping.. omg... my command line looks like a mess with all this mapping but at least it works.. lol [12:07] <cbsrobot> Element9: maybe --enable-encoder=h263 [12:08] <Element9> cbsrobot: oh, that's why h263 got enabled by itself :) and I disabled it explicitely [12:08] <Element9> cbsrobot: thanks. I'll try that [12:08] <cbsrobot> np [12:20] <gst-kaps> how do i make sure ffplay plays single channel files as single single & keep the sample rate as it is ? [13:49] <arpu> hello is it possible to select a stream with the hex data like #0:1[0x45] [13:50] <_stclaws> has anyone here done HTTP Live Streaming with ffmpeg? [13:51] <ubitux> yes [13:51] <ubitux> http://ffmpeg.org/ffmpeg.html#segment_002c-stream_005fsegment_002c-ssegment [13:51] <ubitux> _stclaws: look for "HLS" here [13:52] <ubitux> arpu: isn't this a program id? [13:53] <arpu> hmm could be the problem is if i restart ffmpeg the order of the streams (more audio streams) is different [13:58] <_stclaws> ubitux: I am just making my first tests but can't get the video to play on the web page. Can you see anything wrong here: http://pastebin.com/9tKdL7cJ [13:58] <_stclaws> ubitux: This is the page with the test video: http://217.70.33.15/hls/ [14:01] <_stclaws> ubitux: This was my ffmpeg command line: http://pastebin.com/HyRmtXgC [14:21] <_stclaws> Is this normal output when you do hls streaming from ffmpeg: http://pastebin.com/E55BLSrh [14:32] <ubitux> arpu: http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1 [14:32] <ubitux> look for "program id" [14:33] <arpu> ubitux, thx [14:33] <ubitux> _stclaws: ffplay 'http://217.70.33.15/hls/test1.m3u8' works for me [14:34] <arpu> i will try maybe like -v:p:0x45 ? [14:34] <_stclaws> Ok, now I get video. But it only shows up after I stop streaming. [14:35] <_stclaws> While ffmpeg is working and outputting the .ts and .m3u8 nothing plays on the web [14:36] <ubitux> arpu: select for what? [14:36] <ubitux> _stclaws: you need a player on front [14:37] <ubitux> or you need to just output a standalone webm [14:37] <arpu> Stream #0:0[0x44](deu): Audio: mp3 ([3][0][0][0] / 0x0003), 44100 Hz, stereo, s16, 128 kb/s [14:37] <arpu> Stream #0:1[0x45]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc [14:38] <ubitux> arpu: i mean, for -map? [14:38] <saste> _stclaws, did you try the +live flag? [14:42] <_stclaws> for the rtmp in ffmpeg, yes. The input is no problem. The problem is that the player on my web page cannot see the output until the streaming stops. [14:43] <_stclaws> ubitux: I am using the <video> tag. Isn't that enough? [14:43] <ubitux> depends on the browser [14:43] <_stclaws> ios [14:43] <_stclaws> safari [14:44] <ubitux> i don't know that browser [14:44] <ubitux> i though most of the browsers were able to play only mp4 & webm [14:44] <ubitux> dunno about the playlist contents [14:44] <_stclaws> well the browser plays it ok, but only after I close the stream. [14:45] <_stclaws> It's as if the files need to close first or something. [14:46] <_stclaws> I am streaming to the iphone. That's why I am doing this. [14:47] <_stclaws> ubitux: You said you could see the video. Can you see me now holding a cup in my hand? [14:48] <ubitux> i see gray artefacts [14:48] <_stclaws> any image at all? [14:49] <ubitux> sure i see your face [14:49] <_stclaws> no cup? [14:50] <ubitux> http://b.pkh.me/_stclaws.jpg [14:50] <ubitux> this is what i see. [14:50] <_stclaws> interesting. so you actually see the live stream [14:51] <_stclaws> what client do you use for looking [14:51] <ubitux> 14:33:53 <@ubitux> _stclaws: ffplay 'http://217.70.33.15/hls/test1.m3u8' works for me [14:51] <_stclaws> ok, cool. so it is something wrong with my web page then [14:51] <_stclaws> thanks [14:51] <_stclaws> got any sound? [14:52] <_stclaws> oh wait, i don't think you do. anyway. [16:47] <dv_> hello [16:47] <dv_> I see some ARM optimizations for the libavcodec h264 decoder [16:47] <dv_> but how good are these optimizations? is there still a lot of room for improvement, or has the maximum been reached? [16:48] <JEEB> I'd say in general !IA32/x86_64 is not really optimized /that/ much [16:48] <JEEB> so if you know ARM assembly, kick yourself out [16:48] <dv_> okay, because on a cortex-A8, a 640x480 24 fps h264 baseline video takes about 95% CPU usage when decoding [16:49] <dv_> (cortex-a8 running at 720mhz) [16:49] <JEEB> well, that sentence by itself makes no sense unless you note that you are indeed limiting the decoding to ~24fps [16:50] <dv_> i did. it is a video I generated [16:50] <dv_> specifically with 24 fps [16:50] <JEEB> uuhh [16:50] <dv_> oh wait [16:50] <dv_> now I get it :) [16:50] <dv_> yes, it is limited to 24 fps [16:50] <JEEB> ok, and it gets to 95% there [16:51] <dv_> yeah, a few % idle remain [16:51] <JEEB> feel free to see what kind of room for optimization you have [16:52] <dv_> also, any other profile completely overwhelms the cpu, but thats not really surprising [18:37] <stclaws_> does anyone know what the parameters "listen" and "live" for rtmp actually do? I can't find any descriptions of them except that they have something to do with live streams. [19:04] <Simex> Anyone around who would like to help me with a application I'm working on? [19:05] <Simex> I'm working on a application that acts as a ffmpeg wrapper so that Linux users can run ffmpeg easily without knowing the command syntax [20:07] <RunawayDevil> Hail [20:08] <RunawayDevil> Hey guys, there is a bug in the libass? When I open the HUD to see the converted file .srt to .ass is everything wrong [20:09] <RunawayDevil> EVERYTHING. the grammar, I mean, it should be correct like it was when I used .srt (ocidental latin 1) [20:13] <diegoviola> hi [20:13] <diegoviola> what kind of frame rate should i use for desktop capture? [20:13] <diegoviola> screencasting [20:14] <diegoviola> i'm using webm [20:32] <diegoviola> f** [20:32] <diegoviola> ffmpeg is segfaulting for me [20:32] <diegoviola> ffmpeg version 1.0 Copyright (c) 2000-2012 the FFmpeg developers [20:32] <diegoviola> when using x11grab feature [20:43] <diegoviola> strange [20:43] <diegoviola> the x11grab dies only when i screencast firefox [20:43] <diegoviola> when i do everything else it's ok [20:46] <llogan> diegoviola: that's weird. [20:47] <Simex> Anyone around who would like to help me with a application I'm working on? [20:47] <Simex> I'm working on a application that acts as a ffmpeg wrapper so that Linux users can run ffmpeg easily without knowing the command syntax [20:48] <llogan> Simex: that will be a busy task due to all of the ffmpeg versions floating out there and the various syntax changes...or do you mean via the API? [20:49] <Simex> Really? [20:50] <Simex> Mainly for Ubuntu in the beginning though [20:50] <llogan> ubuntu doesn't use FFmpeg [20:50] <Simex> It uses avconv I know [20:51] <Simex> But luckily I haven't come across any syntax problems [20:52] <llogan> it would be easier to either support one version, like a certain package from a certain distro in a certain repository, or distribute/include ffmpeg with your app. [20:52] <Simex> Maybe I will distribute it with it [20:52] <Simex> I have to for Windows [20:53] <llogan> make sure to read http://ffmpeg.org/legal.html [20:53] <Simex> I'm not really working on the packaging at the moment [20:53] <Simex> At the moment users should download ffmpeg their self but it is far from done [20:53] <Simex> Would you like to try it? [20:54] <llogan> did you talk with the WinFF guys (Big Matt, etc)? They will probably have good ideas. [20:54] <llogan> sure, i'll try it [20:54] <Simex> It is Java [20:54] <llogan> i don't have java installed. [20:54] <Simex> http://jenkins.marlinc.nl/job/2STREAM/ is where you can download it [20:54] <Simex> OpenJRE is fine [20:55] <rbennacer> hey guys , is there a way to repair a corrupted mp4? when i use the ffmpeg -i , i get [mov,mp4,m4a,3gp,3g2,mj2 @ 0x62d650] moov atom not found [20:55] <diegoviola> fflogger: http://pastie.org/5099617 [20:56] <llogan> Simex: i can just look at the code. where in github can i view the code that tells ffmpeg what to do? [20:57] <Simex> https://github.com/MMS-Projects/2stream/blob/develop/src/net/mms_projects/t… [20:58] <Simex> Then it uses https://github.com/MMS-Projects/2stream/blob/develop/src/net/mms_projects/t… to do some input device selecting [20:58] <llogan> diegoviola: does it crash only when using libvpx as the encoder (as you are now)? [21:00] <diegoviola> llogan: nope [21:00] <llogan> rbennacer: generally, no. or at least not with ffmpeg. [21:00] <diegoviola> llogan: take a look: http://pastie.org/5099644 [21:01] <llogan> why did you --disable-debug? [21:04] <diegoviola> llogan: you talking to rbennacer or me? [21:04] <diegoviola> I'm using ffmepg from archlinux [21:05] <llogan> that was for you [21:06] <diegoviola> oh, i have no idea how archlinux devs built ffmpeg [21:06] <diegoviola> i'm using their package [21:06] <llogan> a gdb output would be useful. [21:07] <llogan> and you can see if ffmpeg-git in AUR also seg faults [21:07] <diegoviola> ok [21:08] <rbennacer> what is the best way to repair mp4 file that have no moov atom? [21:09] <llogan> diegoviola: (or just get the code, ./compile --enable-gpl, make. no need to install) [21:09] <llogan> *./configure [21:10] <brontosaurusrex> rbennacer: perhaps qt-quickstart [21:11] <cbsrobot> rbennacer: that's not easy [21:11] <cbsrobot> there are some apps doing it [21:11] <cbsrobot> but to my knowlage ffmpeg cant (atm) [21:11] <rbennacer> i am jsut missing the moov atom in my metatdata though [21:11] <cbsrobot> qt-faststart is only for moving the moov atom to the boginning of the file [21:12] <diegoviola> llogan: ok, i will try that, thanks [21:12] <cbsrobot> normally the metadata is in the moov atom [21:12] <cbsrobot> not the other way round [21:13] <cbsrobot> what atoms do you have ? [21:13] <rbennacer> i don't know [21:13] <rbennacer> how can i check? [21:14] <cbsrobot> hexedit [21:14] <rbennacer> ok let me check [21:14] <cbsrobot> :P [21:14] <cbsrobot> you know what a hexeditor is ? [21:15] <rbennacer> yes [21:15] <rbennacer> :) [21:16] <rbennacer> i jsut did a hexedit on my file [21:16] <rbennacer> i am not sure i can find the atoms i have [21:16] <rbennacer> hahhaha [21:17] <cbsrobot> does vlc play it ? [21:17] <cbsrobot> maybe you could try http://www.videohelp.com/tools/My-MP4Box-GUI [21:17] <cbsrobot> though I never used it [21:31] <llogan> diegoviola: i'm unable to duplicate the issue using the same arch package. [21:33] <diegoviola> llogan: weird [21:43] <diegoviola> llogan: it must be an issue with the driver maybe? [21:43] <diegoviola> i'm on nouveau [21:43] <diegoviola> video driver [21:49] <llogan> diegoviola: you need to A) test ffmpeg git master, and/or B) get a gdb output from your crashing ffmpeg [21:49] <llogan> http://ffmpeg.org/bugreports.html [21:49] <llogan> gdb info in that link [21:49] <llogan> you may have to recompile the package via ABS without --disable-debug [21:49] <diegoviola> ok [21:54] <llogan> diegoviola: it is easy to test git-master: http://pastebin.com/YfdzxpaE [21:55] <diegoviola> ok i'll give that a try in a min [21:55] <diegoviola> thanks [21:56] <diegoviola> just been busy with something else here as well (unrelated to ffmpeg) [21:56] <diegoviola> :) [21:56] <diegoviola> i appreciate your help [21:56] <llogan> i forgot: sdl libxfixes [21:56] <llogan> add those to your pacman line [21:57] <undercash> hi [21:57] <undercash> anybody has a little idea how to transform this bash script in php [21:57] <undercash> http://pastebin.com/qDhb2VcZ [21:57] <undercash> it s to launch a playlist using the browser so i dont give server credentials.. [21:58] <llogan> no, but you may want to add "-pix_fmt yuv420p" [21:59] <undercash> not looking to improve my ffmpeg command right now but thanks :) [21:59] Action: llogan hits ctrl+z [22:00] <llogan> this isn't really the channel for such questions [22:00] <undercash> really.. [22:00] <undercash> i had to try.. [22:01] <dTal> undercash: you know, you can still invoke bash scripts using CGI [22:01] <dTal> there's not neccesarily any benefit to porting to PHP [22:10] <Igneom> could someone help me with a compile error? [22:12] <Igneom> anyone? [22:18] <llogan> Igneom: use a pastebin service to show your configure line and any useful errors. [22:21] <Igneom> here it is http://pastebin.com/eH6aWL76 [22:23] <llogan> Igneom: you have two different versions of x264 installed simultaneously in two different places [22:25] <llogan> why --enable-memalign-hack? [22:27] <Igneom> i was just following a tut on ffmpeg.org [22:28] <llogan> which one? [22:28] <Igneom> whereis command would find the other place where it could have the other x264 installed? [22:28] <Igneom> http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide this one [22:29] <llogan> enable-memalign-hack isn't shown there [22:29] <llogan> your configure options are very different than the guide [22:30] <Igneom> now i remember i mixed the configure options form that guide and from this guide http://www.grafxsoftware.com/faq.php/HOW-TO-install-ffmpeg-Mplayer-mencoder… [22:31] <llogan> a terrible idea [22:31] <Igneom> this second guide also gives the same error [22:32] <llogan> that's an awful guide. outdated and used shitty configure options. [22:33] <llogan> symlink libraries? messy. [22:33] <Igneom> so the best one to use is from ffmpeg.org? [22:33] <llogan> that one is kept up to date and should not be using dumb options [22:33] <llogan> but now you have to clean up the mess from the other guide [22:34] <Igneom> yeah, well i'm installing it on a vps, so i think it would be easier to make a complete reset than finding and correcting a lot of mistakes [22:35] <llogan> if that's not a pain in the ass, then go for it [22:36] <Igneom> won't be, i thought that this could get a little complicated , so it is almost the first thing i was doing on the vps, just have to harden the ssh again and it's all good to go [22:37] <Igneom> anyway, thanks for the help llogan [22:38] <llogan> good luck [22:38] <Igneom> just one last question, the guide is for ubuntu, and i'm using debian, will all the dependencies listed work on debian? [22:41] <llogan> which debian? [22:42] <Igneom> 6.0 [22:42] <llogan> the hardy version might be better: https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuideHardy [22:43] <llogan> it's missing some stuff though like fdk-aac [22:44] <Igneom> if i compile the missing things and add the correct flags on the ffmpeg configure options, it should work right? [22:45] <llogan> actually it appears only fdk-aac is missing [22:45] <llogan> which you may not need if you don't need an AAC encoder (and ffmpeg has one anyway, but people say it's not that great) [22:47] <Igneom> well the things i need are meconder, mplayer, ffmpeg (with support to jpeg, gif, png, freetype, h264, lame, xvid, mad, win32codecs, dv, 3gp, mp4, theora), yamdi or flvtool2 and mediainfo [22:48] <llogan> the guide won't install freetype support in ffmpeg. and i just removed theora instructions seconds ago (see Hardy guide history if you need it) [22:49] <llogan> and by support, do you mean decoding or encoding? [22:51] <Igneom> i think both for most of them, dunno, it's for a tube like site, and tht's what it says as requirements on the script site [00:00] --- Tue Oct 23 2012

1 0

ffmpeg-devel.log.20121021
by burek 22 Oct '12

22 Oct '12

[00:13] <ubitux> srt demuxer now outputing subrip packets, check. [01:11] <ubitux> bot broken? [01:11] <durandal_1707> i can reproduce mp3 seeking issue (though sample rate does not change for me) [01:12] <ubitux> (banned by tinyurl? :p) [01:14] <Compn> we are tired of the tinyurl [01:14] <Compn> because we dont twitter [01:28] <ubitux> saste: coverity spotted some stuff in xface [01:28] <saste> ubitux, where can i find coverity reports? [01:29] <ubitux> http://ubitux.fr/pub/pics/_xface-coverity.png [01:29] <ubitux> saste: log in http://scan5.coverity.com:8080/ [01:29] <ubitux> don't you have an account? [01:29] <saste> ubitux, no [01:29] <ubitux> ask michaelni :p [01:30] <ubitux> there are a few spotted issues in your filters ;) [01:31] <Compn> i dont know why michaelni doesnt just make generic ffmpeg shared login [01:31] <Compn> :P [01:32] <Compn> then anyone who just wants to look at it for 2 mins doesnt have to register [01:32] <durandal_1707> Compn: in this special case it does not make any sese [01:32] <durandal_1707> *sense [01:32] <ubitux> Compn: there is some kind of comment system, with login associated with who fixed the bug etc [01:33] <saste> goodnight [01:33] <ubitux> if saste had an account, i could have assigned him some issues ;) [01:33] <ubitux> 'night saste [01:33] <Compn> like anyone cares who fixes what bug on a 3rd party bug tracker [01:33] <Compn> :P [01:33] <ubitux> well, it's pretty nice :) [01:40] <durandal_1707> ubitux: jacosub support only some styles? [01:41] <ubitux> what do you mean? [01:42] <ubitux> the decoder supports only a limited subset of styles yes [01:42] <durandal_1707> i mean using mplayer -sub <file> [01:42] <ubitux> i guess it will use the internal mplayer decoder? [01:43] <durandal_1707> why why why [01:43] <ubitux> because mplayer has its own subtitles handling [01:43] <ubitux> anyway, did you really find jacosub subtitles? @_@ [01:44] <ubitux> i thought i got all the jacosub subtitles in the world [01:44] <durandal_1707> no, i use one from fate [01:44] <ubitux> ah, ok [01:44] <durandal_1707> this sucks that nothing use subtitle code [01:44] <ubitux> because there are < 70 jacosub files in the world [01:44] <ubitux> yeah, a bit [01:44] <ubitux> but it might not be mature enough yet [01:46] <ubitux> it might be time to look into this new api... [01:46] <ubitux> but i'd like to get done with the lavfi metadata first [04:55] <Plorkyeran> @ubitux> because there are < 70 jacosub files in the world <-- there's a lot more than 70 episodes of LoGH [09:23] <ubitux> Plorkyeran: hehe :) [09:24] <ubitux> http://samples.ffmpeg.org/sub/jss/ seems i don't have them :P [09:34] <ubitux> michaelni: btw i was wondering, isn't the split/merge side data problematic for stream copy? [09:36] <ubitux> (since the demuxer would add the side data to the packet, which is directly copied without the data being split at decoding) [09:36] <nevcairiel> imho the whole merged side-data thing is problematic. What if you just use avformat and send it so something else entirely afterwards? [12:39] <j-b> https://trac.videolan.org/vlc/ticket/7625 [12:39] <j-b> Any idea? [12:43] <michaelni> nevcairiel, in that case, if it doesnt work as is you can set AVFMT_FLAG_KEEP_SIDE_DATA [12:44] <michaelni> also if we assume the side data did has some purpose then you have to do something with it anyway [12:44] <michaelni> or that "purpose" will not work anymore [12:45] <ubitux> well isn't it problematic if for instance you remux a rgb8 avi? [12:45] <ubitux> or any side data thing [12:45] <ubitux> i don't think that's really a good idea to have that :( [12:49] <michaelni> do you have some testcase that fails ? if so iam happy to fix it [12:50] <saste> j-b: ratecontrol trouble [12:50] <saste> it's weird because i'd expect the rc_eq to be always set, but seems to be NULL [12:51] <michaelni> j-b, is that using ffmpeg or libav ? [12:51] <michaelni> i cant help with libav bugs [12:51] <j-b> michaelni: this is a linux VLC, so your guess is as good as mine [12:51] <ubitux> michaelni: i don't have any file at hand now, but any media with a rgb8 videostream (and so a palette side data) will fail, right? [12:51] <j-b> cvlc is only on Linux and so, of course, I don't control the packaging... [12:52] Action: j-b welcome to my life [12:52] <saste> j-b: don't you support a --report option or like that [12:53] <saste> must be an hell to have to guess which are the involved library versions [12:54] <j-b> saste: yes. As if it was not hard enough with libavcodec going from 52.25 to 54.99, with versions with and without decoders, with fork from Ubuntu-universe and Debian-Multimedia, now you have at least 2 forks... [12:56] <michaelni> j-b, can this issue be reproduced somehow with command line ffmpeg ? [13:00] <michaelni> also does vlc do something with rc_eq ? the message would indicate that its NULL which is not the default [13:09] <saste> michaelni, can you create a coverity account for me? [13:10] <michaelni> saste, i think i created one days ago, can you check your spam folder ? [13:15] <saste> michaelni, found none [13:18] <michaelni> saste, created another one [13:19] <saste> thanks [13:22] <saste> michaelni: should I expect a mail or so? [13:22] <michaelni> saste, yes but i do not know how quick [13:22] <saste> ok [13:23] <michaelni> if you got nothing till tomorrow evening then ping me and provide a different email address than the gmail one maybe [13:38] <saste> in some cases the spotted issues are bogus [13:38] <saste> what should we do in that case? [13:39] <saste> for xfaceenc the fix is trivial, but not required [13:39] <saste> and adds an unnecessary check (so it slows it down, even if in an unnoticeable manner) [13:40] <saste> [NULL @ 0x2158100] [13:41] <saste> why do we print the funny hexdigit? [13:41] <saste> is it somehow useful, or should we replace it with something more useful (e.g. the called function name?) [13:45] <michaelni> saste, you can mark CIDs as false positive or intentional [13:45] Action: michaelni misses cone [13:47] <michaelni> saste, the idea behind the address for av_log is that if you have 2 abccontext to be able to distinguish them [14:41] <j-b> michaelni: cone seems to be restarted [14:47] <cone-618> ffmpeg.git 03Luca Barbato 0771f7b22dba60: ffv1: split decoder and encoder * 03http://tinyurl.com/9qgeb3603 [14:47] <cone-618> ffmpeg.git 03Michael Niedermayer 0769fd0b7adbd2: Merge commit '71f7b22dba60524b2285643ae0b49d8f64977129' * 03http://tinyurl.com/922yrqz03 [14:47] <michaelni> cone :) [14:48] <ubitux> still no plan to remove that tinyurl thing? [15:02] <cone-618> ffmpeg.git 03Luca Barbato 074a2a4524a3f5: ffv1: propagate errors * 03http://tinyurl.com/9zlfaln03 [15:02] <cone-618> ffmpeg.git 03Michael Niedermayer 0799ea47fe5a55: Merge commit '4a2a4524a3f50ed302820ba971ddd48e78c7436f' * 03http://tinyurl.com/9clddju03 [16:19] <cone-618> ffmpeg.git 03Luca Barbato 070f13cd318719: ffv1: update to ffv1 version 3 * 03http://tinyurl.com/8rodexv03 [16:19] <cone-618> ffmpeg.git 03Mans Rullgard 07ebe46b8063ae: ARM: reinstate optimised intmath.h * 03http://tinyurl.com/8ber35h03 [16:19] <cone-618> ffmpeg.git 03Derek Buitenhuis 07f2a7236d0c73: doc/platform: Move a caveat down to the notes section * 03http://tinyurl.com/9ef5zg803 [16:19] <cone-618> ffmpeg.git 03Derek Buitenhuis 072d09b36c0379: doc/platform: Add info on shared builds with MSVC * 03http://tinyurl.com/8s5z93e03 [16:19] <cone-618> ffmpeg.git 03Michael Niedermayer 07aa760b173590: Merge commit '2d09b36c0379fcda8f984bc8ad8816c8326fd7bd' * 03http://tinyurl.com/8fa4ort03 [16:36] <ubitux> $(call ALLYES, FFPROBE LAVFI_INDEV SELECT_FILTER AVCODEC MOV_DEMUXER SVQ3_DECODER ZLIB) [16:37] <ubitux> nice dependencies. [16:37] <ubitux> :) [16:47] <ubitux> michaelni: i'll commit soon the lavfi metadata injection with fate tests for scene & silence detection, unless you want me to wait more [17:04] <cone-618> ffmpeg.git 03Diego Biurrun 076cfca5b6ae5a: ffv1: Add missing #includes to header file * 03http://tinyurl.com/8kn9fhp03 [17:04] <cone-618> ffmpeg.git 03Diego Biurrun 07886087829157: fate: Introduce ENCMUX macro for tests that require encoders and a muxer * 03http://tinyurl.com/8sduptg03 [17:04] <cone-618> ffmpeg.git 03Diego Biurrun 07a7d2861d3675: svq3: K&R formatting cosmetics * 03http://tinyurl.com/8arttqk03 [17:04] <cone-618> ffmpeg.git 03Michael Niedermayer 070e097616865e: Merge commit 'a7d2861d36756b913e85681b86ed3385274e8ced' * 03http://tinyurl.com/92ne85o03 [17:05] <ubitux> https://github.com/ubitux/FFmpeg/compare/master...lavfi-metadata [17:15] <cone-618> ffmpeg.git 03Diego Biurrun 07af1ede069cb6: svq3: cosmetics: Drop useless parentheses * 03http://tinyurl.com/9njxlrz03 [17:15] <cone-618> ffmpeg.git 03Diego Biurrun 072e0c410485ef: fate: adpcm: Add dependencies * 03http://tinyurl.com/8dsklj803 [17:15] <cone-618> ffmpeg.git 03Diego Biurrun 07620345f930d1: fate: adpcm: cosmetics: Sort test entries * 03http://tinyurl.com/995aqz803 [17:15] <cone-618> ffmpeg.git 03Michael Niedermayer 07d7b8a9a589d9: Merge remote-tracking branch 'qatar/master' * 03http://tinyurl.com/9ptywg503 [17:15] <ubitux> meh i need to rebase again [17:16] <michaelni> ubitux, make sure you dont access the new fields directly from outside avcodec [17:16] <michaelni> i think i saw at least one acces from lavfi [17:17] <ubitux> huh, really? [17:17] <michaelni> AVFrame.metadata i think [17:18] <ubitux> oh! [17:18] <ubitux> ok, will fix [17:18] <ubitux> thank you [17:21] <ubitux> fixed in avfilter_copy_frame_props() [17:49] <ubitux> wtf i just -c copy and the h264 input got transformed into rawvideo.. [17:50] <ubitux> oh my bad. [17:56] <cone-618> ffmpeg.git 03Clément BSsch 076fb2fd895e85: lavc: add lavfi metadata support. * 03http://tinyurl.com/8bosxvf03 [17:56] <cone-618> ffmpeg.git 03Clément BSsch 07de23953de2d8: lavfi/select: store scene score in buf ref metadata. * 03http://tinyurl.com/9b3v8l203 [17:56] <cone-618> ffmpeg.git 03Clément BSsch 07fbedce6b8512: lavfi/silencedetect: export silence info to metadata. * 03http://tinyurl.com/9qwcoex03 [18:08] <cone-618> ffmpeg.git 03Michael Niedermayer 0738797dc31b65: compat/getopt: add {} to complex ifs * 03http://tinyurl.com/94ugnq903 [18:08] <cone-618> ffmpeg.git 03Michael Niedermayer 074ce9312d7617: libavcodec/cook: add {} to complex ifs * 03http://tinyurl.com/9ygau6d03 [18:08] <cone-618> ffmpeg.git 03Michael Niedermayer 07f72b735d41f9: libavcodec/vorbisenc: add {} to complex ifs * 03http://tinyurl.com/95pqdv803 [18:09] <cone-618> ffmpeg.git 03Michael Niedermayer 07217193facd3e: bavformat/mov: add {} to complex ifs * 03http://tinyurl.com/8lkucus03 [18:09] <cone-618> ffmpeg.git 03Michael Niedermayer 07fefe9bd78266: libavformat/movenc: add {} to complex ifs * 03http://tinyurl.com/9j2bq9903 [18:09] <cone-618> ffmpeg.git 03Michael Niedermayer 07b8a64d69a927: libavformat/mpegvideodec: add {} to complex ifs * 03http://tinyurl.com/8wbjlqw03 [18:09] <cone-618> ffmpeg.git 03Michael Niedermayer 07c2c066784f19: libavformat/mxfenc: add {} to complex ifs * 03http://tinyurl.com/94gbsnq03 [18:09] <cone-618> ffmpeg.git 03Michael Niedermayer 0740ceb6d49f19: libavformat/oggenc: add {} to complex ifs * 03http://tinyurl.com/8rsyoxt03 [18:12] <ubitux> michaelni: is there some warnings with some compiler? [18:26] <cone-618> ffmpeg.git 03Marton Balint 07d6e95669496a: h264: add support for AFD detection * 03http://tinyurl.com/8scfs9h03 [18:31] <michaelni> ubitux, i dont know but that kind of code is just fragile, the develoer sees the association based on indention but the compiler doesnt care about the indention at all [18:33] <michaelni> its a recipe for bugs [18:33] <ubitux> of course [18:33] <ubitux> i was just wondering how you spotted them all [18:37] <michaelni> i dont know if i spotted all, it was just an ugly grep command that i used [18:42] <Compn> michaelni : you should put it in patcheck :P [18:42] <Compn> if possible [18:42] <Compn> and all other sneaky code things [18:42] <Compn> does libav use patcheck ? [18:43] <michaelni> patcheck works on diffs, i greped *.c so its not going to work [18:44] <Compn> for new code i mean [18:44] <Compn> but ok [18:45] <saste> michaelni, ping on "propagate or return meaningful error codes in avcodec_open2()" [18:54] <saste> really, we should properly document all the quantization stuff (a dedicated libavcodec manual would fit) [19:02] <cone-618> ffmpeg.git 03Marton Balint 07da569556217f: ffplay: use framedrop by default when sync is not done to video * 03http://tinyurl.com/9ukuag203 [19:02] <cone-618> ffmpeg.git 03Marton Balint 07eaa91ed863a7: ffplay: fix nosync threshold check in synchronize_audio * 03http://tinyurl.com/8dyzujs03 [19:02] <cone-618> ffmpeg.git 03Marton Balint 0766bb5b1bc9b1: ffplay: initialize audio and video pts drift * 03http://tinyurl.com/8ebxfra03 [19:02] <cone-618> ffmpeg.git 03Marton Balint 0777bd595ad2ff: ffplay: fix external time sync mode * 03http://tinyurl.com/8wb66wm03 [19:02] <cone-618> ffmpeg.git 03Marton Balint 072a4c7e6540c4: ffplay: add serial field to PacketQueue entry and populate it * 03http://tinyurl.com/96rp3a503 [19:02] <cone-618> ffmpeg.git 03Marton Balint 07b2a8850969b8: ffplay: only check external clock if current frame serial matches the displayed frame serial * 03http://tinyurl.com/8smuwbz03 [19:02] <cone-618> ffmpeg.git 03Marton Balint 07fca16a15712b: ffplay: add get_master_sync_type function * 03Error03 [19:02] <cone-618> ffmpeg.git 03Marton Balint 07d30c69251f04: ffplay: use get_master_sync_type where necessary * 03Error03 [19:02] <cone-618> ffmpeg.git 03Marton Balint 073166a6fc3797: ffplay: if there is no audio stream, use external clock by default * 03Error03 [19:02] <cone-618> ffmpeg.git 03Michael Niedermayer 07747a00b688d7: Merge remote-tracking branch 'cus/stable' * 03Error03 [19:20] <saste> ubitux: http://fate.ffmpeg.org/report.cgi?time=20121021160939&slot=x86_32-msvc10-wi… [19:20] <saste> damned windows path [19:21] <saste> but it raises an unconfortable issue [19:24] <ubitux> oh :( [19:24] <ubitux> saste: any idea how i should fix it? [19:24] <saste> automatic escaping [19:24] <saste> that's why i implemented tools/ffescape [19:25] <saste> but then we need to wrap the test in a script, which performs automatic escaping [19:25] <saste> that's why the graph syntax sucks hard, compared to avisynth [19:25] <saste> since requires up to two levels of escaping, three considering shell escaping [19:25] <saste> most users get it wrong [19:26] <saste> ubitux: also BSD is failing [19:27] <saste> http://fate.ffmpeg.org/report.cgi?time=20121021163915&slot=x86_64-freebsd8.… [19:27] <saste> though I have no idea why [19:27] <ubitux> -video|1|14100|23.500000|14100|23.500000|1|0.001667|2006431|320|240|rgb24|1:1|I|0|0|0|0|0|0|0.850000 +video|1|14100|23.500000|14100|23.500000|1|0.001667|2006431|320|240|rgb24|1:1|I|0|0|0|0|0|0|0.840000 [19:27] <ubitux> looks like yet another floating thing [19:28] <saste> what is the last value? [19:28] <ubitux> the scene score [19:28] <saste> ah ok [19:29] <ubitux> saste: so how/where would you implement the automatic escaping? [19:30] <saste> ubitux, I have a patch pending [19:30] <ubitux> ah, great :) [19:30] <saste> i want to fix a few things before committing i [19:30] <saste> *it [19:31] <saste> but then deploying it in the test system may be complicated [19:31] <cone-618> ffmpeg.git 03Carl Eugen Hoyos 07fa190b3cd736: Improve MPEG-PS-in-MOV detection. * 03http://tinyurl.com/8hqxwdl03 [19:32] <ubitux> i wonder how i'm supposed to fix the freebsd one :( [19:32] <saste> avoiding float computations? [19:33] <ubitux> i want the value in the [0;1] range [19:34] <saste> ubitux: you may add an option where you specify the value in integer units [19:35] <saste> and return 0-100 integer values [19:35] <saste> maybe someone has a better idea (this may complicate the internal code) [19:47] <michaelni> ubitux, distclean + fate-filter-metadata-scenedetect fails with no ffprobe having being build [19:47] <michaelni> the 0.84/0.85 can be reproduced with --disable-asm --disable-yasm on linux [19:47] <michaelni> make sure swscale gets its bitexact flags! [19:48] <ubitux> i added -bitexact [19:48] <ubitux> mmh CONFIG_FFPROBE isn't enough, i will add the ffprobe dep [19:48] <ubitux> thanks [19:48] <ubitux> i'll fix all of this asap [19:48] <michaelni> try -sws_flags +accurate_rnd+bitexact if its not there yet [19:50] <ubitux> mmh ok [20:06] <cone-618> ffmpeg.git 03Clément BSsch 07c0d56bf8a4c6: fate: fix filter metadata dependency to ffprobe. * 03http://tinyurl.com/8f39mdf03 [20:13] <ubitux> the sws flags don't seem to help :/ [20:17] <cone-618> ffmpeg.git 03Clément BSsch 07e168165489fe: doc/muxers: document mov faststart option. * 03http://tinyurl.com/9okq7vw03 [20:51] <michaelni> ubitux, is the input to the filter the same and the output score differs or the input isnt the same in the first place ? [20:52] <ubitux> i'm going to check in a moment [20:53] <ubitux> i was just fooling around with something right now :p [20:53] <ubitux> saste: just added the sliding mode to showspectrum :P [20:53] <saste> nice [20:56] <ubitux> just sent [20:56] <ubitux> now serious matter, fate. [21:06] <cone-618> ffmpeg.git 03Michael Niedermayer 07571309181859: mpegaudiodec: Fix buffer handling on random access * 03http://tinyurl.com/8gwvuxw03 [21:12] <cone-618> ffmpeg.git 03Stefano Sabatini 078c2dbc380500: lavc/utils: provide more feedback in case of experimental codec * 03http://tinyurl.com/8qrq7vo03 [21:13] <ubitux> michaelni: the input isn't the same in the first place indeed, maybe the flag are not properly transmited to swscale [21:13] <ubitux> most likely a stupid mistake from me [21:14] Action: Compn waits for autocrop filter [21:14] <Compn> :P [21:15] Action: Compn afk [21:18] <ubitux> michaelni: any simple check to see if the flags reach sws? [21:23] <michaelni> ubitux, print the flags from sws_scale() [21:34] <ubitux> ok they don't seem to be transmited [21:38] <ubitux> scale_args=[0:0:(null)] [21:38] <ubitux> mmh. [21:39] <ubitux> wasn't this fixed? [21:39] <saste> ubitux, is it a problem? [21:39] <ubitux> ok it's done in ffmpeg & ffplay. [21:39] <ubitux> but not ffprobe [21:39] <ubitux> saste: i see at least two problems :p [21:40] <saste> i mean with your fate test [21:40] <ubitux> ah yes it's a problem [21:40] <ubitux> the bitexact flags for sws are not honored [21:40] <ubitux> and so the input differs [21:40] <ubitux> (since select is converting to rgb when using scene detection) [21:41] <saste> ok i'll commit them soon [21:41] <saste> i just want to test them another time [21:41] <ubitux> ah you have a fix for that? i wasn't inventing thing? [21:50] <ubitux> well i guess i'll wait for your patches then [21:55] <saste> ubitux: 6fb2fd895e858ab93f46e656a322778ee181c307 [21:55] <saste> wow, longest commit message in FFmpeg history [21:55] <ubitux> :) [21:56] <saste> btw fate is failing on your last test [21:56] <ubitux> you mean the scene detect? [21:56] <saste> yes [21:56] <ubitux> yes that's what we were talking about [21:57] <saste> amovie=/amrwb/seed-12k65.awb,silencedetect=d=.1: No such file or directory [21:57] <ubitux> huh? [21:58] <saste> i believe it is because I'm not using SAMPLES [21:58] <ubitux> ah i use --samples at configure time [21:58] <ubitux> mmh i guess i'll need to add them to some kind of samples dep [22:00] <ubitux> anyway, for the scene detect, there are multiple things to do [22:00] <ubitux> first make sure ffprobe is transmitting the sws flags, and then add the bitexact flags to the FILTER_METADATA_COMMAND [22:00] <ubitux> but you told me you had some pending patches for this [22:01] <ubitux> about the SAMPLES issue, i don't know how i'm supposed to fix that properly.. [22:04] <ubitux> saste: the sws flags thing looks a bit messy and inconsistent [22:05] <ubitux> it's only set in ffplay and ffmpeg, but not in ffprobe; and maybe it should be set in the lavfi device? [22:12] <saste> ubitux, yes [22:12] <michaelni> btw if you cant fix this, the not working test should be disabled [22:13] <Daemon404> didable broken things to make stuff appear to pass isnt a very good idea... [22:13] <Daemon404> disabling* [22:14] <ubitux> michaelni: i'll fix as soon as saste push what he was talking about [22:14] <ubitux> it can be fixed without much trouble i believe [22:15] <ubitux> we just need to pass the bitexact flags properly to sws :p [22:20] <saste> ubitux: can you fix the samples thing in the meanwhile [22:20] <ubitux> yes, trying to figure out how to :p [22:23] <michaelni> Daemon404, i think we speak about different things [22:26] <michaelni> if code is commited that breaks something it should be fixed or reverted, ubitux is working on that. But if noone had time, the code that broke something should be reverted [22:26] <nevcairiel> isnt it new code that just doesnt work everywhere? [22:26] <nevcairiel> its not a regression, just open TODOs, basically =) [22:27] <ubitux> michaelni: what's my deadline for fixing fate? [22:27] <michaelni> its causing a regression for "make fate" [22:27] <nevcairiel> a regression is existing tests breaking, not new tests not passing, imho [22:28] <nevcairiel> anyhow fixes seem to be on their way [22:28] <michaelni> ubitux, no idea, it doesnt matter, just if you give up / work on something else please disable it first [22:28] <ubitux> ok [22:28] <ubitux> i can go to sleep tonight without fixing it? :) [22:29] <michaelni> if you continue dreaming about how to solve it :) [22:29] <michaelni> nevcairiel, its a regression in "make fate" [22:29] <michaelni> it worked before [22:30] <ubitux> of course ;) [22:31] <michaelni> "make fate" is used to test if code is ok, for example before push, if it fails developers work is hindered [22:33] <michaelni> thats IMHO not so much about "hiding a problem" more "not having all alarm bells ringing for days while its being worked on" [22:35] <michaelni> and these alarm bells could hide other new problems or regressions [22:37] <Daemon404> michaelni, i thought you were talkign about a fate test [22:37] <Daemon404> by bad [22:37] <nevcairiel> considering its only broken on one os/toolchain combination, i think we'll survive a day until its fixed [22:39] <nevcairiel> i think its not even the change itself, just the way the test is setup with the movie source, yes? [22:40] <michaelni> i think its broken on all non x86 platforms [22:41] <nevcairiel> fate looks rather normal [22:42] <nevcairiel> except the fail on native windows builds which is new [22:45] <michaelni> many didnt test the problematic revission yet i think [22:45] <cone-618> ffmpeg.git 03Stefano Sabatini 072969abd908f1: lavfi/graphparser: fix parsing error in case of NULL sws_opts addition * 03http://tinyurl.com/8tk9gkf03 [22:45] <cone-618> ffmpeg.git 03Stefano Sabatini 078f37a1e8dc93: lavfi/avfiltergraph: avoid to print "(null)" in the scale args * 03http://tinyurl.com/8otm4q403 [22:46] <nevcairiel> didnt it just uncover a parsing problem when a string parameter includes a :, which happens in windows path strings? [22:46] <ubitux> saste: thx! [22:46] <nevcairiel> are we talking about the same problem? :P [22:46] <saste> nevcairiel, you need to escape it [22:46] <saste> the problem is doing it in an automated fashion [22:47] <nevcairiel> thats rather unintuitive though :p [22:47] <nevcairiel> how do you escape in that place even? normal backslash? [22:56] <ubitux> ok almost fixed it [23:00] <ubitux> ok, fixed [23:02] <ubitux> i wonder how safe this is though [23:03] <ubitux> saste: http://b.pkh.me/0001-lavd-lavfi-honor-sws-flags.patch i guess that needs to be made safer, right? [23:04] <ubitux> (with some #if CONFIG_...) [23:10] <michaelni> ubitux, you are accessing global variables that reside in ff* user tools from the libs [23:11] <michaelni> that is definitly not safe [23:12] <ubitux> mmh it's in the cmdutils, missed that [23:14] <ubitux> i guess i'll have to do that in ffprobe [23:15] <ubitux> but i don't have access to the filtergraph.. [23:16] <michaelni> quick solution is to add a AVOption to lavfi that allows setting the sws flags [23:16] <michaelni> libavdevice/lavfi.c that is [23:16] <ubitux> what about getting a new sws context in the lavfi device? [23:17] <ubitux> mmh wait i'll need the link with the options [23:19] <michaelni> another way is to make avfilter_graph_parse() somehow set the "global" flags from the graph string [23:22] <ubitux> my brain is dead, i'll see that tomorrow [23:22] <michaelni> ok no hurry, maybe ill look at it while you sleep [23:24] <ubitux> you're a real guardian ;) [23:24] <ubitux> 'night :) [23:25] <ubitux> this is a test case btw: http://b.pkh.me/0002-fate-add-bitexact-flags-to-filter-metadata-test-comm.p… [23:25] <ubitux> and i have another patch to add the scale filter dep to scene detect as well [23:26] <ubitux> anyway, cya [23:26] <michaelni> good night ubitux [23:31] <Compn> i thought boxx codec was dead, whats it doing with 10bit ? [23:41] <saste> since when "being experimental" is an error? [23:41] <saste> what about AVERROR_PINK? [00:00] --- Mon Oct 22 2012

1 0

ffmpeg.log.20121021
by burek 22 Oct '12

22 Oct '12

[02:39] <ZeloZelos> what would be the paramaters to take png's and turn them into wmv? [02:40] <ZeloZelos> i tried ffmpeg -i "C:\Users\Zelo\Desktop\earth\frames\*.png" -vcodec wmv2 -y C:\Users\Zelo\Desktop\earth.wmv [07:25] <robertzaccour> I'm trying to figure out how to include audio recording in screencasts with ffmpeg. [07:26] <robertzaccour> so far I'm at ffmpeg -f x11grab -r 25 -s 1024x768 -i :0.0 -vcodec huffyuv -sameq screencast.avi now just trying to figure out how to get it to record my voice mic and whatever audio is on the pc [07:47] <robertzaccour> Ok I almost got it figured out. I've got x11grab working and i-pulse. Might someone be able to help me figure out how to capture analog audio as well? [07:49] <hendry> robertzaccour: https://github.com/kaihendry/recordmydesktop2.0/blob/master/r2d2.sh#L26 [07:50] <hendry> robertzaccour: i use alsa, I don't understand how/why pulse is used by so many ppl [07:51] <robertzaccour> hendry, pulse works best for me, especially when using guvcview [07:52] <robertzaccour> hendry, [NULL @ 0x265e440] Requested output format 'x11grab' is not a suitable output format [07:52] <robertzaccour> :0.0: Invalid argument [07:58] <robertzaccour> Could someone help me figure out how to record from my speakers? I already got my screen and mic with ffmpeg -f alsa -i pulse -f x11grab -r 25 -s 1440x900 -i :0.0 -acodec pcm_s16le -vcodec huffyuv -sameq Screencast.avi just gotta get my speakers in there and I'm set. [08:03] <fling> how may I use something like Deshaker with ffmpeg? > http://www.guthspot.se/video/deshaker.htm [08:11] <robertzaccour> Is there anything in my question I'm leaving out that might be helpful in figuring out? [08:12] <hendry> robertzaccour: can't you control that sort of mixing from alsamixer ? [08:12] <hendry> robertzaccour: sorry, I would love to help, but I am an newbie [08:13] <hendry> robertzaccour: when you figure out how to do it, I would like to incorporate it to https://github.com/kaihendry/recordmydesktop2.0 [08:45] <fling> can ffmpeg do this with deshake? > http://isenmann.wordpress.com/2011/03/22/deshaking-videos-with-linux/ [08:52] <robertzaccour> Is it even possible to record from the mic and the speakers at the same time? Could someone please help me with this? [08:53] <fling> robertzaccour: /j #alsa [08:56] <robertzaccour> fling, this is the ffmpeg channel [09:13] <hendry> is there a sane way to figure out the duration of a .webm file ? [09:20] <robertzaccour> hendry, right-click, select properties, look in the audio/video tab [09:20] <robertzaccour> or double click and see how long it is [09:27] <robertzaccour> dadadew, you familiar with ffmpeg commands? [09:28] <dadadew> absolutely [09:30] <hendry> robertzaccour: i'm on linux [09:31] <robertzaccour> dadadew, here's what I got so far so ffmpeg -f alsa -ac 2 -i plughw:1,0 -f x11grab -r 15 -s 1360x768 -i :0.0 -acodec pcm_s16le -vcodec libx264 -preset ultrafast -threads 0 output.mkv now I just need to figure out how to get my analog speakers in on the recording [09:37] <ubitux> fling: we have a deshake filter, but it might be that effective [09:38] <ubitux> fling: your post reminds me the thing from google [09:38] <ubitux> there is an opened issue to improve the deshake filter, if you want to contribute :) [09:39] <ubitux> hendry: use ffprobe [09:39] <ubitux> with -show_format, and you might be interested in -of as well [09:43] <hendry> thank you ubitux [09:44] <robertzaccour> anyone here familiar with ffmpeg commands? [09:44] <ubitux> you likely need to configure it with alsa [09:44] <robertzaccour> ya think so? [09:45] <hendry> ubitux: ffprobe man page is mental. what does -of do [09:45] <ubitux> dunno, but maybe something like making your speakers in recording mode [09:45] <ubitux> hendry: try -of json, or -of flat=s=_ [09:45] <ubitux> you'll get the point [09:48] <hendry> ubitux: got it. that ffprobe man page needs a sanity check nontheless ;) [09:49] <ubitux> really? [09:49] <ubitux> http://ffmpeg.org/ffprobe.html#Writers [09:49] <ubitux> and the -of description looks pretty obvious [09:50] <ubitux> you even have an example :) [09:50] <robertzaccour> ubitux, I'm tryin to figure out how to get ffmpeg to record the speakers directly, just gotta figure out the input and where to put it [09:51] <ubitux> robertzaccour: i understand, but to my knowledge, everytime i had to record from the speaker, it needed some configuration on alsa side as well [09:51] <ubitux> and everytime, it was a pain anyway [09:51] <robertzaccour> ubitux, you mean with alsamixer? [09:51] <ubitux> for example [09:53] <hendry> ubitux: http://r2d2.webconverger.org/2012-10-21/ubitux.html [09:54] <ubitux> haha [09:55] <ubitux> now you could use -of json, and print the info in javascript [09:56] <hendry> ubitux: yes, bit too tired to that atm, but I understand [09:56] <ubitux> and btw, since you're using shell scripting, -of flat=s=_ makes possible to eval directly the variables [09:57] <hendry> ubitux: i'll make a note of that [09:57] <hendry> ubitux: i wish the output wasn't so lossy, can't clearly see my desktop [09:57] <ubitux> so $(eval ffprobe -v 0 -of flat=s=_ -show_format ...) and then you can access for instance $format_duration [09:59] <ubitux> looks readable to me [09:59] <ubitux> but you can play with -b:v to get a better quality [10:00] <hendry> ubitux: on this line ? https://github.com/kaihendry/recordmydesktop2.0/blob/master/r2d2.sh#L34 [10:00] <hendry> ubitux: what's an example value ? [10:01] <ubitux> i'd say on line 36 [10:01] <ubitux> i don't understand why you have so much ffmpeg cmd lines [10:01] <ubitux> why is audio & video separated? [10:01] <ubitux> example value? like -b:v 2000k ? [10:03] <hendry> first i'm recording in flac because i'm told that's fast [10:03] <hendry> ubitux: then i need to convert to vorbis as webm only accepts that as an input [10:04] <hendry> ubitux: so there is 3 steps :( [10:04] <ubitux> you can replace the two last one with a single step afaict [10:05] <ubitux> like... ffmpeg -i $temp $out [10:05] <ubitux> codecs will be autopicked [10:05] <ubitux> you could explicit them with ffmpeg -i $temp -c:a libvorbis -c:v libvpx $out if you prefer [10:06] <ubitux> then play with the quality, like ffmpeg -i $temp -c:a libvorbis -q:a 7 -c:v libvpx -b:v 3000k $out [10:06] <hendry> not sure what 2000k and 3000k mean [10:06] <ubitux> bitrate [10:06] <hendry> and -q:a 7 ? [10:07] <ubitux> audio quality [10:07] <ubitux> variable bitrate, likely [10:07] <hendry> ok, i'll play. spent many hours on this now. :} my exp is that the audio had to be in vorbis in advance [10:08] <ubitux> why? [10:08] <hendry> ubitux: isn't there just a *lossless* codec, or is webm inheriently lossy? [10:08] <ubitux> not sure if vpx has a lossless mode [10:08] <ubitux> and afaik webm only allows vp8 [10:14] <hendry> where can i figure out if vp8 has a lossless mode? [10:44] <hendry> ubitux: two lines works btw! http://r2d2.webconverger.org/2012-10-21/ubitux.html [10:46] <ubitux> i'm on the internet @_@ [10:49] <hendry> ubitux: file sizes seem a bit big. i'll have to play with -b:v [10:49] <ubitux> yep [10:52] <hendry> ubitux: any other possibilities to improve it? I need a H264 version too, need to figure out that line from $temp [10:55] <ubitux> in a mp4? afaict libx264 has better defaults anyway [10:55] <ubitux> i'd say to just play with -crf option [11:05] <hendry> ubitux: https://gist.github.com/3926442 ffmpeg -i /tmp/./r2d2.sh.8Sp5.mkv 2012-10-21/forIOSsafari.mp4 # simply doesn't work I guess I need to manually tell it to convert ? [11:06] <ubitux> [NULL @ 0x19efec0] Codec is experimental but experimental codecs are not enabled, try -strict -2 [11:06] <ubitux> so add -strict -2, or use an external aac encoder [11:06] <ubitux> like fdk-aac, libfaac, etc [11:08] <hendry> ubitux: ffmpeg -i /tmp/./r2d2.sh.8Sp5.mkv -crf 0 -c:v libx264 -preset slow -c:a libmp3lame 2012-10-21/forIOSsafari.mp4 # works ! [11:08] <ubitux> yes, you can encode in mp3 as well [11:08] <hendry> not sure what "-preset slow -c:a libmp3lame 2012-10-21/forIOSsafari.mp4 # works ! [11:08] <hendry> not sure what "-preset slow" does [11:09] <ubitux> it makes it slow? :) [11:09] <ubitux> anyway, you'll likely want -q:a 3 or something (or reduce even more to get a better quality) [11:09] <ubitux> crf 0 is a bit low OTOH [11:10] <ubitux> -crf 21 should be more than enough [11:10] <ubitux> 15-25 should be the range you should play with [11:10] <ubitux> look at libx264 for the appropriate settings [11:11] <hendry> quantizer scale is 0-51; where 0 is lossless [11:11] <hendry> i want lossless don't I? [11:11] Action: hendry reads http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide [11:12] <ubitux> ah if you want some lossless... [11:12] <ubitux> i'd use another codec for lossless video though [11:12] <hendry> i want the desktop I'm capturing to be sharp [11:13] <hendry> ubitux: surely with mp4 I'm limited to h264 ? require playing on IOS6's safari [11:13] <ubitux> i guess so [11:14] <hendry> mp4 seems 1/3 of the size of the webm regardless! [11:14] <ubitux> though you can likely put another codec if you want, indeed the issue is your destination device [11:14] <ubitux> yes, you have to play more with the libvpx bitrate and maybe other settings [11:21] <hendry> damn, doesn't work on my iphone http://r2d2.webconverger.org/2012-10-21/test.html [11:31] <ubitux> hendry: it's possible some apple device are limited to a few h264 levels [11:32] <hendry> is crf the level? [11:33] <ubitux> no [11:34] <hendry> hmm, doesn't work in Android either [11:34] <ubitux> https://en.wikipedia.org/wiki/H.264/MPEG-4_AVC#Levels [11:34] <ubitux> look at the levels supported by your devices [11:35] <ubitux> and then use -level [11:35] <hendry> oh no, there is depressing http://stackoverflow.com/questions/9144574/no-matter-what-i-do-i-cannot-get… [11:35] <ubitux> :) [11:55] <hendry> none of these presets work for me -vpre, maybe there are in a seperate ffmpeg distro? [11:56] <ubitux> try -preset:v [11:57] <ubitux> these options are a bit tricky [12:08] <Element9> which filters do I need in order to be able to capture audio and video to file? [12:09] <Element9> I disabled all filters when I compiled ffmpeg and I'm getting: "'aresample' filter not present, cannot convert audio formats." [12:10] <Element9> anything else? ...before I just go and do the possible long compile-test-repeat cicle :) [12:17] <hendry> can't get audio working on safari ubitux, video works though [12:17] <hendry> this (flac -> aac) is to blame [12:18] <ubitux> try another aac encoder [12:18] <ubitux> Element9: are you using ffmpeg 1.0? [12:18] <Element9> ubitux: yes [12:18] <hendry> Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height [12:18] <Element9> ubitux: should i be using 1.0, or is it too early? [12:19] <hendry> i fell like i just need the right options [12:19] <ubitux> why did you disable filters? [12:19] <ubitux> it's likely you'll need the resample and scale filters for most of the op [12:19] <hendry> ubitux: what other aac encoder ? [12:19] <Element9> ubitux: to get the smallest executable possible [12:19] <ubitux> hendry: fdk-aac, libfaac, etc [12:20] <hendry> Unknown encoder 'fdk-aac' [12:21] <Element9> ubitux: oh, it's not the released 1.0, it's clone from git [12:21] <ubitux> it's fine :p [12:21] <ubitux> hendry: it's an external one, you need to link it with [12:21] <ubitux> Element9: 1.0 or git/master is fine [12:22] <hendry> -c:a libfaac # failes too :/ [12:25] <ubitux> hendry: same problem. [12:28] <hendry> ubitux: going to try rebuild ffmpeg in desperation https://bugs.archlinux.org/task/27465 [12:29] <ubitux> --enable-libopencore_amrnb --enable-libopencore_amrwb [12:29] <ubitux> isn't this ok? [12:29] <ubitux> i though it was yet another aac encoder [12:34] <hendry> ubitux: yes... it works with re-compiled ffmpeg! http://r2d2.webconverger.org/2012-10-21/baseline.html [12:35] <ubitux> :) [12:44] <CryingRiver> hello, i need help :Ž( [12:45] <CryingRiver> if i try to use the "configure" command to build ffmpeg gcc says that he dont know what this command means, can someone help me maybe? [12:47] <Element9> CryingRiver: tryed "./configure" ? [12:47] <CryingRiver> yes [12:47] <Element9> s/tryed/tried [12:47] <Element9> which OS? [12:47] <CryingRiver> windows [12:47] <CryingRiver> 7 [12:48] <Element9> and which compiler do you plan to use? [12:49] <CryingRiver> that the output: http://pastebin.com/N9nPgc53 [12:49] <Element9> configure is a bash script [12:49] <Element9> for linux [12:50] <Element9> here are the guids for compiling ffpmeg: http://ffmpeg.org/trac/ffmpeg/wiki/CompilationGuide [12:51] <Element9> I'm compiling it in linux for usage on windows - haven't tried to compile it on windows [12:53] <CryingRiver> mhh... my real problem is that I need to build ffmpeg with a special gcc version, and I dont know how to do it... can I use mingw with a special gcc version? [12:56] <Element9> CryingRiver: did you read the compilation guide? [12:57] <CryingRiver> yes, but i got evrytime a "undefined reference to "av_register_all" error [12:58] <CryingRiver> if i build it with mingw [12:58] <CryingRiver> and use this build in my application [12:59] <Element9> and everything works if you use prebuilt one? [13:00] <Element9> no errors while compiling with mingw? [13:03] <CryingRiver> i dont tried a prebuilt one [13:06] <CryingRiver> Element9: no, it dont works, same error [13:06] <Element9> so it's not due to the custom build then :) [13:07] <CryingRiver> yes, but whats then the source of this error... :( [13:11] <Mavrik> you're not linking the libav dlls to your project [13:13] <CryingRiver> i link the .a files, how can i link the .dlls in the makefile? [13:44] <hendry> ubitux: do you know how to get "scrubbing" or seeking working in H264? [13:46] <ubitux> scrubbing? [13:46] <ubitux> seeking should just work [13:53] <hendry> ubitux: it doesn't :/ [13:53] <ubitux> you can't seek in the file with ffplay? [13:53] <hendry> ubitux: when i move the cursor on safari's media player thingy in Youtube, it says "hi speed scrubbing" [13:54] <ubitux> no idea [14:06] <Konigsberg> i dotnt get it, i install ffmpeg on windows on cygwin without using mingw [14:06] <Konigsberg> i cant remember how and i need to update it for youtube-dl and other things to work [14:06] <Konigsberg> i googled like a vicous man but no go [14:18] <Konigsberg> i also didnt have yasj [14:18] <Konigsberg> yasm [14:45] <Konigsberg> i know, i must delete the older ffmpeg version [14:45] <Konigsberg> before i make install the new one [14:45] <Konigsberg> for some reason it's in use [14:45] <Konigsberg> cant find it anywhere [14:56] <GRMrGecko> I'm trying to find a input method I found awhile back this year. I'm trying to convert a multi chapter DVD into one MP4. The method I saw was not a cat command and it was not http://ffmpeg.org/ffmpeg.html#join [14:56] <Element9> is vorbis encoder now marked as experimental and it previously wasn't? [14:56] <GRMrGecko> I cannot use a cat command because I'm already using the std input [14:57] <klaxa> Element9: i think you want libvorbis to use the external encoder, there is an internal one which is marked as experimental [14:57] <Element9> klaxa: thanks. that's what I suspected [15:00] <GRMrGecko> https://evilshit.wordpress.com/2012/09/13/how-to-combine-multiple-files-wit… [15:00] <GRMrGecko> that must have been it [15:04] <GRMrGecko> yup:D [15:06] <fling> ubitux: ok :] [15:17] <Element9> every time I stop the capature, windows thinks ffmpeg crashed and says "ffmpeg.exe has stopped working" [15:18] <Element9> is that a known problem? I haven't had that problem before [15:52] <fling> ubitux: the deshaker used for this video looks very good > http://www.youtube.com/watch?v=UBeBHshMsWI [15:56] <Sashmo> good day eveyone, can anyone help me figure out why I keep getting lip sync issues after 8+ hours of live transcoding? [16:01] <Sashmo> oke doke [16:02] <ubitux> fling: great :) [16:07] <Sashmo> fflogger: http://pastebin.com/ebjBAjwu [16:17] <klaxa> Sashmo: >-strict experimental [...] -acodec aac [16:17] <klaxa> i think that's not optimal, i was told to install fdk-aac if i wanted aac [16:18] <Sashmo> klaxa: ? yes? [16:18] <klaxa> you have to recomple from source though, that might fix it [16:18] <klaxa> i'm no expert though... i can't guarantee anything [16:22] <Sashmo> hmmm interesting [16:22] <Sashmo> I'll look into it. [16:23] <Sashmo> can anyone else recommend the same? [16:23] <JEEB> fdk is the best available AAC encoder atm, but just non-optimal AAC encoder shouldn't create a/v desynch [16:24] <Sashmo> does anyeont think it has something to do with timestamps? [16:26] <JEEB> well, yes if the input timestamps are borked then yes, but I would check if you are always fast enough first, although the fps=26 does show that you are indeed over the seemingly 25fps input stream [16:27] <JEEB> also I see you get decoding errors, do those just happen at the beginning and never after that? [16:28] <Sashmo> JEEB: yeah those are just at the begining [16:28] <Sashmo> JEEB: how can I check if the timestamps are borked up [16:29] <JEEB> no idea [16:29] <JEEB> also, can I slap you for using manual settings with libx264? [16:29] <JEEB> I'd really like you to slap you [16:30] <JEEB> -preset was made for a reason [16:30] <JEEB> (and presets within x264) [16:30] <Sashmo> slap me witha trout please [16:30] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset [16:30] <JEEB> you set those with -preset [16:30] <JEEB> (and then set profile/level) [16:33] <Sashmo> JEEB: but is that really effecting my audio issue? [16:42] <JEEB> Sashmo, nah -- I am just noting other things in that command line of yours [16:42] <Sashmo> cool, thanks [16:42] <JEEB> also if you're streaming you probably want to use maxrate and bufsize [16:43] <JEEB> maxrate being the maximum rate allowed (aka the minimum speed that lets a person watch the stream), and bufsize being the buffer over which that maximum rate is calculated [16:43] <Sashmo> JEEB: I'll play with it [16:43] <Sashmo> JEEB: any idea why VLC reports back that the stream is 50FPs? instead of 25? [16:43] <JEEB> but neither might be related to the a/v desynch. Which stream? [16:46] <Sashmo> that same one [16:46] <Sashmo> any of my streams really [16:46] <Sashmo> all report back double [16:52] <JEEB> Sashmo, I meant input or output :P [16:53] <Sashmo> JEEB: sorry I dont udnerstand? [16:53] <JEEB> your mpeg-ts input or the rtmp stream you create [16:54] <Sashmo> my source is mpegg-ts h2.64 25fps, and ffmpeg reports that its encoding at 26fps, but vlc reports it as 50fps [16:56] <JEEB> well, yadif shouldn't at least in your case be bobbing [16:56] <JEEB> in any case, not sure where the a/v desynch comes up, other than possible hick-ups in encoding [17:00] <Sashmo> JEEB: forget about the dsync termporaily, why am I getting this dumb 50fps on vlc? [17:01] <JEEB> no idea :P [17:01] <JEEB> VLC can fail, and something somewhere else could be failing as well [17:03] <Sashmo> all the players show it [17:07] <jelly1> hi is it normal that i get a 5 sec. lag between my webcam and the stream i get from ffserver [17:08] <jelly1> or could that be buffering? [17:15] <Sashmo> jelly1: are you using HLS? [17:15] <jelly1> Sashmo: what's HLS? [17:15] <jelly1> Sashmo: i could paste my ffserver.conf [17:16] <jelly1> i want to stream only video from my webcam to a website and use the other source for opencv [17:16] <Sashmo> jelly1: sorry cant help you there [17:16] <jelly1> Sashmo: ok :( [17:16] <jelly1> i'll continue searching [17:17] <jelly1> Sashmo: could it be the convertion? [17:17] <jelly1> from raw to in my case webm [17:17] <Sashmo> jelly1: I think the lowest I have ever seen it is down to 2 seconds [17:17] <jelly1> aha [17:18] <jelly1> Sashmo: or does ffserver itself buffers, and then cpu speed doesnt matter for encoding? [17:26] <Sashmo> jelly1: no idea, but I recall if you change the keyframe interval that you can get it to load faster, think of it like this, if you set a key frame interval of 60 frames, than the decoder will wait for frame 60 to output the video, therefor, 60 frames is 2 seconds, if you are encoding at 30fps, anyone else correct me here, I could be wrong, but this is what I was told before [17:27] <jelly1> aha [17:27] <jelly1> Sashmo: yes logical [17:27] <Sashmo> jelly1: what are you setting it to now? [17:27] <jelly1> Sashmo: FpS> [17:27] <jelly1> *fps? [17:27] <Sashmo> no key frame [17:28] <Sashmo> and sure fps [17:28] <jelly1> VideoFrameRate 24 [17:28] <Sashmo> and your keyframe? [17:28] <jelly1> isn't set [17:29] <Sashmo> hmm I dont know what the defalut is, try adding it [17:29] <Sashmo> keyint=48 [17:29] <Sashmo> should be for 2 seconds [17:29] <Sashmo> if your fps is 24 [17:29] <Sashmo> why 24? [17:30] <jelly1> Sashmo: oh i fetched this sample from a website [17:30] <Sashmo> try this, set your fps to 15, and see if that makes it any quicker [17:30] <Sashmo> dont add any ket int [17:30] <Sashmo> for a test [17:30] <jelly1> ok [17:30] <Sashmo> pastebin your command [17:30] <Sashmo> let me check it [17:30] <Sashmo> FYI, im no expert here [17:30] <Sashmo> just helping [17:30] <jelly1> ffmpeg -f video4linux2 -i /dev/video0 http://localhost:8090/webcam.ffm [17:31] <jelly1> hmm i should take notes [17:31] <Sashmo> just log your irc chat, best notes ever! [17:31] <jelly1> Sashmo: nah [17:31] <jelly1> Sashmo: for the delay :p [17:31] <Sashmo> use pastebin.com [17:31] <jelly1> sure [17:31] <Sashmo> paste the full command and output [17:32] <Sashmo> and the ffm config [17:32] <jelly1> Sashmo: http://sprunge.us/ACLC ffserver.conf and the command is ffmpeg -f video4linux2 -i /dev/video0 http://localhost:8090/webcam.ffm [17:33] <Sashmo> I've never used the ffm before, so I dont know what changes work here [17:33] <Sashmo> straight command line is what I have experiance with [17:33] <Sashmo> sorry [17:34] <jelly1> aha [17:34] <Sashmo> I guess its all the same though [17:34] <jelly1> there isn't much in the ffm [17:34] <Sashmo> I dont see much of a difference [17:34] <Sashmo> well your config is there [17:34] <jelly1> yeah ;) [17:34] <Sashmo> instead of the command line [17:34] <Sashmo> so it would be the same [17:35] <jelly1> hmm not sure what my stream plays now [17:35] <Sashmo> what are you playing it with? [17:35] <Sashmo> from the web page? [17:35] <jelly1> Sashmo: oh its just my webcam [17:35] <Sashmo> your probably using swf [17:35] <jelly1> now i will use fplay [17:35] <Sashmo> no I mean [17:35] <jelly1> Sashmo: no webm [17:35] <Sashmo> ah ok [17:36] <Sashmo> sorry dude, dont really know how to help you further [17:36] <Sashmo> ask the others [17:36] <jelly1> ok [17:36] <Sashmo> Im definitely not the guy to point you in the right direction, at least with this type of setup [17:36] <jelly1> ok [17:36] <jelly1> about 7 sec. with swf [17:37] <jelly1> Sashmo: hmm still lags about the same with VideoFrameRate 15 it seems [17:37] <Sashmo> try the keyint command [17:37] <jelly1> ok [17:38] <jelly1> Sashmo: which value? [17:38] <Sashmo> keyint [17:38] <Sashmo> add that [17:38] <Sashmo> and make it 1 [17:39] <Sashmo> see what happens [17:39] <jelly1> ok [17:39] <jelly1> [flv @ 0x7f39a0009860] warning: first frame is no keyframe [17:39] <jelly1> hmm that's what ffplay also says :) [17:39] <jelly1> Sashmo: looks the same [17:40] <jelly1> Sashmo: maybe reading this should shed some light http://ffmpeg.org/pipermail/ffserver-user/2011-March/000006.html [17:42] <jelly1> Sashmo: googling gives me some more hits [17:42] <jelly1> will study them later [17:42] <jelly1> Sashmo: thanks btw! [17:50] <Sashmo> jelly1: np [18:01] <_stclaws> I am republishing a live rtmp stream from a fms server. Works fine, but the incoming stream sometimes switches sources (we made it like that) which makes ffmpeg think it ends. Is there a way to make ffmpeg just keep on listening to incoming stream no matter what? [18:10] <louisdk> I want to edit start and stop time of some videos. It goes well if I don't use the "-t" option but when I do I get this error: http://pastebin.com/7Tq6KS7W [18:32] <_stclaws> I saw there are parameters to the rtmp protocol in ffmpeg, but how do I use them? For instance, if I want to do "ffmpeg -i rtmp//server/path/stream" and use the parameter "listen", what is the syntax? [19:42] <Element9> I've enbled wmv2 encoder when calling configure (--enable-encoder=wmv2), but resulting ffmpeg don't have wmv2 encoding support. is there anything else that I should enable? something that wmv2 depends on? [20:03] <iive> Element9: can you check config.h , look for the line with CONFIG_WMV2_ENCODER [20:04] <Element9> iive: I've run configure again with some other options and have a different problem now. I'll check that once I get it built again [20:04] <Element9> iive: I take it that config.h is generated during ./configure? [20:05] <iive> yes, config.h config.mak and config.asm I think. there must be a config.log too, in case you need more info about failed test. [20:06] <iive> have in mind that the order of configure options may matter. [20:06] Action: iive is not sure for the last one... but. [20:07] <Element9> iive: i have kind of put them in the order they appear in "./configure --help" [20:08] <iive> :) [20:08] <Element9> iive: don't know if I could do better than that... except to look in the code :) [20:09] <iive> well, i'm just saying that it may be good idea to put the options that disable all codecs first, and then put the options that enable specific codecs after. [20:11] <Element9> iive: oh yeah, sure. i did that :) [20:22] <sparkst> hello all - i've gotten awesome help from folks here in the past so I'm hoping someone can help me out with this issue today. here's the situation. I have 720p ProRes .movs that I am using ffmpeg to trim & transcode into H.264 .movs for YouTube upload. [20:22] <sparkst> the command I'm using is this: ffmpeg -i input.mov -ss 1201 -t 5000 -c:v libx264 -crf 18 -c:a libfaac -b:a 64k -ac 1 output.mov [20:22] <sparkst> (obviously the ss and t values change) [20:23] <sparkst> sometimes this works like a charm. other times, tho, youtube just rejects the file [20:23] <sparkst> often if I just rerun the same command it'll work fine the second time [20:23] <sparkst> any thoughts on what's going wrong here? [20:24] <sparkst> oh wait i miscopied the command [20:24] <sparkst> hang on [20:24] <sparkst> ffmpeg -i input.mov -pix_fmt yuv420p -c:v libx264 -crf 18 -c:a libfaac -b:a 64k -ac 1 -ss 1201 -t 5000 output.mov [20:29] <sparkst> ok i have to step away from my pc for a min, i will be back before too long tho. would appreciate any thoughts folks have. [20:51] <Element9> compiling ffmpeg from master for windows gives an executable which crashes on exit while release 1.0 works fine. does somebody, by any chance, know when was this regression introduced? :) [23:03] <bjrohan> Hey everyone. I am relatively new to Linux (running Kubuntu), and just installed ffmpeg, when I try to convert an ogv file from made with (using winff) desktop recorder (which plays fine) I get error messages no matter what I try, mov, mp4, avi [23:13] <sparkst-afk> hi folks - asked this earlier, but didn't get a response, forgive me for the repeition: I have 720p ProRes .movs that I am using ffmpeg to trim & transcode into H.264 .movs for YouTube upload. [23:13] <sparkst-afk> the command i'm using is this: ffmpeg -i input.mov -pix_fmt yuv420p -c:v libx264 -crf 18 -c:a libfaac -b:a 64k -ac 1 -ss 1201 -t 5000 output.mov [23:13] <sparkst-afk> (obviously the -ss and -t values change) [23:13] <sparkst-afk> sometimes this works like a charm. other times, tho, youtube just rejects the file [23:14] <sparkst-afk> and often when tha thappens, I can just rerun that command in ffmpeg and the resulting file works [23:14] <sparkst-afk> any thoughts on what's amiss here? [00:00] --- Mon Oct 22 2012

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