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February 2013
- 1 participants
- 56 discussions
[00:08] <cone-48> ffmpeg.git 03Michael Niedermayer 07master:29436c1ab2c5: libavcodec/mips/aacsbr_mips: fix ;;
[03:01] <cone-48> ffmpeg.git 03Michael Niedermayer 07master:b16bf913d8ed: dxva2: move _WIN32_WINNT handling to public header
[04:43] <tonsofpcs> just ran ffprobe -show_packets on a .ts and it is showing a dts = pts on all packets that don't have a dts. Why is this? (it's an ATSC TS if that matters)
[05:42] <llogan> i'm wondering if "qualification tasks" are an impediment to getting shit done for gsoc.
[06:59] <av500> by experience, a lot of "shit" has been done
[08:33] <highgod> Hi,Stefano Sabatini,I have re-submit the opencl patch
[11:01] <cone-583> ffmpeg.git 03Matt Wolenetz 07master:311443f6c7eb: x86: h264: Don't use redzone in AVX h264_deblock on Win64
[11:01] <cone-583> ffmpeg.git 03Michael Niedermayer 07master:db05f7ab1626: Merge remote-tracking branch 'qatar/master'
[13:18] Action: durandal_1707 looking at drawutils, and what to change for >8 bit support ; result = headache
[13:40] <Compn> (Remove the sws project, its mentor thinks it is pointless and a complete rewrite is needed instead.)
[13:40] <Compn> snort, rewrite of swscale :D
[13:44] <durandal_1707> hmm how to know what number of threads are appear when calling execute?
[13:48] <durandal_1707> i could allocate 16 * height bytes in worst case
[13:49] <durandal_1707> because each image scan line could be decoded in separate thread
[13:50] <durandal_1707> but having 8000 thread jobs may be too much
[13:57] <durandal_1707> ught there is memory leak in lavf/mxg
[14:42] <cone-449> ffmpeg.git 03Paul B Mahol 07master:a72b41035cc4: mxg: do not leak mxg->buffer
[14:52] <cone-449> ffmpeg.git 03Michael Niedermayer 07master:21f68528e39d: aacdec: dont pop random configurations
[14:55] <cone-449> ffmpeg.git 03Paul B Mahol 07master:c46400ddecab: cafenc: do not leak caf->pkt_sizes
[15:38] <tclarke> hi, I'm trying to write a custom muxer which write raw frames (using the rawvideo codec) to a circular framebuffer in a memmapped file...the problem I have is that ffmpeg is opening the file I specify and performing flush's, etc. so the second time my write_packet function is called the mmap section seems to be overritten with 0's
[15:39] <tclarke> how can I tell ffmpeg that my AVOutputFormat will handle it's own I/O and an avio context should not be opened?
[15:41] <Compn> sounds complicated
[15:43] <tclarke> you have a better solution?
[15:43] <ubitux> tclarke: maybe look how it's done in the segment muxer
[15:44] <ubitux> or lavf/hlsenc
[15:44] <tclarke> ok, I'll look there, thanks for the pointers
[16:31] <Compn> j-b : could just do what winamp does for interfaces , put it in html , make user use his web browser to see bittorrent status :)
[16:31] <Compn> for bittorrent playback thing
[17:27] <creep> hi
[17:27] <creep> what kind of music do you consume ?
[17:27] <creep> Flyleaf - i'm so sick, Fully Alive http://www.youtube.com/watch?v=iWIADZKU9dw hackers used to listen stuff like this on films
[17:32] <gnafu> Cool story, bro.
[17:44] <Compn> hey hes back
[17:45] <mystica555> [libvorbis @ 0x119c8e0] Que input is backward in time
[17:45] <mystica555> ...que hora es? >.>
[17:46] <Compn> is that a libvorbis error or our error ?
[17:47] <mystica555> im not entirely sure
[17:48] <mystica555> it visibly appears similar (with a teal background) to
[17:48] <mystica555> [libx264 @ 0x11b0600] frame I:46 Avg QP:25.17 size: 26041
[17:48] <mystica555> que... was brown
[17:48] <mystica555> libvorbis teal
[17:50] <Compn> i dont see that error message in the code.
[17:50] <mystica555> then its probably in libvorbis.
[17:50] <Compn> no
[17:50] <Compn> its in libfaac to
[17:50] <Compn> somehow
[17:50] <Compn> lets see where it comes from...
[17:51] <mystica555> hmm
[17:51] <creep> nah
[17:52] <creep> spam? :P
[17:52] <mystica555> /usr/lib64/libavcodec.so.54.59.100: Que input is backward in time
[17:52] <mystica555> hm
[17:52] <mystica555> says strings
[17:52] <mystica555> so definitely there, somewhere.
[17:53] <Compn> audio_frame_queue.c: av_log(afq->avctx, AV_LOG_WARNING, "Queue input is backward in time
[17:53] <Compn> is it somehow eating "ue input" ?
[17:53] <mystica555> likely a newer version patched it
[17:53] <mystica555> ive maybe not updated in a few months
[17:53] <Compn> maybe its just a typo
[17:53] <Compn> que > queue
[17:54] <Compn> english isnt the first language of many devs in the project
[17:54] <Compn> and there are multiple ways to spell the same english words :\
[17:54] <mystica555> well yes im sure it is a typo
[17:54] <mystica555> que just isnt one of the right ways
[17:54] <mystica555> either you "cue" something - to advance it to a certain poinbt in time in preperation for playback for example; or you put something into a queue
[17:55] <mystica555> que is spanish however
[17:55] <mystica555> ;)
[17:56] <Compn> que is slang for queue then :P
[17:56] <Compn> in this case
[17:56] <mystica555> que paso
[17:56] <mystica555> ;)
[17:56] <Compn> vato loco!
[17:58] <Compn> but yeah it appears its been fixed in the meantime
[18:20] <cone-345> ffmpeg.git 03Michael Niedermayer 07master:6f77122bf571: aac: reconfigure output on pop
[18:28] <saste> ubitux: I'm asking because an av_free() would have fail.
[18:28] <saste> what do you mean?
[18:40] <BBB-work> "Description: Not many VP7 samples are in the wild, but no open-source decoder exists although Google has released a specification. Write a decoder that reuses as much as possible of existing FFmpeg code, it is likely that functions of the existing decoders for On2-based formats will be useful." ?
[18:40] <BBB-work> I didn't know google released a vp7 spec?
[18:43] <nevcairiel> all i know of is this: http://multimedia.cx/mirror/VP7_Data_Format_and_Decoder_Overview.pdf
[18:44] <nevcairiel> it says to be used in conjunction with the reference decoder
[18:45] <BBB-work> that is not released by google, that is dropped off a truck
[18:45] <BBB-work> that's not the same thing :)
[18:45] <nevcairiel> since when do we care about that specifically? :d
[18:46] <BBB-work> you're writing a wiki page to give it to a company named google for inclusion in gsoc, right?
[18:47] <BBB-work> that very same company is claimed to have done something it didn't do on that very wiki page
[18:51] <nevcairiel> for all i know, they have released it, whoever added that shall be questioned
[19:21] <ubitux> saste: mmh sorry, it would have fail in some configuration only
[19:21] <saste> ubitux, i still can't understand
[19:22] <saste> how can av_free fail?
[19:22] <ubitux> in case of memalign hack for instance
[19:22] <ubitux> if the allocated buffer is not from av_malloc or similar
[19:22] <ubitux> it would fail
[19:23] <saste> but we are using av_malloc
[19:23] <saste> strdup -> av_strdup should be obviously fixed
[19:23] <ubitux> yes that's the point
[19:23] <ubitux> on a memalign config, the buffer allocated with strdup() would have fail if the pointer was freed with a av_free()
[19:24] <ubitux> memalign-hack* configu
[19:24] <saste> so if i fix strdup -> av_strdup() it will be safe, right?
[19:25] <ubitux> sure
[19:25] <saste> good i'll fix and push then
[19:28] <ubitux> btw, about the [HH:] thing the guy actually sent an updated patch
[19:29] <saste> ubitux, for some reason I got my mails stucked on the server
[19:29] <saste> so they are released just after many hours
[19:30] <saste> the more time i don't fetch mails, the more time they tend to stay stucked
[19:30] <saste> so in my mailbox i don't always have all the mails
[19:31] <saste> it sucks big time, but nothing i can do much about without spending a few hours debugging or nailing tech support
[19:31] <ubitux> :(
[19:34] <ubitux> ffcat
[19:34] <ubitux> fast forward cat
[19:42] <tonsofpcs> I ran ffprobe -show_packets on a .ts and it shows dts = pts on all packets (video and audio) that don't have a dts. Why is this? (it's an ATSC TS if that matters)
[19:46] <saste> ffcat -> fast frenzy kittie
[19:55] <creep> http://images.puszta.com/hazi_kifli/Hazi%20kifli%201000.jpg !
[19:55] <Tjoppen> oh mama
[20:01] <cone-345> ffmpeg.git 03Stefano Sabatini 07master:f30979e0954e: lavfi/boxblur: add support to named options
[20:01] <cone-345> ffmpeg.git 03Stefano Sabatini 07master:8902f5b89c16: doc/filters: apply minor spacing fixes to boxblur docs
[20:18] <durandal_1707> michaelni: have anything to add to exr patches? (i have improved version with slice threading)
[20:21] <cone-345> ffmpeg.git 03Paul B Mahol 07master:345689129102: exr: check version too
[20:29] <michaelni> durandal_1707, the patches should be fine
[20:35] <cone-345> ffmpeg.git 03Paul B Mahol 07master:88d7259fd246: exr: merge multiple if () blocks
[20:35] <cone-345> ffmpeg.git 03Paul B Mahol 07master:1178868683d2: exr: fix overflow check in lineoffset processing
[20:35] <cone-345> ffmpeg.git 03Paul B Mahol 07master:cc0bd7841389: exr: support more line orders
[20:35] <cone-345> ffmpeg.git 03Paul B Mahol 07master:9c75c24de40d: exr: check if line offset table is complete only once per frame
[20:35] <cone-345> ffmpeg.git 03Paul B Mahol 07master:fefd4b0924cc: exr: cosmetics: reindent
[20:35] <cone-345> ffmpeg.git 03Paul B Mahol 07master:7a930f310ad2: exr: remove superfluous check
[20:35] <cone-345> ffmpeg.git 03Paul B Mahol 07master:89c77b6b155c: exr: shorter channel buffer names
[20:58] <cone-345> ffmpeg.git 03Michael Niedermayer 07master:2c21d34ea44d: avutil/mem: do a small set of checks for memalign hack before freeing.
[21:30] <code-guru> hi all
[21:31] <code-guru> I can't set mpegts metadata when use hls muxer.
[21:31] <code-guru> for example when I add "-metadata service_provider="Some provider" it doesn't change service_metadata when I use hls m3u8 muxer
[21:31] <code-guru> can anybody help me with that?
[21:33] <llogan> code-guru: sorry. wrong channel. #ffmpeg is for user help.
[21:34] <code-guru> llogan, it seems more of a bug to me ? isn't it ?
[21:53] <ubitux> btw, about GSoC
[21:53] <ubitux> i can suggest some subtitles stuff
[21:54] <ubitux> and i'd be very happy to see improvements in the gif muxer/encoder
[21:54] <ubitux> like non-intra frames, and better colors
[21:56] <llogan> would you be interested in mentoring?
[21:56] <ubitux> subtitles yes, gif maybe
[21:58] <llogan> can you add this to the ideas page?
[21:58] <ubitux> maybe in an hour or two
[21:58] <llogan> or email me the info and i can do it if that will be easier for you.
[21:59] <ubitux> i think i need an account too :p
[22:00] <llogan> but i can't do it until later either...more video editing to do
[22:09] <cone-345> ffmpeg.git 03Malcolm Bechard 07master:239b88c284af: Improve QTRLE encoding performance, no change to output file size/content.
[22:50] <code-guru> https://ffmpeg.org/trac/ffmpeg/ticket/2296
[22:53] <cone-345> ffmpeg.git 03Martin Storsjö 07release/1.1:65bf4c9c4578: mpegvideo: Do REBASE_PICTURE with byte pointers
[22:53] <cone-345> ffmpeg.git 03Michael Niedermayer 07release/1.1:0b2b8ab97962: qdm2: check array index before use, fix out of array accesses
[22:53] <cone-345> ffmpeg.git 03Michael Niedermayer 07release/1.1:52b18c1fde65: roqvideodec: check dimensions validity
[22:53] <cone-345> ffmpeg.git 03Michael Niedermayer 07release/1.1:ab434bf0d051: vqavideo: check chunk sizes before reading chunks
[22:53] <cone-345> ffmpeg.git 03Vicente Jimenez Aguilar 07release/1.1:6626a7df5345: doc: Fix some obsolete references to av* tools as ff* tools
[22:53] <cone-345> ffmpeg.git 03Michael Niedermayer 07release/1.1:80ddf7889eec: Merge remote-tracking branch 'qatar/release/9' into release/1.1
[23:11] <ubitux> fun fact: there are around 12k additionnal commits in ffmpeg against l. (without counting merges)
[23:12] <nevcairiel> now do that excluding commits that they ported over
[23:12] <nevcairiel> :D
[23:12] <ubitux> 12k-3 ?
[23:15] <ubitux> ok so anyway
[23:15] <ubitux> i need to mail mike for a multimedia.cx account, right?
[23:15] <wm4> simply subtracting the number of commits shows that ffmpeg has slightly more commits, I think?
[23:15] <wm4> though you'd have to subtract merge commits too, hm
[23:15] <ubitux> i didn't count them
[23:16] <ubitux> otherwise it's 14k or something iirc
[23:16] <ubitux> but TBH i should have count them somehow :)
[23:16] <ubitux> llogan: ping?
[23:21] <cone-345> ffmpeg.git 03Michael Niedermayer 07release/1.1:a642be972d48: h264: ensure that get_format() is called when changing format but not otherwise.
[23:21] <cone-345> ffmpeg.git 03Michael Niedermayer 07release/1.1:469cb6119386: avcodec_decode_audio4: check got_frame_ptr before handling initial skip
[23:21] <cone-345> ffmpeg.git 03Michael Niedermayer 07release/1.1:02d1efdd5b61: h264: check that luma and chroma depth match
[23:21] <cone-345> ffmpeg.git 03Andrea3000 07release/1.1:ef688e742557: matroska: fix missing ,
[23:21] <cone-345> ffmpeg.git 03Michael Niedermayer 07release/1.1:705e89d75fde: update for 1.1.3
[23:41] <michaelni> BBB-work, how should the VP7 text on the SOC page be worded ? (i dont know at all where which spec came from)
[23:42] Action: michaelni just found his un/pw for wiki-multimedia again ;)
[23:42] <ubitux> can you find mine if it exists?
[23:44] <michaelni> ubitux, if you had an account try to login theres a button to get a pw by email
[23:44] <michaelni> if you didnt then well i cant help
[23:44] <ubitux> yeah i think i never had one :)
[23:45] <michaelni> you need to talk to mike then
[23:45] <michaelni> thats the disadvantage of it over our wiki
[23:45] <michaelni> delay to get access
[23:47] <ubitux> yep right, just sent an email
[23:49] <michaelni> hey beastd, do you have a wiki multimedia account ? if so you could help improve our SOC page
[23:50] <cone-345> ffmpeg.git 03Senthilnathan M 07master:12a269a5229d: lavu/parseutils: allow MM:SS format for duration in av_parse_time()
[23:51] <cone-345> ffmpeg.git 03Stefano Sabatini 07master:ce84c14ab173: lavu/parseutils: remove spurious "]]" in date format specification
[23:51] <cone-345> ffmpeg.git 03Stefano Sabatini 07master:160def32da43: doc/muxers: fix reference to @anchor defined in another manual
[23:51] <beastd> hi michaelni, i have. will try to have a look at it this weekend.
[23:51] <michaelni> ok thx
[00:00] --- Sat Feb 23 2013
1
0
[00:17] <tonsofpcs> just ran ffprobe -show_packets on a .ts and it is showing a dts = pts on all packets that don't have a dts. Why is this?
[00:18] <tonsofpcs> (the .ts is an ATSC transport stream, if it helps)
[02:22] <rcombs> not directly ffmpeg related, but...
[02:22] <rcombs> I've got some files here with the .WAV extension, but which start with 14 28 08 32. Does anyone recognize that header?
[02:37] <jagginess> hi
[02:37] <jagginess> "ERROR: libx264 not found", been trying to get this fixed.. been using flags, no use :(
[02:40] <klaxa> did you try installing x264?
[02:40] <jagginess> yup that's installed.. this is a prefix of ~/local
[02:40] Action: jagginess is following https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide#x264
[02:40] <klaxa> what is your compilation configuration for ffmpeg?
[02:41] <jagginess> latest git,
[02:41] <jagginess> ./configure --enable-gpl --enable-libass --enable-libfaac --enable-libfdk-aac --enable-libmp3lame \
[02:41] <jagginess> --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libspeex --enable-librtmp --enable-libtheora \
[02:41] <jagginess> --enable-libvorbis --enable-libvpx --enable-x11grab --enable-libx264 --enable-nonfree --enable-version3
[02:41] <jagginess> and --prefix=/home/username/local
[02:41] <jagginess> i also have 3 export variables of flags
[02:41] <klaxa> i take you compiled x264?
[02:42] <klaxa> and have a libx264.a in the x264 directory?
[02:42] <jagginess> ( export CPPFLAGS=-I/home/user/local/include; export CXXFLAGS=-I/home/user/local/include; export LDFLAGS=-L/home/user/local/lib )
[02:43] <klaxa> mhh... did you install the x264 libraries to /home/user/local?
[02:43] <jagginess> ./usr/local/include/x264.h
[02:43] <jagginess> ./usr/local/include/x264_config.h
[02:43] <jagginess> ./usr/local/lib/pkgconfig/x264.pc
[02:43] <jagginess> ./usr/local/lib/libx264.a
[02:43] <jagginess> ./usr/local/bin/x264
[02:43] <jagginess> ./bin/x264
[02:43] <jagginess> (in ~/local)
[02:43] <klaxa> that's.... well it certainly *should* work i think
[02:43] <jagginess> hmm
[02:43] <jagginess> i think i need to use ~/local/usr/local/lib too
[02:49] <jagginess> klaxa, how can i add multiple paths for flags? (export CPPFLAGS='-I/path/abc -I/path/def')
[02:49] <jagginess> :/
[02:49] <klaxa> yes that should work like that
[02:50] <jagginess> i no expert, i usually dont' compile at all..
[02:50] <jagginess> you sure ? :}
[02:50] Action: jagginess replaces comma with space
[02:50] <klaxa> it works whenever i compile stuff
[02:52] <jagginess> can i use --extra-cflags twice with ./configure ?
[02:53] <jagginess> hmm
[02:53] <klaxa> i'm not sure, but you should be able to add all the flags with using it only one time
[02:54] <jagginess> klaxa, how would i separate paths, i use a comma?
[02:54] <klaxa> in what context? commas are genereally not used i think
[02:54] <jagginess> --extra-cflags=/path/abc,/path/def ?
[02:55] <klaxa> i doubt that that will do anything even, since it lacks parameters like -L or -I
[03:15] <jagginess> klaxa, mind if i ask you can try it? :} lol
[03:16] <klaxa> mmmh my system is probably quite different from yours and i compile ffmpeg without problems every now and then
[03:16] <relaxed> jagginess: --extra-cflags="-I/path/to/prefix/include" --extra-ldflags="-L/path/to/prefix/lib"
[03:17] <relaxed> if you need multiple, --extra-cflags="-I/path/to/prefix/include -I/some/other/place/include"
[03:17] <jagginess> yeah tried both :/
[03:17] <relaxed> make distclean and try again
[03:18] <jagginess> hmm did that too
[03:18] <jagginess> ./configure passes without x264 enabled.
[03:18] <jagginess> always.
[03:18] <relaxed> pastebin or it didn't happen
[03:19] <jagginess> i'll live without x264 for now.. thanks kla
[03:19] <jagginess> klaxa
[06:04] <magn3ts> Is there ever any hope for libav and ffmpeg remerging?
[06:06] <klaxa> magn3ts: doubtful i think
[06:07] <Guest82135> hi
[06:09] <llogan> magn3ts: it's hard to say, but i'd leave open the possibility at least.
[06:26] <Guest82135> /msg nickserv ghost code-guru
[06:26] Last message repeated 1 time(s).
[06:26] <Guest82135> /msg nickserv regain code-guru
[06:26] <Guest82135> /msg nickserv help
[06:27] <Guest82135> /msg nickserv ghost code-guru
[06:27] <Guest82135> /msg nickserv identify code-guru 1qaz1qaz
[07:02] <llogan> i hope he changes his password
[07:37] <hendry> hi, I'm looking for test videos that print their elapsed time. Working on a syncronized video player. Any tips?
[08:34] <klaxa> hendry: if you succeed i'd be interested in a synchronized video player (you are talking about synchronized over a network right?)
[08:55] <SpmP> cross purpose... can ffmpeg be used to convert over under to side by side?
[09:02] <relaxed> SpmP: yes
[09:04] <SpmP> relaxed: great. fair call... <cough> how 8) cheers.
[09:07] <relaxed> SpmP: Use the exapnd filter to create a "canvas" and then the movie/overlay filter to place it.
[09:07] <relaxed> expand*
[09:09] <hendry> klaxa: yes
[09:09] <relaxed> SpmP: sorry, s/expand/pad/
[09:11] <SpmP> yaR your way too ahead of me.. still trying to find expand in your docs 8)...
[09:11] <relaxed> SpmP: I meant the pad filter
[09:12] <SpmP> cool...
[09:15] <relaxed> SpmP: http://pastie.org/6315146
[09:19] <relaxed> That's an old example, you may have to change it a little and use -filter_complex.
[09:23] <SpmP> yarrr... still getting my head around the simple syntax, oh dear, ffmpeg is amazingly complete. I guess this is imagemagik for video 8)... dammit, ille get there
[10:11] <knobo> Hello, I get "moov atom not found" when I play I video I have.
[10:12] <knobo> I read somewhere that AtomicParsley could help me fix the problem. But How do I use it?
[10:12] <knobo> Or is there any other way, using ffmpeg?
[10:12] <Mavrik> um
[10:13] <Mavrik> if you don't have a moov atom in video
[10:13] <Mavrik> your video is badly broken
[10:14] <code-guru> try this. "ffmpeg -i broken.mp4 -acodec copy -vcodec copy -y new.mp4"
[10:15] <klaxa> say code-guru did you change your nickserv password after pasting it in here?
[10:15] <code-guru> no :-o
[10:15] <klaxa> better do that :X
[10:16] <knobo> An instruction video I got from china. Maybe the Great Wall changed it...
[10:16] <code-guru> thanks
[10:16] <JEEB> knobo, basically the moov atom is the "index" of the file
[10:16] <JEEB> with all the offsets and so forth
[10:16] <JEEB> if that is borked
[10:17] <JEEB> you could as well throw the file to the fjords
[10:17] <code-guru> what else I pasted here ? :-D
[10:17] <JEEB> in other words, try to use a way of transfer that tries to keep the data in tact :)
[10:19] <code-guru> that's a funny FAIL !
[10:19] <klaxa> huh? why?
[10:19] <knobo> How do I transfere videos back and forth from China?
[10:19] <code-guru> sending out your password to the channel
[10:19] <knobo> off topic, maybe..
[10:20] <JEEB> no idea, personally I'd use rsync via ssh?
[10:20] <JEEB> that tries to keep the data intact
[10:21] <code-guru> if it's a live stream, you can use ffserver
[10:22] <JEEB> and what would ffserver do with a live stream? It's a streaming component itself.
[10:22] <knobo> But I don't have a server in china.. Well this is off topic. And the sender in china probably don't know how to use ffmpeg, linux, command line or facebook.
[10:23] <klaxa> upload on a filesharing site?
[10:23] <JEEB> or just keep downloading that video until the file size and md5 hash match, if you even have an md5 hash of the file available :D
[10:25] <code-guru> I mean if it's video coming from a closed circuit camera, ffserver can broadcast it to the net based on a protocol like rtp
[10:25] <knobo> That's what they did. http://mail.qq.com/cgi-bin/ftnExs_download?k=7c376165859204c1fbc71f7e166102…
[10:28] <relaxed> curl has support for resuming a download
[10:39] <SpmP> relaxed: Thanks for the intro
[13:01] <spinx60> Hello im trying to stream live to twitch.tv with ffmpeg but when i try and start my script i get this http://pastebin.com/QgPM9PRE
[13:02] <ubitux> you didn't paste the command you used
[13:02] <ubitux> and since this is a syntax error, the error lies into the command you used
[13:02] <spinx60> anyone who can
[13:02] <spinx60> ohh sorry 1 sec
[13:03] <spinx60> streaming live_XXXXXXXXXXXXXX where X is my stream key
[13:03] <spinx60> i have the script in my .bashrc file
[13:04] <spinx60> its wierd couse i got my it to start my stream last night using that command. and i got sound but no video on my stream'
[13:10] <spinx60> Anyone got any idés ?
[13:11] <ubitux> since you still didn't paste the command we can't help you
[13:15] <spinx60> i execute the following command [spinx60@GreenPeace ~]$ streaming live_XXXXXXXXXXXXXXXXXXXXX where XXX is my own stream key
[13:15] <spinx60> streaming live_xxxxxxxxxxxxxx is the command
[13:17] <spinx60> i put X in becouse i dont want to show my stream key ofc
[13:19] <ubitux> how are we supposed to guess what the "streaming" command is?
[13:21] <spinx60> how you mean.. guess the command.. i pasted my script in my -bashrc file and that is the command kind of couse i start it by streaming live_XXXXXX Key etc ? dont you get it =
[13:21] <spinx60> look at this aswell
[13:22] <ubitux> there is a syntax error in the command defined in your .bashrc or wherever you put it
[13:22] <ubitux> unless you show it we can't guess the problem
[13:22] <spinx60> http://pastebin.com/XrvpMYQL
[13:22] <ubitux> ah, finally.
[13:22] <spinx60> there is it
[13:22] <spinx60> sorry my bad there totally
[13:24] <ubitux> with this exactly, you get the exact error pasted above ?
[13:24] <spinx60> yes
[13:25] <ubitux> sh -x streaming xxxx
[13:25] <spinx60> and the thing is my script looks the same and is the same as i used last night when i got it to stream with sound but whitout video
[13:25] <ubitux> does it show a sane exec?
[13:26] <spinx60> gives me sh: streaming: No such file or directory
[13:26] <ubitux> erm.
[13:26] <ubitux> well
[13:26] <ubitux> try to exec it yourself then
[13:26] <spinx60> its something in my script that makes me able to run the command streaming live_XXX
[13:26] <ubitux> i guess yes.
[13:27] <spinx60> yeah that works.. now il paste the outcome but now im back to square one where i dont get any picture/capture on my stream its only a black screen with sound
[13:29] <ubitux> then pastebin ffmpeg output again
[13:29] <spinx60> http://pastebin.com/hcAwEj42
[13:30] <spinx60> i need to go now but i will be back later on check that out and send any message and i llook at it later.. thanks for the help so far
[13:31] <shevy> "Encoder (codec id 86017) not found for output stream #0.0"
[13:31] <shevy> anyone has a listing for what is codec ix 86017?
[13:32] <shevy> command was: ffmpeg -i foo.mp3 -ss 00:00:11 -t 00:00:22 test.mp3
[13:33] <sacarasc> shevy: Try adding -c:a copy
[13:53] <jarno> Hi all, what happens when encoding a video and the PTS (presentation time stamp) overflows? PTS is supposed to be monotonically increasing function (i.e. value), but if encoding either with high frame rate or a long video, it might overflow, right?
[15:54] <spinx60> Im back again and still having issues with my Twich.tv streaming. i use the following command http://pastebin.com/pFVJa5TF
[15:55] <spinx60> and the outcome is http://pastebin.com/ee67TY1n
[15:56] <spinx60> im having my stream live with sound but whitout any video/capture at all.
[16:02] <spinx60> Anyone who can shed some light on this ? must be something rather simple.. i mean i cant be the only one having/had this problem
[16:38] <Wihl-w> Hi, I have 20 audio files (audio1.wav etc) and 20 images (image1.png etc) that I'd like to combine so that image1.png is swn for the duration of audio1.wav and so on
[16:39] <Wihl-w> Is that something that would be possible to do with ffmpeg in an automated fashion?
[16:40] <Wihl-w> *shown for the duration
[16:40] <relaxed> spinx60: remove -s "1280x720" and add -filter:v scale=1280:720,format=yuv420p
[16:42] <relaxed> yuv444p was most likely the problem.
[16:55] <spinx60> testing it now
[17:04] <relaxed> Wihl-w: bash for loop; -loop 1 -i blah.png -shortest
[17:09] <addisonj> hey, looking for a bit of help to transcode a video, copy the audio, and write to stdout, here is my command ffmpeg -i video.mpg -c:v libx264 -preset superfast -f:a copy -f:v h264 -
[17:09] <addisonj> problem right now, video is working, but not getting any audio streams
[17:09] <addisonj> doing an ffprobe confirms its being stripped
[17:10] <relaxed> you're using the wrong format
[17:10] <relaxed> and it's '-c:a copy'
[17:10] <addisonj> I had that earlier as welll...
[17:10] <Mavrik> hmm
[17:11] <relaxed> what kind of audio?
[17:11] <Mavrik> any idea how can I clear the "orientation" setting from transform matrix in mp4?
[17:11] <relaxed> what are you piping to?
[17:12] <addisonj> relaxed: writing to stdout, which is picked up by a little node.js script and then send down in response to http request :P
[17:12] <addisonj> the video works just fine (when I buffer in mplayer) but no audio
[17:12] <relaxed> -f flv
[17:14] <masch> hi. I need to change the frame rate of a clip from 29.97 to 23.976 without dropping frame. i tried decoding the clip to rawvideo and pipe that to another ffmpeg process that reencodes the video but it seems to lose the audio by that :-/
[17:14] <masch> can i tell ffmpeg to keep all frames but just change the framerate?
[17:14] <addisonj> flv? isn't that flash video? I am trying to do this with html5 vid player
[17:14] <relaxed> ok, try -f webm
[17:15] <addisonj> I was going to try webm next, is there something wrong about outputting h264? still don't understand why the video would drop out...
[17:16] <relaxed> masch: Yes. Pipe rawvideo back into ffmpeg like ffmpeg ... - 2>/dev/null|ffmpeg -r 23.976 -s WxH ....
[17:16] <relaxed> addisonj: h264 is not a container.
[17:17] <addisonj> ah...
[17:18] <relaxed> masch: oyu'll have to demux the audio and remux them back together.
[17:18] <relaxed> you'll*
[17:18] <addisonj> relaxed: also, going to webm, I am going to have to transcode audio to vorbis right?
[17:18] <masch> relaxed: oh okay, any idea how to change the audio tempo ?
[17:19] <masch> so it fits to the new duration?
[17:19] <relaxed> masch: sox can do it
[17:19] <relaxed> maybe there's a ffmpeg filter for it, I'm not sure
[17:20] <relaxed> addisonj: be sure to use -c:a libvorbis
[17:27] <relaxed> masch: you know, with mplayer you can simply play back at a different framerate.
[17:29] <relaxed> spinx60: \o/
[17:36] <spinx60> relaxed thanks alot
[18:38] <undercash> hello
[18:38] <undercash> is there a way to hardcode the title in the output?
[18:38] <undercash> i watched the filter options but i didnt see anything related
[18:39] <saste> undercash, drawtext?
[18:39] <undercash> https://github.com/FFmpeg/FFmpeg/blob/master/doc/ffmpeg.texi
[18:39] <undercash> this is yours i assume
[18:39] <undercash> i tried the -metadata option but won't work
[18:40] <undercash> donno about drawtext, let me look
[18:40] <saste> "hardcode the title" it's ambiguous
[18:41] <undercash> well to display the filename or metadata title in the output
[18:42] <saste> undercash, script it and use drawtext
[18:42] <undercash> it can retrieve the filename automatically or you need to specify it everytime?
[18:47] <Fjorgynn> Kör fram!
[18:50] <undercash> ok thx saste, i ll check this , but i need to recompile with libfree
[18:50] <undercash> at least i know what to look at
[18:51] <saste> undercash, you're passing the filename, you know it
[18:51] <saste> no the filter does not know nothing about the input filenames
[18:52] <undercash> $file as text to draw maybe
[18:52] <undercash> need to try :)
[18:52] <spinx60> hello im finally got ffmpeg to stream sucessfully to Twitch.tv with http://pastebin.com/jnS1ra2H command
[18:54] <spinx60> my question now is is there any way for me to make a script to this command where i can easier set more options like quality/resolution/framerate etc can anyone point me in the right direction to a guide or something mabee a complete script that i can edit etc ?
[19:04] <spinx60> Anyone at all who can point me to a good guide of some sort ?
[19:08] <klaxa> spinx60: maybe look at general x264 options ( http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide ) in general crf, preset, computation power and your upload speed limit video quality
[19:09] <spinx60> klaxa Thanks for your answer im not so good at this thats why i asked. but im learning
[19:41] <hughmanwho> I'm trying to pull a H264 stream via RTSP and decode it into YUV frames
[19:42] <hughmanwho> However I've allocated memory via avformat_alloc_context() then called 'avformat_open_input'
[19:43] <hughmanwho> Afterward I am getting a value of 0 for the packet_size within curStream->context->packet_size
[19:45] <hughmanwho> Any thoughts? Due to a conversation I had yesterday I was told that changing the value of packet_size can change how fast avformat_open_input takes to execute.
[19:46] <hughmanwho> But I'm thinking packet_size shouldn't be equal to zero.
[20:40] <hughmanwho> Having some problems connecting to a RTSP stream, my connection code is here: http://pastebin.com/3v5wNMf4
[20:41] <hughmanwho> Then when I call av_read_frame(streamStream->context,&(streamStream->packet)), I always get back an error
[20:41] <hughmanwho> Anyone have any thoughts?
[21:26] <code-guru> hi anyone can help me with hls muxer please?
[21:27] <code-guru> I can't set mpegts metadata when use hls muxer
[21:28] <code-guru> for example when I add "-metadata service_provider="Some provider" it doesn't change service_metadata when I use hls m3u8
[21:34] <durandal_1707> code-guru: if it does not but should it is bug, and should be reported at usual location
[21:37] <code-guru> I'm going to implement the commands functionality in my code finally, if there it's possible to fix this using the api, it's also ok
[21:37] <code-guru> any help please ?
[21:41] <PigDude> hi, i get "Uknown format" for a .flac.
[21:44] <PigDude> sure llogan
[21:44] <PigDude> here is my command and verbose output: http://dpaste.com/970491/
[21:44] <PigDude> with -v, i see error "Unable for find a suitable output format for 'some.flac'"
[21:45] <PigDude> ffmpeg -formats shows i have flac, aac, and alac formats available, I think
[21:46] <PigDude> here is a paste of a grepped -format ouput, showing those: http://dpaste.com/970505/
[21:47] <PigDude> I don't know what the flags D, E, or A mean in this context
[21:47] <llogan> 2004. Why are you using something from 8 years ago?
[21:47] <PigDude> i'm using ffmpegx distribution because it takes a very long time to build ffmpeg on my machine and i wanted to do this quickly
[21:47] <llogan> that isn't supported here. you're wasting your time using it.
[21:48] <PigDude> very well
[21:48] <llogan> there are static builds available
[21:48] <PigDude> where can I find these?
[21:48] <PigDude> these are decent? http://www.evermeet.cx/ffmpeg/
[21:48] <llogan> yes. use most recent one from "snapshots"
[21:48] <PigDude> thanks, installing now
[21:49] <llogan> ubitux: he is OSX user, AFAIK.
[21:49] <ubitux> oh mac os
[21:49] <ubitux> yeah right ok sorry :)
[21:50] <llogan> don't worry. i almost did the same.
[21:50] <ubitux> llogan: btw, it seems libmodplug was added :))
[21:51] <ubitux> that's nice, i can ffplay the keygen musics with the packaged ffmpeg now \o/
[21:51] <llogan> ubitux: as repetence several users are asking how to get a better looking gif output: ffmpeg -i input -pix_fmt rgb24 out.gif any suggestions?
[21:51] <llogan> ah, nice. i'll have to try it out
[21:51] <ubitux> i used to add two scale filters
[21:52] <ubitux> 05:59 <@ubitux> this is generated with: ./ffmpeg -i ~/samples/big_buck_bunny_1080p_h264.mov -vf 'select=gt(scene\,.4),scale=-1:120,format=rgb8,format=rgb24' -vsync vfr -frames:v 10 -r 1 -y bbb-scenes.gif
[21:52] <ubitux> i have this in my irc log
[21:52] <ubitux> try to adapt
[21:52] <llogan> ah, thanks.
[21:53] <PigDude> llogan, thanks for showing me that, this new version of ffmpeg has a much nicer command line interface
[21:53] <ubitux> llogan: it's far from perfect, but that's nice anyway
[21:53] <llogan> PigDude: good. did it work as expected?
[21:54] <PigDude> i now see a more detailed error message, and the conversion makes some progress. my output file is written and the operation itself takes perhaps 15 seconds to error out.
[21:54] <PigDude> here is the command and output: http://dpaste.com/970549/
[21:55] <PigDude> i thought on a whim that "operation not permitted" might be related to unix permissions, but `sudo` does not remedy this (I would not think this an issue anyway, as this is all occurring in my home directory)
[21:56] <PigDude> note that i am using the "libvo_aacenc" format and not "aac", as recommended by ffmpeg when I first specified "aac"
[21:56] <llogan> libvo_aacenc isn't any better than aac
[21:57] <divVerent> yes, libfdk_aac is the good one
[21:58] <PigDude> oh, ok, i will attempt with that format
[21:58] <PigDude> it appears i did not build with that
[21:59] <PigDude> in any case, i see this error with mp3 output also ... so my input is the source of this issue
[22:00] <PigDude> the audio tracks play in VLC well, so it's not like i'm throwing crap at ffmpeg
[22:00] <PigDude> though VLC certainly copes well with crap
[22:01] <llogan> PigDude: you most likely won't find one with libfdk_aac support due to it being non-free and non-redistributable, IIRC.
[22:02] <llogan> does it only occur with that input?
[22:03] <PigDude> it occurs with other flac files too. i have not tested each file, but none of my tests have a different result
[22:03] <PigDude> what does "Error while decoding stream #0:0: Operation not permitted" mean?
[22:07] <PigDude> huh, i'm not even sure what sort of an error this is. the generated audio file works fine
[23:27] <hughmanwho> what is the difference between avcodec_open & avcodec_find_decoder? Do you need both of them?
[23:33] <exutux> hi
[23:41] <exutux> http://pastebin.ubuntu.com/5556480/ can someone can help me to convert this type of file into mp3 audio?
[23:42] <exutux> when I try to convert it I have a rustle audio on .mp3
[23:42] <exutux> :(
[23:47] <exutux> http://pastebin.ubuntu.com/5556480/ can someone can help me to convert this type of file into mp3 audio? when I try to convert it I have a rustle audio on .mp3
[23:49] <exutux> using ffmpeg -i test.wav -acodec libmp3lame -ac 1 -ab 92k test.mp3
[00:00] --- Sat Feb 23 2013
1
0
[00:12] <cone-561> ffmpeg.git 03Michael Niedermayer 07master:47e92153b61b: mjpegdec: check for packed rgb mjpeg.
[00:28] <cone-561> ffmpeg.git 03Michael Niedermayer 07master:551d8b58ebc0: brender_pix: use a grayscale palette as fallback instead of a uninitialized one.
[01:01] <michaelni> someone wants to test(&review) "v4l2: bug #1570 and possible solution" ?
[01:03] <ubitux> saste:
[01:03] <ubitux> libavfilter/vf_overlay.c:201:19: warning: overlay_formats may be used uninitialized in this function [-Wmaybe-uninitialized]
[01:03] <ubitux> libavfilter/vf_overlay.c:202:19: warning: main_formats may be used uninitialized in this function [-Wmaybe-uninitialized]
[01:04] <ubitux> michaelni: isn't it specific to some devices?
[01:04] <ubitux> iirc originally i wasn't able to reproduce
[01:04] <ubitux> but the issue might have evolved
[01:05] <michaelni> could be specific yes but some testing is still better than none
[01:06] <ubitux> http://fate.ffmpeg.org/report.cgi?time=20130220225403&slot=x86_64-archlinux… erm it seems we have some more in vf idet too
[01:06] <ubitux> (which looks unrelated to the others)
[01:09] <cone-561> ffmpeg.git 03Clément BSsch 07master:580e2285577c: lavf/swfdec: remove 8-bit audio FIXME.
[01:09] <cone-561> ffmpeg.git 03Clément BSsch 07master:7346354b5597: lavf/microdvd: fix muxing.
[01:09] <cone-561> ffmpeg.git 03Clément BSsch 07master:9ed6af4be2c6: fate/subtitles: add MicroDVD remux test.
[01:59] <cone-561> ffmpeg.git 03Michael Niedermayer 07master:dc9edb06b011: vf_idet: Fix macro arguments sideeffect
[01:59] <cone-561> ffmpeg.git 03Michael Niedermayer 07master:cdb7a1ac6345: vf_idet: adapt default scores due to bugfix
[02:01] <ubitux> thx :)
[02:30] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.8:2b6f3be08250: avcodec_align_dimensions2: Ensure cinepak has large enough buffers.
[02:30] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.8:dfeef3a209ec: cook: check js_subband_start for validity
[02:36] <pajamaboy> quick question, does ffmpeg support authentication against an rtmp target, it appears to in a recent patch, but doesn't seam to be working for me
[03:26] <cone-561> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n0.8.14': unknown revision or path not in the working tree.
[03:26] <cone-561> Use '--' to separate paths from revisions
[03:26] <cone-561> refs/tags/n0.8.14:HEAD: vf_idet: adapt default scores due to bugfix
[03:54] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:1fd86f9a2136: vqavideo: fix return type
[03:54] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:8aedb7515674: tiff: check bppcount
[03:54] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:5af2fd317d84: wma: check byte_offset_bits
[03:54] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:17966ae5bb43: qdm2: increase noise_table size
[03:54] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:0a57df38f468: Update for 0.8.14
[03:54] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:2b6f3be08250: avcodec_align_dimensions2: Ensure cinepak has large enough buffers.
[03:54] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:dfeef3a209ec: cook: check js_subband_start for validity
[03:55] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:a92cdd896345: Merge branch 'release/0.8' into release/0.7
[03:55] <cone-561> ffmpeg.git 03Michael Niedermayer 07release/0.7:3be6a8e1eb97: update for 0.7.15
[04:04] <Compn> ubitux : i have an interesting question. do any containers have subtitle codepage metadata ?
[04:05] <Compn> like does mkv have stream 3 srt rus cp1251 ?
[04:06] <Compn> or utf8 maybe
[04:18] <cone-561> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n0.7.15': unknown revision or path not in the working tree.
[04:18] <cone-561> Use '--' to separate paths from revisions
[04:18] <cone-561> refs/tags/n0.7.15:HEAD: vf_idet: adapt default scores due to bugfix
[08:10] <Belinda> hi
[08:11] <Belinda> I am trying to make a patch to add S16 support to aacdec.c but i am failing continuously. Any ideas?
[09:12] <cone-190> ffmpeg.git 03Carl Eugen Hoyos 07master:d714576037e3: Use SMPTE standard notation when showing mxf Universal Labels.
[09:12] <cone-190> ffmpeg.git 03Carl Eugen Hoyos 07release/1.0:4c2b6ce26cf9: Fix bits_per_coded_sample when encoding png with frame-level multithreading.
[09:12] <cone-190> ffmpeg.git 03Carl Eugen Hoyos 07release/1.1:4be63111d1a2: Fix bits_per_coded_sample when encoding png with frame-level multithreading.
[10:10] <highgod> Hi, I want to ask a question. I want to modify the configure file to make it detect cl/cl.h and libOpenCL.a and make the ffmpeg link libOpenCL.a,the code I add is "enabled opencl && require opencl CL/cl.h OPENCL_init -lOpenCL", but the output is "ERROR: opencl not found", I am not familiar with modifing configure file. I used --extra-cflags=-I/opt/include --extra-ldflags=-L/opt/lib/x86
[10:10] <highgod> --extra-libs="-lOpenCL" to compile the opencl code
[10:31] <Belinda> I am trying to make a patch to add S16 support to aacdec.c but i am failing continuously. Any ideas?
[10:31] <durandal_1707> why?
[10:32] <durandal_1707> aacdec is floating point decoder
[10:34] <nevcairiel> decoders intentionally only output their native format, if you need another format, use swresample to convert it
[10:49] <Belinda> actually i am using it on android and android only recognizes s16, so i wanted to add support for "requested_name".
[10:55] <durandal_1707> the aac decoder is float one, but there are fixed decoders on web
[10:55] <av500> one can also convert float to fixed
[10:56] <av500> clamp( lrintf( *(float*)src * (1 << 15)))
[10:57] <durandal_1707> does that give good results?
[10:58] <av500> define good
[10:59] <divVerent> it sure is good enough :P
[10:59] <divVerent> actually, shouldn't one use ((1 << 15) - 1)? nitpicking :P
[10:59] <Belinda> i tried float_to_int16_interlave_c but it didn't work
[10:59] <av500> why not?
[10:59] <nevcairiel> like i said, use swresample to convert float to int16
[11:00] <nevcairiel> thats what its there for
[11:00] <nevcairiel> among other things
[11:00] <durandal_1707> av500: because floats from aacdec are between -1 and 1
[11:00] <nevcairiel> a patch to allow s16 output would not really be accepted
[11:01] <Belinda> :/
[11:01] <nevcairiel> it used to be possible, but that was removed from all sorts of decoders in favor of native output
[11:02] <nevcairiel> so instead of converting inside the decoder, just convert after the decoder
[11:02] <av500> yes, but one can easily find the old code
[11:02] <nevcairiel> same code required, jus
[11:02] <nevcairiel> just in your app
[11:02] <av500> yes
[11:02] <nevcairiel> or just use the library which has functions for this =p
[11:03] <Belinda> actually i found the old code but there are many structural changes, it is easy to ruin
[11:04] <Belinda> since i am doing this on the fly, i try to reduce cpu and ram usage
[11:04] <av500> its only audio
[11:04] <av500> doing that conversion wont kill you
[11:04] <av500> and you need to do it anyway
[11:05] <nevcairiel> the decoder works in float anyway, so it needs conversion at some point, the cpu required is the same no matter where its done
[11:06] <av500> yep
[11:07] <durandal_1707> so on slow arch with float it is slow already
[11:07] <av500> aac takes like 30mhz on an ARM
[11:08] <av500> find me an android device that you can even clock that low
[11:08] <av500> its a non issue
[11:10] <durandal_1707> av500: on arm? maybe
[11:10] <av500> yes
[11:10] <av500> on an armv7
[11:11] <av500> basically you have more cpu power due to leakage than due to that bit of audio decoding
[11:12] <durandal_1707> so fixed point decoder would not be faster?
[11:13] <av500> probably
[11:13] <av500> well
[11:14] <av500> since armv7 with neon etc has float I am not even sure fixed is faster
[11:27] <Belinda> AV_SAMPLE_FMT_S16 is not supported by swr_convert?
[11:28] <durandal_1707> it is supported
[11:28] <durandal_1707> read documentation
[11:28] <Belinda> ok sorry, i've made a mistake
[11:37] <ubitux> Compn: some formats contain the info yes
[11:38] <ubitux> (microdvd for instance)
[11:54] <durandal_1707> michaelni: this ^ spammer is really retarded
[11:55] <durandal_1707> delete him and png (that links to another site)
[11:56] <durandal_1707> also can permission for WikiStart be more restrictive?
[11:56] <durandal_1707> is there better bug tracker which have better way of handling fine grained permissions?
[11:57] <durandal_1707> if yes, switch to it
[12:38] <cone-212> ffmpeg.git 03Nicolas George 07master:01649c79c87c: lavfi/buffersrc: set channel layout if it is known.
[12:43] <cone-212> ffmpeg.git 03Nicolas George 07master:9fa5a47197c6: ffmpeg: set resample_pix_fmt for sub2video.
[12:51] <highgod> Hi,I add check opencl code to configure file, and set the check code to enabled opencl && require opencl CL/cl.h clEnqueueNDRangeKernel -lOpenCL,clEnqueueNDRangeKernel is a function of libopencl.a, but it failed,Then I review the libopencl.a and found that the function is clEnqueueNDRangeKernel@36, but if I set the code to enabled opencl && require opencl CL/cl.h clEnqueueNDRangeKernel@36
[12:51] <highgod> -lOpenCL, it si still wrong. does anyone konws how to fix it? Thanks
[13:01] <highgod> require_cpp?
[13:03] <cone-212> ffmpeg.git 03Vicente Jimenez Aguilar 07master:b5ad422bf4e6: doc: filters: Correct BNF FILTER description
[13:03] <cone-212> ffmpeg.git 03Diego Biurrun 07master:3e85b46ecf3c: arm: vp8: Add missing #includes for header to compile standalone
[13:03] <cone-212> ffmpeg.git 03Diego Biurrun 07master:a0a1ac0f091d: dxva2: Add missing #define to make header compile standalone
[13:03] <cone-212> ffmpeg.git 03Michael Niedermayer 07master:6b8f21190d5a: Merge remote-tracking branch 'qatar/master'
[13:05] <cone-212> ffmpeg.git 03Paul B Mahol 07master:a3d890527ed3: exr: reset compression in decode_frame()
[13:05] <cone-212> ffmpeg.git 03Paul B Mahol 07master:59015c1118ac: exr: constify s in rle_uncompress()
[13:05] <highgod> sill don't work
[13:16] <highgod> require2 works
[13:21] <Compn> durandal_1707 / michaelni : also, it would help to give devels more access on trac to ban spammers.
[13:27] <michaelni> theres a bayesian spam filter only problem is we have too few spamers so it hasnt enabled itself yet
[13:29] <michaelni> also trac can use various external spam detectors like akismet but IIUC we need a api key for each of these
[13:34] <michaelni> and about banning, that slavon888 guy used a different IP each time IIRC
[13:35] <iive> might be good idea to block known exit nodes of tor network and vpn/proxy tunnels.
[13:35] <iive> probably some spam lists have them all.
[13:37] <michaelni> trac already uses multipe ip blacklists, i dunno whats on them exactly though
[13:46] <michaelni> Compn, if you want to ban IPs or content see for example trac.edgewall.org/wiki/BadContent and trac.openpsa2.org/wiki/BadIP
[13:50] <michaelni> also in case i worded it unclearly, some volunteer needs to get us api keys for akismet and the others listed at http://trac.edgewall.org/wiki/SpamFilter with exception of where they key is just for training the external service)
[13:50] <michaelni> also i dont know if these keys are free
[13:51] <michaelni> that is if someone wants trac to use these external spam detectors ...
[14:04] <durandal_1707> i gonna push rest of exr patches
[14:04] <durandal_1707> than gonna try slice threading
[15:04] <blackmage> Does anyone know if pushing to an rtmp destination that requires authentication is supported
[15:04] <blackmage> it looks like it was added in a patch, but i'm not certain, based upon some of the docs i've seen
[15:06] <durandal_1707> what docs?
[15:06] <durandal_1707> didn't you said in another channel that your buid is missing ssl support?
[15:06] <cone-212> ffmpeg.git 03Michael Niedermayer 07master:bdeb61ccc679: h264: check that luma and chroma depth match
[15:09] <blackmage> i'm seeing mentions of it, but here is the one that says you need ssl
[15:09] <blackmage> http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=7&t=723#p2581
[15:10] <durandal_1707> ssl is needed
[15:12] <durandal_1707> or not
[15:32] <durandal_1707> could fate survive 8 mb of new samples?
[15:52] <durandal_1707> it looks like it needs user[:pass]@hostname
[15:52] <durandal_1707> (reading source code)
[15:53] <blackmage> okay, i'll try that, thanks durandal
[16:29] <wm4> so it seems some libavformat demuxer actually expect a special AVIOContext and use pb->opaque without checking
[16:30] <wm4> e.g. hls.c accesses this, and if the caller passes a custom AVIOContext, it will crash
[16:45] <cone-212> ffmpeg.git 03Michael Niedermayer 07master:669d50b44596: ec: check that reference pictures have matching parameters before using them.
[17:06] <Compn> wm4 : btw , write down any info you are learning, so the next person who wants to do what you do, doesnt have to guess and figure out ... :)
[17:06] <Compn> aka please update the docs or make source code comments, or wiki entry , anything
[17:08] <wm4> uh what, like what
[17:52] <Compn> wm4 : i thought you were adding a new format, not investigating a bug, nm
[18:07] <durandal_1707> finally! ffmpeg is recognised by google+: https://plus.google.com/+ffmpeg
[18:08] <Compn> finally, they can track ffmpeg people on google servers, yay...
[18:17] <durandal_1707> https://docs.google.com/presentation/d/1awg1CHM1w128iOBp_JOxE2DgHfywBeyjDe2…
[18:20] <ubitux> :)
[18:23] <ubitux> i like slide 6 too
[18:31] <durandal_1707> you can even chat with other readers
[18:52] <durandal_1707> http://revolution.byu.edu/programmingconcepts/potatoPseudocode.png
[19:31] <durandal_1707> anyone want to review my patches?
[20:04] <ubitux> durandal_1707: can i help?
[20:05] <saste> ubitux, durandal_1707: a second look at blend may help, especially with regards to interface/syntax
[20:08] <ubitux> blend looked good to me when i had a look at it
[20:08] <saste> ubitux, also boxblur and overlay patch
[20:08] <ubitux> named options?
[20:09] <ubitux> saste: btw, did you see the warnings in overlay?
[20:09] <saste> yes, and we're still missing a few filters if you want to help with low hanging fruits
[20:09] <saste> ubitux, nope
[20:25] <cone-48> ffmpeg.git 03Michael Niedermayer 07master:8036ba299d33: h264: move next/last picture init to decode_slice_header
[20:27] <durandal_1707> sase left
[20:46] <ubitux> erm max_samples isn't honored.
[20:58] <ubitux> looks like a mistake in 82541d83
[20:58] Action: ubitux mailing Nicolas
[20:59] <durandal_1707> "Failed to inject frame into filter network" <--- what that means
[21:01] Action: ubitux reviewing boxblur
[21:18] <cone-48> ffmpeg.git 03Paul B Mahol 07master:47619e41c4a1: blend filter
[21:19] <ubitux> yay :)
[21:19] <durandal_1707> now fix bugs in it
[21:19] <ubitux> :))
[21:20] <ubitux> durandal_1707: how i imagine you currently: http://z0r.de/5024
[21:21] <durandal_1707> ubitux: whatever it is i do not have flash/java or some other plugin
[21:22] <ubitux> i think it works with ffplay
[21:22] <ubitux> but it won't loop
[21:22] <ubitux> too bad :)
[21:22] <ubitux> indeed it works
[21:22] <ubitux> you need to imagine the audio and video loop now
[21:23] <ubitux> (they are not in sync)
[21:23] <durandal_1707> ubitux: ???
[21:23] <ubitux> durandal_1707: wget 'http://z0r.de/L/z0r-de_5024.swf'
[21:24] <ubitux> the basic idea of the loop (at least most of them) is to have a/v not really in sync
[21:24] <ubitux> for instances 5 frames looping, and a longer background audio, looping too
[21:24] <durandal_1707> wget: Command not found.
[21:24] <ubitux> (the a/v sync doesn't really matter)
[21:24] <ubitux> mmh
[21:24] <ubitux> fetch?
[21:25] <ubitux> curl ... > out.swf?
[21:25] <durandal_1707> hmm why clip is in rawvideo/audio, feared of patents?
[21:27] <ubitux> no idea
[21:27] <ubitux> happens often
[21:27] <ubitux> (in swf)
[21:48] <durandal_1707> saste: gonna review [PATCH] Handling special characters in a URL ? it take really to loong
[21:49] <saste> durandal_1707, too long? he posted a reply tomorrow
[21:49] <durandal_1707> ping reply
[22:00] <ubitux> actually the min/max samples works really well
[22:00] <ubitux> awesome Nicolas is awesome
[22:00] <ubitux> saste: you should use that feature too ;)
[22:01] <ubitux> i wonder if asetnsamples couldn't use it too
[22:02] <wm4> why isn't there libswblend :(
[22:02] <saste> ubitux, you should thank anton to be fair
[22:02] <saste> wm4, you know the answer
[22:02] <wm4> "patches welcome"?
[22:02] <saste> yes
[22:02] <wm4> "no thank you"
[22:03] <ubitux> saste: no
[22:03] <ubitux> it's a feature from Nicolas
[22:03] <ubitux> not present in libav
[22:03] <saste> is it?
[22:03] <ubitux> yes :)
[22:03] <ubitux> see 0689d5e17ac0a3269c604b4df2c01140be328647
[22:04] <durandal_1707> wm4: why libswblend?
[22:04] <ubitux> durandal_1707: to blend subtitles \o/
[22:04] <ubitux> (fast!)
[22:04] <wm4> durandal_1707: it would gather useless functionality into a library, instead of shattering it across libavfilter where it's needed for various things
[22:04] <durandal_1707> there are actually many libs that do blend but they all sucks afaik
[22:05] <llogan> saste: worthless fact for the day: there are more than 1500 -user subscribers.
[22:05] <ubitux> how many -devel?
[22:05] <llogan> also i heard back from mike. he'll be able to deal with account stuff.
[22:05] <durandal_1707> day before were 1800 (so it is 0.3 % drop)
[22:06] <llogan> ubitux: 1327
[22:06] <ubitux> nice
[22:06] <ubitux> 10 left
[22:06] <durandal_1707> that all trackers (gmane,google, etc)
[22:06] <llogan> digest members are included but digests are disabled and they do not recieve mail
[22:08] <llogan> saste: so i'll try to get the gsoc ideas page up later today (unless someone gets to it first), but i've been in a timestuck editing videos for the last few days...
[22:08] <saste> llogan, no hurry
[22:09] <saste> i'll be able to work actively on gsoc not before the first week of march
[22:09] <durandal_1707> llogan: so is number of users/devels stagnating or they are all spammers?
[22:10] <llogan> i don't actively watch the subscriber numbers and there are no spammers with accounts AFAIK (not including hijacked [yahoo mostly] accounts)
[22:14] <durandal_1707> there is no log or something like graph?
[22:17] <llogan> durandal_1707: not in the admin web panel, AFAIK.
[22:20] <llogan> but a graph would be interesting
[22:20] <wm4> durandal_1707: have you found a sample for vf_stereo3d yet?
[22:21] <saste> anyone wants to review the partial overlay patch?
[22:21] <saste> it's the missing one in my overlay patchset
[22:21] <durandal_1707> wm4: not yet there, why?
[22:21] <wm4> just wondering
[22:22] <durandal_1707> it is not that mandatory for porting
[22:22] <wm4> its output looks like modern art for me
[22:22] <durandal_1707> anything with bunch of different colors can be used
[22:23] <durandal_1707> wm4: get glasses
[22:23] <wm4> and I wonder what stuff like "anaglyph_green_magenta_half_color" actually means
[22:24] <durandal_1707> nothing relevant for anime scene
[22:29] <cone-48> ffmpeg.git 03Michael Niedermayer 07master:8a6449167a6d: avcodec_decode_audio4: check got_frame_ptr before handling initial skip
[22:29] <cone-48> ffmpeg.git 03Andrea3000 07master:8d8c59480e1b: matroska: fix missing ,
[22:31] <durandal_1707> wm4: looks like that filter is only way to experience 3d at home for free
[22:43] <cone-48> ffmpeg.git 03Michael Niedermayer 07master:04220b473e9d: h264: ensure that get_format() is called when changing format but not otherwise.
[22:43] <cone-48> ffmpeg.git 03Matt Wolenetz 07master:82a4a4e7caa9: Fix Win64 AVX h264_deblock by not using redzone on Win64
[23:06] <cone-48> ffmpeg.git 03James Almer 07master:67b3fcf65578: configure: Use libiconv on systems that lack built-in iconv
[23:06] <cone-48> ffmpeg.git 03Mirjana Vulin 07master:8d2eb5fe58ae: mips: optimization for float aac decoder (sbr module)
[23:06] <cone-48> ffmpeg.git 03Mirjana Vulin 07master:46d52a0b098f: mips: optimization for float aac decoder (ps module)
[23:10] <ubitux> it almost makes me wanting a mips device
[23:11] <michaelni> ubitux, ask the @mips.com people, maybe they would give you one for free
[23:16] <ubitux> not sure if i'll really make use of it, it's just a whim :)
[23:16] <ubitux> but it's nice to see all these optims
[23:18] <wm4> durandal_1707: it appears matroska even has hints for this 3D crap...
[23:20] <JEEB> H.264 can have hints as well, but yeah -- matroska had some crap too
[23:22] <durandal_1707> there is unsupported H.264 extension IIRC
[23:23] <JEEB> MVC you mean? that's separate, and pure H.264 decoders just don't have to care as it will get one of the views that way just fine
[23:23] <JEEB> I mostly meant the SEI (or VUI was it?) stuff
[23:23] <JEEB> where you can note what kind of multi-view stream it is
[23:23] <JEEB> x264 has support for those
[23:24] <Skyler_> SEI
[23:41] <ubitux> huh
[23:41] <ubitux> seems i have the audio normalization working
[00:00] --- Fri Feb 22 2013
1
0
[00:08] <xangadix> I have two questions
[00:08] <xangadix> 1) so I can code to quicktime animation codec using -f mov -vcodec qtrle -ac 2
[00:08] <xangadix> but how to set the quality of the image sequence?
[00:08] <xangadix> in most editing programs, I can set a quality from 0 to 100%, but I don't know how to tell ffmpeg that
[00:08] <xangadix> 2)
[00:09] <xangadix> is it possible to have ffmpeg make dxv files ?
[00:09] <xangadix> http://resolume.com/software/dxv.php
[00:10] <xangadix> and how would i add that :)
[00:10] <xangadix> these are my humble questions
[00:10] <xangadix> ow great and mighty people in #ffmpeg
[00:15] <ubitux> specs anyway?
[00:16] <ubitux> or it will need some reverse?
[00:16] <xangadix> specs for dxv ?
[00:16] <ubitux> yes
[00:16] <xangadix> I'm not sure where to find those either
[00:16] <xangadix> google stumbles on suggestions for divx, and dxva of which I'm not sure is the same
[00:17] <ubitux> for your first question, the quality is controlled by -qscale:v and relatex (qmin, qmax, etc)
[00:17] <xangadix> much obliged
[00:18] <xangadix> saw that for making jpgs, I was hoping it was kind of the same :)
[00:19] <xangadix> if the codec is not open source, can it still be used in ffmpeg? I ran through this conversation, http://lists.ffmpeg.org/pipermail/ffmpeg-devel-irc/2010-October/000227.html but couldnt make it out completely
[00:21] <ubitux> it will need some reverse
[00:21] <xangadix> by that you mean reverse engeneering ?
[00:21] <ubitux> yes
[00:21] <xangadix> darn
[00:21] <ubitux> you can a open a feature request on the trac
[00:21] <ubitux> and wait for a decade
[00:21] <ubitux> ;)
[00:21] <xangadix> :)
[00:21] <xangadix> I will
[00:21] <ubitux> unless more ppl are interested
[00:22] <ubitux> but it seems no one care atm
[00:22] <xangadix> in the meantime I will have to render to photojpg first, and then batch it with sony vegas or so
[00:22] <xangadix> how come that most video editing software will render it then?
[00:22] <xangadix> they are using the same dlls I'm assuming ?
[00:50] <sabton> i call avformat_open_input() with an AVFormatContext that I have pre-allocated, and have specified custom AVIOContext on, but it is opening a new http connection... so i end up with 2 connections. how can i get avformat_open_input() to use my custom AVIOContext?
[01:03] <meelu> i don't get it, i do "/usr/bin/ffmpeg /var/www/videos/filename.flv -vol 256 -y -acodec libmp3lame -ab 256k /var/www/output/filename.mp3 2> /var/www/logs/filename.txt" and it outputs "File '/var/www/videos/filename.flv' already exists. Overwrite ? [y/N]"
[01:10] <klaxa> meelu: you're missing "-i"
[01:10] <meelu> oh thanks
[01:10] <meelu> :_)
[01:13] <meelu> klaxa
[01:13] <klaxa> meelu
[01:13] <meelu> nothing
[01:13] <meelu> sorry bout that
[01:13] <klaxa> kk
[01:27] <votz> How can I instruct ffmpeg to avoid writing any id3 tags (id3v1 and id3v2) for an mp3? I just want the mp3 output, starting with the encoder information, LAME3.87.4UUU...
[01:49] <spinx60> Hello are there any friendly and late night ppl like me out here ?
[01:59] <xangadix> spinx60 not sure about the friendly part :p
[02:32] <pajamaboy> is it possible to perform rtmp authentication when trying to push a stream to an rtmp target
[02:32] <pajamaboy> such as akamai
[02:33] <xangadix> does rtmp support something like rtmp://user:pass@address.com ?
[02:33] <xangadix> like ftp
[02:33] <pajamaboy> don't think so, i tried that actually
[02:34] <xangadix> hm :-/
[02:35] <pajamaboy> i also tried pubUser and pubPasswd
[02:35] <pajamaboy> maybe i need to ask the devs
[02:35] <pajamaboy> :(
[02:36] <xangadix> dunno, I have only done it through wowza, and that uses a local autherisation file (on the server in that case)
[02:37] <pajamaboy> yea, thats how i'm doing it now too
[02:37] <pajamaboy> i'm trying to move to using nginx-rtmp
[02:39] <xangadix> can't help you there :(
[02:41] Action: xangadix zZz
[08:20] <Garryco> I am trying to make a patch to add S16 support to aacdec.c but i am failing continuously. Any ideas?
[08:21] <Garryco> I am trying to get s16 pcm format out from decoded aac, but it currently only gives FLTP
[09:25] <firewalker> hello everyone!
[09:26] <Garryco> yo
[09:26] <firewalker> I am not sure if a problem I face with FFmpeg 1.1.2 is an ffmpeg bug or not
[09:26] <firewalker> the issue is this one https://bugs.archlinux.org/task/33901
[09:27] <firewalker> should I open a bug report for ffmpeg, or not?
[10:32] <divVerent> double WTF:
[10:32] <divVerent> WTF1: ffmpeg play MIDI files?!?
[10:32] <divVerent> WTF2: it totally fails at them
[10:32] <divVerent> as if notes never end
[10:32] <durandal_1707> how are they detected?
[10:32] <divVerent> cause is apparently libmodplug "supporting" MIDI files
[10:32] <divVerent> but failing totally at them
[10:33] <divVerent> Input #0, libmodplug, from 'VEGGIESG.MID':
[10:33] <durandal_1707> what happens?
[10:33] <divVerent> instrument : Timidity GM patches
[10:33] <divVerent> it sounds TOTALLY wrong, many stuck notes
[10:33] <divVerent> can't even recognize the song
[10:34] <divVerent> apparently, from listening closely... apparently the last note on each channel continues playing until a next note comes
[10:34] <divVerent> also, it doesn't sound AT ALL like the timidity GM patches I am using
[10:34] <divVerent> haha, I see in strace... it did ATTEMPT to load timidity patches, but didn't succeed at any
[10:35] <divVerent> [pid 17742] open("/usr/local/share/timidity/instruments/.pat", O_RDONLY) = -1 ENOENT (No such file or directory)
[10:36] <divVerent> I actually remember noticing this like 3 years ago when libmodplug ADDED midi support... and was really disappointed at the bad output... so much, I made darkplaces whitelist file extensions to pass to modplug
[10:37] <malinens> ffmpeg version git-2012-07-02: ffmpeg -i "a.mp4" -an -ss 3 -t 00:00:01 -r 1 -threads 1 -y "a.%d.jpg" and I get this error on one video: Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)
[10:38] <malinens> when started from -ss 10 then screenshots are created correctly. is there fool-proof way to generate screenshot with ffmpeg?
[10:41] <malinens> http://pastebin.com/ZPmkSWCL
[11:02] <juanfont> hi. is there any way to force the use of mpegtsraw demuxer from the stream (or the url, or any place except the code)? thanks :)
[11:02] <juanfont> *from the stream -> from the transport stream
[11:24] <n1n0> the movie I need to extract images from is in the home folder. when using this command ffmpeg -ss 00:07:00 -t 00:02:00 -i i.MOV -r 0.1 P1000072/image%3d.jpg
[11:25] <n1n0> I get this output ffmpeg -ss 00:07:00 -t 00:02:00 -i i.MOV -r 0.1 P1000072/image%3d.jpg
[11:25] <n1n0> ffmpeg version 0.8.5-4:0.8.5-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers
[11:25] <n1n0> built on Jan 24 2013 18:03:14 with gcc 4.6.3
[11:25] <n1n0> *** THIS PROGRAM IS DEPRECATED ***
[11:25] <n1n0> This program is only provided for compatibility and will be removed in a future release. Please use avconv instead.
[11:25] <n1n0> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'i.MOV':
[11:25] <n1n0> Metadata:
[11:25] <n1n0> major_brand : qt
[11:25] <n1n0> minor_version : 512
[11:25] <n1n0> compatible_brands: qt
[11:25] <n1n0> creation_time : 1970-01-01 00:00:00
[11:25] <n1n0> encoder : Lavf52.93.0
[11:25] <n1n0> Duration: 00:03:10.00, start: 0.000000, bitrate: 713 kb/s
[11:25] <n1n0> Stream #0.0(eng): Video: h264 (Constrained Baseline), yuv420p, 634x360, 613 kb/s, 25 fps, 25 tbr, 50 tbn, 50 tbc
[11:25] <n1n0> Metadata:
[11:25] <n1n0> creation_time : 1970-01-01 00:00:00
[11:25] <n1n0> Stream #0.1(eng): Audio: aac, 44100 Hz, stereo, s16, 95 kb/s
[11:25] <n1n0> Metadata:
[11:25] <n1n0> creation_time : 1970-01-01 00:00:00
[11:25] <n1n0> Incompatible pixel format 'yuv420p' for codec 'mjpeg', auto-selecting format 'yuvj420p'
[11:25] <klaxa> don't...
[11:25] <n1n0> [buffer @ 0x87b2780] w:634 h:360 pixfmt:yuv420p
[11:25] <n1n0> [avsink @ 0x87b2920] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out'
[11:25] <n1n0> [scale @ 0x87b2da0] w:634 h:360 fmt:yuv420p -> w:634 h:360 fmt:yuvj420p flags:0x4
[11:25] <klaxa> post all that shit in the channel
[11:25] <n1n0> Output #0, image2, to 'P1000072/image%3d.jpg':
[11:25] <n1n0> Metadata:
[11:25] <n1n0> major_brand : qt
[11:25] <n1n0> minor_version : 512
[11:25] <n1n0> compatible_brands: qt
[11:25] <n1n0> creation_time : 1970-01-01 00:00:00
[11:25] <n1n0> encoder : Lavf53.21.1
[11:25] <klaxa> never do that
[11:25] <klaxa> ever
[11:25] <n1n0> Stream #0.0(eng): Video: mjpeg, yuvj420p, 634x360, q=2-31, 200 kb/s, 90k tbn, 0.10 tbc
[11:25] <n1n0> Metadata:
[11:25] <n1n0> creation_time : 1970-01-01 00:00:00
[11:25] <n1n0> Stream mapping:
[11:25] <n1n0> Stream #0.0 -> #0.0
[11:25] <n1n0> Press ctrl-c to stop encoding
[11:25] <n1n0> frame= 0 fps= 0 q=0.0 Lsize= -0kB time=10000000000.00 bitrate= -0.0kbits/s dup=0 drop=10
[11:25] <n1n0> video:0kB audio:0kB global headers:0kB muxing overhead -inf%
[11:26] <n1n0> ok, can you check it this time only
[11:26] <klaxa> you can go to #libav because you are not using ffmpeg
[11:27] <klaxa> and please... use pastebin or something to paste logs and post the link
[11:27] <klaxa> spare them from this spam
[11:29] <n1n0> ok, cause on another software support channels (within the freenode) was ok to post terminal lines
[11:29] <n1n0> thanx anyway
[11:29] <klaxa> you do not post more than 3 lines in any irc channel that is sane
[11:30] <n1n0> ok, thanks
[11:38] <durandal_1707> malinens: what you are actually trying to do?
[11:39] <malinens> durandal_1707: create screenshot from video
[11:41] <durandal_1707> malinens: single screenshot at exact time?
[11:41] <malinens> Basically at the start of the video, single
[12:47] <malinens> is there chat log for #ffmpeg channel somewhere on interwebs?
[12:49] <durandal_1707> http://lists.ffmpeg.org/pipermail/ffmpeg-devel-irc/
[12:50] <ubitux> only the devel channel is logged afaik
[12:51] <durandal_1707> not really
[12:51] <durandal_1707> user too
[12:51] <malinens> durandal_1707: I just wanted to check if You wrote someting to me when I was offline about ffmpeg screenshots
[12:52] <durandal_1707> malinens: i think using select filter may work better
[12:52] <malinens> ok, I'll check it
[12:58] <ubitux> durandal_1707: oh, right, my bad
[14:06] <ubitux> heh it seems the ffmpeg package in archlinux is now linked against a bunch ffmpeg-specific libraries
[14:06] <ubitux> nice.
[14:07] <ubitux> fontconfig, libass, libbluray, libmodplug, libv4l2, postproc, ...
[14:08] <ubitux> that's gonna get tricky if libav wants to replace the package at some point :)
[14:08] <durandal_1707> for example?
[14:09] <ubitux> for example?
[14:09] <ubitux> all the libs i quoted are not supported in libav afaik
[14:13] <durandal_1707> but libmp3lame is, so i still do not get what you mean
[14:15] <ubitux> what i mean is that if libav wants to replace the ffmpeg package here, they will need to drop a lot of configure switches, which might be very tricky for them to do
[14:15] <ubitux> (because feature-loss)
[14:19] <durandal_1707> why would they want to drop ffmpeg package?
[14:19] <sacarasc> Because many distros are doing it these days.
[14:20] <durandal_1707> many?
[14:20] <sacarasc> Ubuntu and all the Ubuntu likes, Debian...
[14:22] <durandal_1707> really?
[14:23] <durandal_1707> perhaps if every libav developer becomes package maintainer for separate distro that may be true
[14:35] <meelu> any idea why this cuts off without completing? i mean it doesn't run in the background untill completion like i want it to $exec_string = '/usr/bin/ffmpeg -i /var/www/videos/filename.flv -vol 256 -y -acodec libmp3lame -ab 256k /var/www/output/filename.mp3 2> /var/www/logs/filename.txt >/dev/null 2>&1 &'; exec($exec_string, $output, $return); if (!$return){ echo 'yey'; }
[14:43] <ubitux> sacarasc: "many" is a bit excessive; it's present in two distro, gentoo (where ffmpeg is also available), and debian (because there is a libav maintainer here)
[14:43] <ubitux> of course being in debian make it spread in all the variants
[14:46] <blackmage> do you guys know how to do rtmp authentication with ffmpeg?
[14:46] <blackmage> i'm getting an error 403 saying it needs authmod=adobe
[14:46] <blackmage> but i can't find any documentation
[14:48] <ubitux> add authmod=adobe&user=... in the url?
[14:53] <blackmage> i tried that already, doesn't seam to work
[14:53] <blackmage> :(
[14:59] <blackmage> apparently i need to compile with ssl support
[14:59] <blackmage> and add pubUser= and pubPasswd=
[15:17] <meelu> how do i run an ffmpeg command with php in the background
[15:17] <meelu> anyone knows?
[15:20] <blackmage> http://pastie.org/6291660
[15:21] <durandal_1707> you are using old and wrong version
[15:21] <durandal_1707> which does not have native rtmp support either
[15:23] <blackmage> okay, i'll compile it, and try instead of using the ubuntu package
[15:23] <blackmage> i was able to record some streams from an rtmp source, so i thought this one had rtmp support
[15:27] <durandal_1707> adobe and limelight published authentication is available from 1.1 release
[15:28] <durandal_1707> previouse version (guessing) used whatever librtmp needed
[15:29] <blackmage> i have a compiled version of the latest from github, i just didn't use it in that command i ran on my local box (it has the same error)
[15:30] <blackmage> so i'm probably just missing some options, i'm thinking
[15:30] <durandal_1707> post pastie of that one
[15:36] <blackmage> in the v1.1
[15:36] <blackmage> is it passed in by doing username=blah password=blah
[15:37] <sabton> Hello, can someone please have a look at this? trying to convert a flac file, this used to work but now seems like FFmpeg behavior changed when dealing with multiple streams: http://pastebin.com/T15cJZA5
[15:37] <durandal_1707> blackmage: dunno, i don't use it
[15:38] <blackmage> http://pastie.org/6292723
[15:43] <relaxed> sabton: it doesn't know how to handle the cover art, add -vn
[15:43] <Mavrik> *grumble* Compiling ffmpeg for ARM is PITA
[15:45] <durandal_1707> blackmage: looks like args you give are wrong
[15:46] <sabton> relaxed: thanks. it appears -vn disables video extraction. this i fine, any other gotchas like this come to mind?
[15:47] <Mavrik> well, yeah, passing "disable video" probably does that ;)
[15:47] <relaxed> sabton: No. I'm not sure if ogg supports album art, but you could try -vcodec copy to see if it works.
[15:48] <sabton> relaxed: what was the old behavior? did ffmpeg know to ignore the video stream since i don't provide a matching output video stream?
[15:50] <relaxed> I think it's considered metadata and ffmpeg ignored it.
[15:50] <relaxed> s/it's/it was/
[15:50] <relaxed> both work :)
[15:53] <sabton> gotcha, thanks :)
[16:53] <JodaZ> anyone got an idea how i could make webm thumbnails of pictures ? like a few seconds from a few points of a video cut together
[17:14] <relaxed> JodaZ: webm is a container
[17:48] <JodaZ> relaxed, the sky is blue
[18:16] <MattCz> Any way to speed up 'avformat_find_stream_info'? It's taking about 75 seconds to execute right now but seems like it should be a pretty fast one...
[18:21] <Mavrik> MattCz, are you reading a network stream?
[18:57] <spinx60> Hello. im new to ffmpeg and im trying to stream to Twitch.tv / Justin.tv and my script seems to work very well and my stream goes live. But theres no capture at all stream is black screen and even if i try and record desktop to .mkv file i cant even open it in like vlc player etc. i have no clue what to do been searching since last night and now im out of ideés can anyone shed some light on my problem... this is m
[18:57] <spinx60> y scrip http://pastebin.com/b1wsdPH7
[18:58] <JodaZ> spinx60, i usually use rtmpdump to stream twitch, in fact, i am doing that right now
[18:59] <spinx60> then Jodaz can you help me out plz :)
[18:59] <El_Angelo> hi
[18:59] <spinx60> im starting my stream with this command streaming live_xxxxxxxxxxxx for the key ofc
[19:00] <JodaZ> ooh, wait, stream to twitch not from, never mind
[19:00] <El_Angelo> i used ffmpeg to concatenate 2 ts files, but now the timestamps seem to be missing from the final ts files
[19:00] <Mavrik> spinx60, pastebin also the output
[19:00] <El_Angelo> i can't ff in the file...
[19:01] <spinx60> ehhh ? what so thats what im missing ? :) how schould a output look like ? im read guides and watched youtube clips etc and this is all they talked about..
[19:02] <Mavrik> spinx60, the text output of ffmpeg
[19:02] <Mavrik> as it runs.
[19:02] <spinx60> ohh yeah 1 sec
[19:03] <spinx60> here it is http://pastebin.com/AYMwEyqX
[19:04] <spinx60> thats what happends when i execute the command streaming live_xxxxxxxxxxxx the x is my key
[19:04] <Mavrik> hmm
[19:04] <Mavrik> and you say you get black frame only? with or without audio?
[19:05] <Mavrik> spinx60, um
[19:05] <spinx60> whitout audio
[19:05] <Mavrik> spinx60, pass "-pix_fmt yuv420p" to ffmpeg after libx264
[19:05] <spinx60> and yes only a bland screen on my twitch channel
[19:06] <spinx60> schould i paste it in to my scripe after libx264 you mean ?
[19:07] <Mavrik> spinx60, add it as parameter to ffmpeg command after you set vcodec
[19:10] <spinx60> yeah i did so
[19:10] <spinx60> no change
[19:10] <spinx60> my stream is http://www.twitch.tv/undate btw
[19:12] <spinx60> think i did what you wanted me to il paste the new script file with the changes for you too look at to see that i did right
[19:13] <spinx60> stream is live now btw so you can se for yourself
[19:15] <spinx60> http://pastebin.com/Q4uRiW1J
[19:16] <spinx60> hmm not seen it before but in the stream window i got a message now saying. seems to be a problem downloading the stream. not seen it before
[19:20] <JodaZ> spinx60, no "" around the -pix_fmt
[19:21] <spinx60> yeah tried that to. no change in that either
[19:23] <spinx60> what about these lines in the output http://pastebin.com/7V1Adtce
[19:23] <spinx60> can it have something to do with it ?
[19:24] <spinx60> i really cant wait to begin streaming. would love it soo much
[19:24] <JodaZ> dunno, but definitely remove the "" arround the -pix_fmt, thats definitely wrong
[19:25] <spinx60> yeah i did and i tried it whiout those aswell and no differense at all
[19:26] <JodaZ> paste output without them
[19:27] <spinx60> okey i can do that just started the stream again so.
[19:32] <spinx60> here ya go http://pastebin.com/7gNcpscn
[19:32] <spinx60> im using arch linux if it relevant
[19:35] <MattCz> Sorry, wrote up my question then walked away for a second 'avformat_find_stream_info' is currently taking me 75 seconds to execute.
[19:35] <MattCz> Which seems long, mavrik asked me if I'm reading a networks stream.
[19:36] <MattCz> I'm reading from a 'rtsp://' URL coming from a camera
[19:38] <spinx60> you guys finding anything or does it look dark for me to get it to work ?
[19:39] <spinx60> JodaZ: Thanks for helping me anyway i will be here if you find any answers
[19:52] <frogprince_mac> hi is this also a support channel for ffmpeg-php?
[19:58] <durandal_1707> frogprince_mac: nope
[20:03] <spinx60> this is frustrating as hell.. going crazy here :P
[21:05] <dceo> what is the difference between -vcodec h264 and -vcodec libx264 -- I have an ffmpeg compiled on May 16 2006 using the former, and when I play back an mp4 file encoded from it using other media players like vlc it says 64bit atoms are not supported
[21:07] <sabton> when calling av_read_frame(), where is the stream index that it is reading from assigned?
[21:10] <dceo> i'm looking now, it shows av_read_frame( p_sys->ic, &pkt ); I will keep looking around a bit through those structures
[21:10] <JodaZ> wow, 2006 *_* they had 264 back then
[21:11] <dceo> seems old but 2006 was leading up to 2009 so its connected to an era ;)
[21:13] <sabton> it looks like AVFormatContext->cur_st holds the selected stream to decode...
[21:13] <sabton> i guess my question is now, when is cur_st initially set? I am doing custom IO, i think i need to set the stream index myself'
[21:14] <dceo> i guess I can try and print out the number of streams, nb_streams as a first debugging step
[21:15] <Mavrik> dceo, "-vcodec h264" will try to use the internal ffmpeg (experimental) encoder
[21:15] <Mavrik> while libx264 uses external "libx264" encoder for h.264
[21:15] <Mavrik> using the internal one isn't recommended :)
[21:16] <JEEB> uhh
[21:16] <Mavrik> MattCz, try lowering probe_size on AVFormatContext if your stream is very low on bitrate
[21:16] <JEEB> I don't think the libavcodec H.264 encoder never got in
[21:16] <JEEB> it's the decoder
[21:16] <JEEB> s/never/ever/
[21:17] <Mavrik> JEEB, it's in for awhile no AFAIK, just needs the "-experimental 2" switch afaik
[21:17] <Mavrik> even though, if I remember correctly, ffmpeg prefers libx264 when you type "h264"
[21:18] <dceo> I would imagine they have test streams that they run against, maybe support has somehow dropped for these older encoders then
[21:18] <JEEB> no, there is no h264enc
[21:18] <JEEB> there is only libx264 as a H.264 encoder in ffmpeg
[21:19] <JEEB> the very low-feature H.264 encoder (baseline only) got to the mailing list a few years ago
[21:19] <JEEB> but never, never got into the repository as far as I know
[21:22] <JEEB> yeah, it was on the mailing list in '06
[21:22] <Mavrik> hrm, of course,
[21:22] <Mavrik> I was thinking of AAC
[21:22] <Mavrik> I'm gonna go to sleep now, stop talking crap. Sorry.
[21:22] <JEEB> http://lists.mplayerhq.hu/pipermail/ffmpeg-devel/2006-June/012884.html
[21:22] <JEEB> I don't think this ever got into libavcodec, mainly because it was completely useless for real usage :)
[21:22] <JEEB> even back then
[22:32] <timofonic> Hello
[22:33] <timofonic> What about Daala from Xiph? I hope it surpasses H265, H264 and WebM :)
[22:35] <Zeeflo> is .sup supported by nightly builds (burek) ?
[22:36] <JEEB> timofonic, yes -- a format that is not even specified yet...
[22:37] <JEEB> the plan for them is to make something better than HEVC, H.264 or VP8/9
[22:37] <JEEB> but currently there is code to test ideas and discussion on what kind of design the format should take
[22:37] <timofonic> JEEB: I hope Google merges efforts instead being lone gunmen...
[22:38] <timofonic> JEEB: And use a matroska-based container, at least. As for audio... Opus?
[22:38] <JEEB> uhh
[22:38] <JEEB> > opus > matroska
[22:38] <JEEB> > implying they will get that finished
[22:38] <JEEB> also
[22:38] <JEEB> xiph is making up their own new container format as far as I can see
[22:38] <timofonic> What about Dirac too?
[22:39] <timofonic> Another one container? :P
[22:39] <timofonic> Opus < matroska? Uhm?
[22:39] <JEEB> no
[22:40] <JEEB> > is used in many places to mark quoted content
[22:40] <JEEB> also given how early in design and development daala is, I would pretty much say to you that you should keep it "behind your ear" and check up on it every now and then
[22:40] <JEEB> but
[22:40] <JEEB> it's at the very least a few years from being a reality
[22:45] <Zeeflo> why the f all these different containers..
[22:45] <Zeeflo> Its madness
[22:45] <Zeeflo> and makes no sense
[22:45] <Zeeflo> a container is a container..
[22:45] <Zeeflo> doesnt matter if its blue or green
[22:45] <Zeeflo> ffs
[22:45] <JEEB> well, various containers have various problems
[22:46] <JEEB> generally when a sane community makes a new one, it fixes some of the problems related to the previous ones
[22:46] <Zeeflo> its idiotic
[22:46] <JEEB> not really
[22:46] <JEEB> matroska is already starting to show its limitations as opus is showing
[22:47] <JEEB> hacking stuff into "mp4" really isn't worth it because nothing outside of some OSS stuff will support it
[22:47] <JEEB> the new xiph container's ideas finally seemed more sane than their stuff with the original ogg "container" >_>
[22:48] <Zeeflo> the future is streaming containers..
[22:48] <JEEB> and? the new xiph one should be better for that, too
[22:48] <JEEB> than matroska
[22:48] <JEEB> for one
[22:48] <Zeeflo> and no players would support it..
[22:48] <Zeeflo> (flash)
[22:49] <Zeeflo> like weve seen with mkv
[22:49] <Zeeflo> its a great container, but you cant use it for streaming..
[22:49] <Zeeflo> So who cares about it
[22:49] <JEEB> you have a very limited view on what is "streaming" then
[22:49] <Zeeflo> no
[22:49] <Zeeflo> mkv's cant rtmp stream
[22:49] <Zeeflo> not through flash
[22:49] <JEEB> it's an adobe protocol, are you fucking surprised?
[22:49] <Zeeflo> not through html5
[22:50] <Zeeflo> no
[22:50] <Zeeflo> im not surprised.
[22:50] <JEEB> and html5 doesn't have rtmp in it to begin with as far as I know
[22:50] <Zeeflo> But when 90% of the worlds devices use flash to receive audio/video then whos stupid?
[22:50] <Zeeflo> the people who create containers that flash will support or those who dont?
[22:50] <JEEB> ...
[22:50] <JEEB> so you say EVERYONE should use FLV for EVERYTHING?
[22:50] <Zeeflo> you cant really debate that
[22:50] <Zeeflo> no
[22:51] <JEEB> and you do understand how flash doesn't even support shit that is defined in "mp4" official, for example?
[22:51] <JEEB> you do, right?
[22:51] <Zeeflo> the only container that will go with rtmp streaming is mp4
[22:51] <JEEB> ...
[22:51] <Zeeflo> and webm
[22:51] <JEEB> rtmp does NOT use mp4
[22:51] <JEEB> and it does NOT use webm
[22:51] <Zeeflo> but webm sucks even harder
[22:51] <JEEB> rtmp is FLV
[22:51] <Zeeflo> in its truest form yes
[22:52] <Zeeflo> but its a shitty container
[22:52] <JEEB> there's no other form
[22:52] <JEEB> you can use various containers as sources for rtmp streams, yes
[22:52] <Zeeflo> you can rtmp stream mp4's
[22:52] <JEEB> yes, it gets remuxed on fly
[22:52] <Zeeflo> and ogg
[22:52] <Zeeflo> etc
[22:52] <JEEB> jesus christ have you ever actually looked at rtmp streams' content!?
[22:52] <JEEB> instead of talking out of your ass
[22:52] <Zeeflo> yes
[22:53] <Zeeflo> i look at it every day
[22:53] <JEEB> then why the fuck are you talking out of your ass
[22:53] <Zeeflo> cause flv sucks
[22:53] <Zeeflo> for high definition i would never use flv
[22:53] <JEEB> I agree, flv sucks -- but still, look at your fucking rtmp streams
[22:53] <JEEB> and no, SD/HD does not matter
[22:53] <JEEB> flv is a CONTAINER
[22:53] <Zeeflo> not when you have other containers that can be streamed with rtmp
[22:53] <JEEB> not a format
[22:54] <JEEB> oh for fuck's sake
[22:54] <timofonic> JEEB: Opus audio or opus container?
[22:54] <JEEB> GO LOOK AT WHAT DATA YOU GET FROM THE RTMP STREAM
[22:54] <JEEB> I BEG OF YOU FAGGOT, TAKE A FUCKING LOOK
[22:54] <Zeeflo> im not a faggot, and you need to calm down bro
[22:54] <JEEB> I just don't know where to start making you understand
[22:55] <Zeeflo> i do understand
[22:55] <JEEB> what you push to an rtmp(e) service and what is coming out of the rtmp(e) stream do not equal
[22:55] <Zeeflo> I just dont give a crap?
[22:55] <JEEB> then why the hell are you talking as if you know your stuff
[22:55] <Zeeflo> what comes out of the rtmp stream is flv
[22:55] <timofonic> JEEB: So with not make a modern, network robust, low latency, suitable for bad connections, streaming/voip/webcam protocol?
[22:56] <timofonic> JEEB: So say just one instead a dozen depending on the kind of media stream
[22:56] <JEEB> Zeeflo, then it doesn't matter what the hell the source file is as long as the streaming server can process it into what is needed
[22:56] <JEEB> your whole point goes trembling down
[22:56] <Zeeflo> no
[22:56] <Zeeflo> like i said, theres only a few containers that supports rtmp
[22:56] <JEEB> I mean, if it's H.264/AAC it can be whatever you want container-wise as long as your thing that is serving an rtmp(e) stream can remux it into flv
[22:57] <JEEB> ..
[22:57] <JEEB> man, you are talking shit. Please stop that.
[22:57] <Zeeflo> ok
[22:57] <Zeeflo> bro
[22:57] <mauricio> hi, I need some tips about how to stream rtp/rtsp at lowest latency
[22:57] <JEEB> your rtmp(e) streaming provider is limited to some formats, which then get converted/remuxed/used as-is when streamed over rtmp(e)
[22:57] <Zeeflo> you go make a video stream and audiostream in a MKV, put it on CloudFront and try serve it to e.g. JW Player
[22:57] <Zeeflo> let me know what you find out
[22:57] <JEEB> Zeeflo, read my last line
[22:58] <Zeeflo> thats what Ive been saying the whole time!
[22:58] <JEEB> no it hasn't been
[22:58] <Zeeflo> Only a few containers are suited for RTMP
[22:58] <JEEB> ...
[22:58] <JEEB> that is one incorrect way of putting that
[22:58] <Zeeflo> i dont know any CDN's that lets you stream AVI, MKV etc
[22:58] <Zeeflo> do you?
[22:59] <Zeeflo> please tell me
[22:59] <Zeeflo> OH no! They gotta support the shittiest containers of them all!
[22:59] <Zeeflo> webm..
[22:59] <Zeeflo> mp4
[22:59] <timofonic> JEEB: So why not do a proper streaming protocol instead? One that it's very robust and can be even suitable for sucky internet connections
[22:59] <JEEB> everything that can use f.ex. ffmpeg or libav for the input can handle pretty much everything
[22:59] <JEEB> I have no fucking idea about CDNs
[22:59] <JEEB> and that's a whole fucking separate discussion
[22:59] <Zeeflo> its like the whole world is run by freaking Apple!
[22:59] <Zeeflo> fuck em!
[22:59] <JEEB> blame your CDN for providing crappy service
[23:00] <Zeeflo> I do! Every day
[23:00] <JEEB> don't go out saying that the containers are the problem
[23:00] <Zeeflo> well FLV is still a shit poor stream to output!
[23:00] <Zeeflo> The problems arent really the containers.. Its the webbrowsers
[23:00] <timofonic> JEEB: Well, I have tons of problems when playing streams from a bad connection. Tons of cache problems and such...
[23:00] <JEEB> that's a separate discussion, although to be honest FLV is an "OK" simple container for H.264/AAC
[23:01] <JEEB> it's a very simple container, and does what it's supposed to do
[23:01] <timofonic> JEEB: But well, using mplayer/mplayer2/mpv solves partially the problem...
[23:01] <JEEB> I don't like it, but its crappiness comes from something else
[23:01] <JEEB> and yes, video playback in browsers suck dongs
[23:01] <timofonic> JEEB: Adobe?
[23:01] <JEEB> but that's a whole separate discussion
[23:01] <Zeeflo> That actually reminds me
[23:01] <timofonic> JEEB: But the streaming protocol might be something that needs to get improved, too
[23:02] <JEEB> timofonic, I don't give a fuck, I was correcting Zeeflo's incorrect statements
[23:02] <Zeeflo> Can anyone tell me a better way to stream to pc/mac/consoles other then web browsers?
[23:02] <timofonic> JEEB: This HTML5 trend is bullshit, still we don't have proper streaming/voip
[23:02] <timofonic> JEEB: ok...
[23:02] <Zeeflo> would it be to have a player/software created that would link to our CDN?
[23:02] <mauricio> rtsp isnt good enouh?
[23:02] <timofonic> JEEB: You just want to have grumpy discussions :)
[23:03] <Zeeflo> and have our customers download it to their pc's?
[23:03] <Zeeflo> shut up timofonic
[23:03] <Zeeflo> JEEB,
[23:03] <timofonic> Zeeflo: No, thanks. Stop acting rude and Asperger-like
[23:03] <timofonic> Zeeflo: You and JEEB should be more polite
[23:03] <Zeeflo> what do you think would be the ideal way to stream RTMP to a clients desktop?
[23:04] <timofonic> Seriously, IRC is full of morons
[23:04] <Zeeflo> or a clients console (xbox, ps3 etc)
[23:04] <JEEB> timofonic, sorry -- Zeeflo's general attitude of "I know even if I do not know or am saying it incorrectly" basically made me go xkcd 386
[23:05] <timofonic> JEEB: I agree
[23:05] <JEEB> but your comments from the side that were unrelated kind of pissed me off
[23:05] <Zeeflo> FFS shut up with the crying about rude irc dude! you pussies
[23:05] <Zeeflo> im debating
[23:05] <timofonic> JEEB: ok, I don't have enough level and got confused. Sorry
[23:06] <timofonic> Zeeflo: No, you aren't. You are argumenting and requesting in a rude way, so you don't deserve a reply at all
[23:06] <timofonic> Zeeflo: I don't cry, I ignore morons. Sorry, acting like Linus Torvalds isn't cool anymore. Specially is you are older than 15yo
[23:06] <timofonic> is=if
[23:06] <Zeeflo> jesus christ you are a faggot timofonic
[23:06] <Zeeflo> go cry in some corner
[23:06] <timofonic> Zeeflo: No, you are a moron ;)
[23:07] <Zeeflo> "waah the bad internet dude yelled at me"
[23:07] <Zeeflo> ...
[23:07] <timofonic> Zeeflo: YOu are the reason geeky people are known as childish and stupid
[23:07] <Zeeflo> yea sure
[23:07] <timofonic> Zeeflo: No, I just point at you
[23:07] <Zeeflo> youre right, now fuck off
[23:07] <Zeeflo> dont loose your finger while youre at it..
[23:07] <timofonic> Zeeflo: Get a life, get emotional skills and stop acting like a teenage boy
[23:08] <Zeeflo> oh my.. Dad is yelling at me!
[23:08] <timofonic> Zeeflo: You aren't capable of cause danger at all, just annoying
[23:08] Action: Zeeflo is so heartbroken now
[23:08] <timofonic> Like a child
[23:08] <timofonic> Like an antisocial nerd, that's all
[23:08] <Zeeflo> yea, ill be 9 this month, and you still suck..
[23:08] <timofonic> Get a life
[23:08] <Zeeflo> sure, Ill take a life, any life but yours.
[23:09] <Zeeflo> cause it must suck really bad!
[23:09] <Zeeflo> youre still arguing with me, what the f?
[23:09] <Zeeflo> suck it up big boy!
[23:09] <timofonic> Zeeflo: At least I know how to act properly and recognize my errors or stuff I'm ignorant about
[23:09] <Zeeflo> get pas it
[23:09] <Zeeflo> ...
[23:09] <timofonic> Zeeflo: You can be temporarily funny, because you act in a very pathetic way ;)
[23:10] <timofonic> Zeeflo: If I had OP, I would ban you
[23:10] <Zeeflo> let me know when I should cry and beg for remorse ok?
[23:10] <Zeeflo> of course you would.
[23:10] <Zeeflo> people like you end discussions in such ways.
[23:10] <timofonic> Zeeflo: You aren't capable of it, you must develop any kind of social ability first
[23:10] <Zeeflo> i got plenty, just not enough for the internet. And especially whiners like you on it.
[23:10] <timofonic> LOL
[23:11] <Zeeflo> you whine like my 5 year old girl who cant get candy from the store,,
[23:11] <Zeeflo> no..
[23:11] <Zeeflo> You whine worse..
[23:11] <timofonic> Zeeflo: I'm laughing at your childish attitude, but sad about it too :)
[23:11] <Zeeflo> are you done any time soon?
[23:11] <timofonic> Zeeflo: Poor children, I hope you mature very soon
[23:11] <Zeeflo> gues not!
[23:12] <Zeeflo> at first you pissed me off.. Now you just bore me..
[23:12] <timofonic> Zeeflo: At first you were funny, now you just bore me ;)
[23:13] <Zeeflo> Is Bureks build supporting .sup? E.g. subtitles="blabla.sup"
[23:17] <ubitux> .sup are not supported, if it's what i'm thinking about
[23:18] <ubitux> also, since it's similar to vobsub, that wouldn't be supported by the filter, even if there was a demuxer (but it could be hardcoded differently)
[23:18] <Zeeflo> whats up ubitux :)
[23:18] <Zeeflo> thats what I figured..
[23:18] <Zeeflo> .sup is some kind of blueray "vob" format or something
[23:18] <Zeeflo> bluray*
[23:23] <Zeeflo> ubitux, i have some ideas in my head, wanna give your thoughts to em?
[23:23] <Zeeflo> we stream alot of media, but to web browsers only.
[23:24] <Zeeflo> Theres services out there like Netflix, Hulu etc etc.
[23:24] <Zeeflo> They have desktop applications.. Would it be a better way to serve streaming media?
[23:24] <Zeeflo> Have some software/player developed?
[23:25] <ubitux> i don't understand
[23:25] <Zeeflo> or would the result be the same as if a user just used his webbrowser
[23:26] <Zeeflo> Like, if you got netflix. You can view their media on your webbrowser, what ever it might be. But they also got a desktop player / software of some sort..
[23:27] <Zeeflo> Would it be a better video experience for the end user to have some kind of desktop player instead of using a web browser?
[23:27] <Zeeflo> Using RTMP
[23:27] <Zeeflo> a stream is a stream.. And a player is a player, isnt it?
[23:28] <mauricio> I want to know more about rtp/rtsp
[23:30] <Zeeflo> http://en.wikipedia.org/wiki/Real-time_Transport_Protocol
[23:32] <mauricio> ok, testing using ffmpeg outputing -f rtp rtp://.....:9696
[23:32] <mauricio> I cant see anything using ffplay
[23:32] <Zeeflo> uhm
[23:33] <Zeeflo> you need to define the transport layer, dont you?
[23:33] <Zeeflo> tcp, udp etc
[23:33] <Zeeflo> im not sure though. Im not to much into RTP
[23:33] <mauricio> I believe its use tcp by default
[23:34] <Zeeflo> what about port forwarding then?
[23:34] <Zeeflo> RTP is IP to IP
[23:34] <Zeeflo> so if youre behind NAT, you would probably have to redirect the port to a WAN IP
[23:34] <Zeeflo> LAN IP
[23:34] <Zeeflo> *
[23:34] <mauricio> I'm testing inthe same computer
[23:34] <Zeeflo> got no clue then
[23:35] <Zeeflo> sorry
[23:35] <mauricio> to stream rtsp do I need a server?
[23:35] <mauricio> because rtsp fails
[23:35] <mauricio> to stream
[23:36] <Zeeflo> hmm
[23:36] <Zeeflo> i dont know if you are able to stream RTP to and from the same server/pc/whatever
[23:37] <mauricio> btw I'm trying to measure wich config will give me the lowest latency
[00:00] --- Fri Feb 22 2013
1
0
[00:17] <wm4> is there a bug report yet for VDA failing with recent ffmpeg?
[00:18] <wm4> I think I heard something about this, but I haven't found a ticket
[00:28] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:c69315a5deb0: cook: check js_subband_start for validity
[01:26] <cone-841> ffmpeg.git 03Michael Niedermayer 07release/0.8:1fd86f9a2136: vqavideo: fix return type
[01:26] <cone-841> ffmpeg.git 03Michael Niedermayer 07release/0.8:8aedb7515674: tiff: check bppcount
[01:26] <cone-841> ffmpeg.git 03Michael Niedermayer 07release/0.8:5af2fd317d84: wma: check byte_offset_bits
[01:26] <cone-841> ffmpeg.git 03Michael Niedermayer 07release/0.8:17966ae5bb43: qdm2: increase noise_table size
[01:28] <cone-841> ffmpeg.git 03Michael Niedermayer 07release/0.8:0a57df38f468: Update for 0.8.14
[04:15] <cone-841> ffmpeg.git 03Ronald S. Bultje 07master:2b10d414648b: h264: add videodsp dependency.
[04:15] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:f5c00b347dc7: avcodec_align_dimensions2: Ensure cinepak has large enough buffers.
[04:15] <cone-841> ffmpeg.git 03u-bo1b(a)0w.se 07master:4e635e10e29f: cinepak: Use correct colorspace.
[04:15] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:a3adbedf9b63: cinepak: simplify
[04:15] <cone-841> ffmpeg.git 03u-bo1b(a)0w.se 07master:8d1dd5bd5040: cinepak: More correct Cinepak decoder.
[07:48] <highgod> Hi, I want to ask a question, I want to move the opencl init function to filter module. Can I call this function at avfilter_init_filter? because I this function will compile all opencl kernel functions and create a bin file, when ran ffmpeg next time, opencl will read the bin file and do not compile the kernel again
[07:49] <highgod> it can save time
[10:22] <durandal_1707> how i pad both inputs with -filter_complex ?
[10:26] <durandal_1707> got it: they are named: [0:v], [1:v], ...
[10:46] <durandal_1707> michaelni: calling setdar/setsar with no arguments causes segv
[11:06] <cone-768> ffmpeg.git 03Ronald S. Bultje 07master:f859678f3f27: vp56: Remove clear_blocks call, and clear alpha plane U/V DC only
[11:06] <cone-768> ffmpeg.git 03Ronald S. Bultje 07master:0c0828ecc565: x86: Use simple nop codes for <= sse (rather than <= mmx)
[11:06] <cone-768> ffmpeg.git 03Ronald S. Bultje 07master:e5ffffe48d20: h264chroma: Remove duplicate 9/10 bit functions
[11:06] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:536d1ed7934d: Merge commit 'e5ffffe48d20642acc079166f0fa7d93a6a9f594'
[11:21] <cone-768> ffmpeg.git 03Ronald S. Bultje 07master:2ed008204d54: h264: Add add_pixels4/8() to h264dsp, and remove add_pixels4 from dsputil
[11:21] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:088f20a33251: Merge commit '2ed008204d5467be03a0a3af1e293b2f7038d0a0'
[11:31] <cone-768> ffmpeg.git 03Stefano Sabatini 07master:d709a18ac227: lavfi/hue: simplify parsing by making use of option shorthands
[11:31] <cone-768> ffmpeg.git 03Stefano Sabatini 07master:dd3732634528: lavfi/hue: reindent after previous commit
[11:49] <cone-768> ffmpeg.git 03Ronald S. Bultje 07master:9918f57dcfc2: get_bits: Return pointer to buffer that is the result of the alignment
[11:49] <cone-768> ffmpeg.git 03Ronald S. Bultje 07master:7ebfb466aec2: h264: Don't store intra pcm samples in h->mb
[11:49] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:98f63e1f34b3: Merge commit '7ebfb466aec2c4628fcd42a72b29034efcaba4bc'
[11:51] <cone-768> ffmpeg.git 03Ronald S. Bultje 07master:fae6fd5b8739: h264/svq3: Stop using draw_edges
[11:51] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:8c6d9ca6387f: Merge commit 'fae6fd5b873911cfcd7f7b3d72de6598af5e003a'
[11:58] <cone-768> ffmpeg.git 03Martin Storsjö 07master:26ee5ca03d76: h264: Get rid of unnecessary casts
[11:58] <cone-768> ffmpeg.git 03Diego Biurrun 07master:8a11ce43d083: build: Ensure that output directories for header objects are created
[11:58] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:39b03932c2ed: Merge commit '8a11ce43d08352f7a290355ebb5b29c495ad9609'
[12:03] <cone-768> ffmpeg.git 03Stefano Sabatini 07master:271f4112670f: lavf/matroskaenc: avoid assert failure in case of cuepoints with duplicated PTS
[12:05] <cone-768> ffmpeg.git 03Diego Biurrun 07master:c9ad5f59aa84: intreadwrite: tomi: Add missing attributes.h #include for av_always_inline
[12:05] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:f7b46d86dee8: Merge remote-tracking branch 'qatar/master'
[12:36] <ubitux> any hungarian here?
[12:39] <ubitux> mmh hungarian won't do actually
[12:41] <ubitux> Compn: would you happen to have some russian subtitles?
[12:41] <ubitux> or with a weird codepage/charset
[12:41] <ubitux> (to add a fate test)
[12:45] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:951d39fecd1f: avfilter: fix segfault with setsar/dar
[12:55] <ubitux> btw, is there a "is codec id pcm" function helper somewhere?
[12:56] <durandal_1707> nope
[12:56] <ubitux> ok thx
[13:08] <Compn> ubitux : hmmmmmmm, try opensubtitles.org for any big hollywood movie, then look for different languages and hope they arent uft ?
[13:08] <Compn> but i'll look around
[13:10] <Compn> utf*
[13:11] <Compn> ubitux : http://samples.ffmpeg.org/sub/bg.sub
[13:12] <Compn> thats probably hungarian or bulgarian
[13:12] Action: Compn has no idea :)
[13:12] <Compn> strange we didnt keep samples for different languages
[13:14] <Compn> diff subcp that is
[13:16] <ubitux> Compn: nice
[13:16] <ubitux> any idea what's the charset?
[13:16] <Compn> no
[13:16] <Compn> subcp=cp1251
[13:16] <Compn> try that
[13:17] <ubitux> i tried already
[13:17] <ubitux> koi8-r and various other cp
[13:17] <Compn> ah
[13:19] <Compn> -subcp ISO-8859-1 ?
[13:19] <Compn> i'm not actually sure what its supposed to look like :)
[13:20] <Compn> oh, -subcp windows-1251
[13:20] <Compn> that looks much better
[13:20] <ubitux> doesn't seem so, unless i'm doing something wrong
[13:21] <Compn> you try in mplayer ?
[13:21] <ubitux> no :)
[13:21] <Compn> :)
[13:21] <Compn> mr testing subtitles in a untested project
[13:21] <JEEB> subs in various text encodings remind me of a guy telling me that his English subs wouldn't show up correctly and instead showed random symbols
[13:21] <JEEB> I get a sample
[13:22] <JEEB> and I get EUC-KR Korean subtitles :D
[13:22] <Compn> lol
[13:22] <ubitux> :D
[13:22] <Compn> people think whatever subs that come with the movie are for their language
[13:22] <Compn> but most new releases are only the language that the movie is in...
[13:22] <Compn> korean movie, korean subs
[13:36] <ubitux> mmh actually it worked
[13:37] <ubitux> thx Compn
[13:50] <Compn> ubitux : so subtitle filter converts it to ass ?
[13:51] <ubitux> subtitles filter is for hardcoding
[13:51] <ubitux> not sure to understand your question
[13:52] <Compn> uhhh
[13:52] <Compn> i guess i dont know what my question was indending to ask
[13:52] <Compn> ehe
[13:52] <ubitux> @_@
[13:52] <Compn> intending*
[13:52] Action: Compn kinda distracted
[13:54] <Compn> but thats cool. subtitle encoding support
[13:54] <Compn> whats next on your list ?
[13:54] <ubitux> world domination
[13:55] <ubitux> there are some important changes to do to subtitles
[13:56] <ubitux> notably fix the ass packets (which is far from being obvious)
[13:56] <ubitux> the StyledSubtitles API (to avoid the internal representation in ASS)
[13:56] <ubitux> subtitles in lavfi...
[13:56] <ubitux> teletext
[13:56] <ubitux> and other related crazy shit :)
[14:01] <Compn> are vobsub in mkv supported ?
[14:01] <Compn> i remember something like that wasnt
[14:01] <nevcairiel> should work fine
[14:01] <Compn> ok :)
[14:12] <ubitux> <@Compn> are vobsub in mkv supported ? // you can mux and demux vobsub yes
[14:13] <ubitux> (into mkv)
[14:18] <ubitux> hey, it seems that in haskell "f $ g x $ h y $ a + b" is equivalent to "f (g x (h y (a + b)))"
[14:18] <ubitux> who is up to add this awesome syntax to ffeval ?
[14:20] <saste> ubitux, tools/ffeval -h | head -n
[14:20] <saste> ubitux, tools/ffeval -h | head -n1
[14:21] <ubitux> saste: i mean, in eval() api
[14:21] <ubitux> of course
[14:21] <ubitux> so everyone can enjoy it
[14:27] <Compn> ubitux : i thought i gave you a "windows-1251" subcp but i see "cp1251" in your fate test...
[14:27] <Compn> arent they different?
[14:27] <av500> 1251 is 1251
[14:27] <ubitux> Compn: i believe it's the same
[14:28] <av500> 850 was the non-windows one
[14:28] <Compn> oh , i guess so.
[14:29] Action: Compn should probably take a break from computer now
[15:50] <durandal_1707> saste: eval is slow
[15:51] <saste> durandal_1707, one evaluation per frame?
[15:51] <saste> it is slow in geq, when you evaluate an expression for each pixel
[15:51] <durandal_1707> that usecase is fine
[15:52] <saste> but one/few evaluations per frame are hardly measurable
[15:52] <saste> (unless you do really fancy stuff in the expression of course)
[16:04] <ubitux> saste: what do you think of making filter request window sizes?
[16:04] <ubitux> (to auto-insert asetnbsamples filters)
[16:06] <saste> ubitux, implies auto-reconfiguration
[16:06] <saste> which is still missing
[16:06] <ubitux> auto reconfig? how is that different that auto inserting scale filter etc?
[16:07] <saste> ubitux, which use case do you have in mind?
[16:07] <saste> i don't usually do things just because it is possible to do them
[16:08] <ubitux> i need a fixed granularity of samples if ebur128
[16:08] <ubitux> in*
[16:08] <ubitux> so i want ebur128 to receive fixed frames of 100ms
[16:09] <ubitux> otherwise i would need to introduce a fifo in ebur128, add a request_frame, etc and basically reproduce the asetnbsamples code
[16:09] <ubitux> which is a pain
[16:11] <cone-137> ffmpeg.git 03Michael Niedermayer 07master:2ccaab9709aa: mov: fix self referencing timecode tracks
[16:17] <ubitux> saste: so anyway, i was thinking of having such thing in the format negociation (maybe)
[16:17] <ubitux> which might be semantically incorrect
[16:18] <ubitux> the other solution might be to let the current filter responsible from auto inserting the filter itself
[16:18] <ubitux> i don't think we do that already, that might be a bit tricky
[16:18] <ubitux> and another one, call the filter internally
[16:19] <saste> ubitux, what if the filter is not compiled?
[16:19] <ubitux> error out at configuration?
[16:19] <saste> or if the filter changes its syntax?
[16:19] <ubitux> it's not a hard dep
[16:19] <saste> another option would be to factorize the code at the C-api level
[16:19] <ubitux> it's just needed/used in a particular context
[16:19] <saste> but otoh should not be so hard to use audio queue internally
[16:21] <ubitux> asetnbsamples filter code is not that obvious, duplicating its code might be pretty ugly
[16:23] <saste> you could have a filter hook which auto-inserts a filter at the configuration stage
[16:24] <ubitux> how would you do that?
[16:26] <ubitux> <@saste> another option would be to factorize the code at the C-api level // basically export all the asetnbsamples symbols? :P
[16:27] <saste> ubitux, no, i mean create an internal API which provides the correct level of abstraction
[16:27] <saste> in case this is doable
[16:28] <ubitux> btw, this may be very useful with show{spectrum,waves}
[16:28] <ubitux> to avoid flickering
[16:30] <saste> ubitux, showwaves should not need it, as it auto-adjusts itself already
[16:30] <saste> but from what i see it, the solutions are two: C-level internal API, or per-filter hooking
[16:31] <ubitux> can you detail the per-filter hooking?
[16:31] <saste> the latter could address other use cases as well, even if I can't say at the moment
[16:31] <saste> you define a callback in your filter, which is executed at some point during the configuration stage
[16:31] <saste> this may do something such as inserting a filter or such
[16:32] <ubitux> mmh sounds good, will check
[16:32] <saste> for example, earwax may auto-insert an aresample
[16:32] <saste> the problem in that case is that you don't have control over the many params of aresample
[16:33] <saste> unless you pass them to earwax, which may be ugly (and imply the usual escaping problems)
[16:33] <ubitux> what kind of aresample feature non handled by the format negociation earwax needs?
[16:33] <saste> an audiophile (the kind of guy who may want to use earwax) may be picky about the resampling algo
[16:33] <mateo`> i'm looking for some interlaced j2k/mxf samples, is there a place where i can find this kind of files ?
[16:34] <durandal_1707> saste: you picked broken filter for your explanation
[16:34] <saste> durandal_1707, then fix it! :-)
[16:34] <ubitux> mateo`: maybe ask kierank or TimNich :p
[16:35] <durandal_1707> saste: thats why there is ERROR for reson if sample rate is wrong
[16:35] <ubitux> saste: mmh well yeah ok why not
[16:36] Action: durandal_1707 gonna write another showaudio filter
[16:36] <ubitux> NIH!
[16:37] <durandal_1707> ubitux: are you saying if wrong frame size is given to ebu you get useless output?
[16:38] <ubitux> durandal_1707: nope, but the metadata will if i add the feature
[16:38] <ubitux> since you need to get information on very small slice of audio
[16:39] <ubitux> OTOH, i can also just put the normalization inside ebur128 filter...
[16:40] <ubitux> but loosing volume optimizations would be too bad
[16:41] <durandal_1707> ubitux: play ape with insane profile with ebur128
[16:41] <durandal_1707> each frame have bunch of samples
[16:42] <ubitux> we could also have a asetnbsamples=random :)
[16:42] <ubitux> that would be nice to do some testing
[16:44] <ubitux> an eval in the n field should do the trick
[16:45] <durandal_1707> i added asetnsamples (there is no asetnbsamples filter, want to write one :-/ ) and it does not help
[16:46] <ubitux> yes sorry asetnsamples
[16:46] <ubitux> what do you mean it does not help?
[16:46] <durandal_1707> i added asetnsamples=n=1024,ebur....
[16:47] <ubitux> wait, you actually have issues with ebur128?
[16:48] <durandal_1707> yes
[16:48] <saste> ubitux, also i never loved that name "asetnsamples", what about asetframesize?
[16:48] <ubitux> saste: oh please, we had this discussion thousand times :D
[16:48] <ubitux> durandal_1707: how to reproduce?
[16:49] <saste> ubitux, i always forget
[16:49] <saste> because to me it is a natural association: asetnsamples -> no good
[16:49] <saste> will send a patch then soon or later
[16:51] <Compn> that ebur128 sound sreally complicated
[16:53] <durandal_1707> ubitux: http://samples.ffmpeg.org/A-codecs/lossless/luckynight.ape
[16:56] <ubitux> durandal_1707: ffplay -f lavfi amovie=luckynight.ape looks broken
[16:56] <ubitux> sounds like lavfi has some trouble with it
[16:57] <durandal_1707> lol
[16:57] <ubitux> it doesn't even play the same thing
[16:57] <ubitux> wtf
[16:57] <durandal_1707> http://en.wikipedia.org/wiki/Goniometer_%28audio%29
[16:59] <ubitux> anyway, doesn't look like a problem in ebu
[17:00] <saste> anyone knows what's the problem with that file?
[17:00] <durandal_1707> how does agoniometer sounds?
[17:01] <ubitux> saste: the format
[17:01] <saste> i suppose may depend on the fact that it contains *huge* packets
[17:02] <saste> so may be a problem in movie or in the lavfi device
[17:02] <saste> but i bet on movie
[17:41] <durandal_1707> how this one actually works? By just ploting dots in 2dimensions when L is x-axis and R is y-axis?
[18:24] <durandal_1707> ahh i found one that use jack audio system
[18:33] <durandal_1707> you don't like my filters?
[18:50] <wm4> so uh how do I specify the video decoder with ffmpeg
[18:50] <wm4> it seems -codec:v sets the encoder?
[18:51] <durandal_1707> you set it before -i ?
[18:52] <wm4> no
[18:58] <Compn> ubitux : that subtitle seeking -ss bug is kinda important :P
[18:58] <ubitux> it's not a bug
[18:58] <ubitux> it's a feature request
[18:58] <ubitux> and not easy to solve
[18:58] <Compn> i dont think its a tough fix, dont just pass the current timepos to the sub code and it goes from there ?
[19:00] <ubitux> you can't pass it easily
[19:13] <cone-561> ffmpeg.git 03Stefano Sabatini 07master:49c8fe304f69: doc/filters: apply misc fixes to drawbox documentation
[19:13] <cone-561> ffmpeg.git 03Stefano Sabatini 07master:a2a1e20fa2c8: lavfi/decimate: use named options
[19:13] <cone-561> ffmpeg.git 03Stefano Sabatini 07master:7f07c61c2f6d: doc/filters: rework introductory description for decimate
[19:21] <cone-561> ffmpeg.git 03Stefano Sabatini 07master:3d77a275487b: lavfi/overlay: implement shortest option
[19:21] <cone-561> ffmpeg.git 03Stefano Sabatini 07master:2db0056f9680: lavfi/overlay: apply minor consistency fixes
[19:37] <saste> ubitux: libavfilter/vf_subtitles.c:313:14: warning: assignment discards const qualifier from pointer target type [enabled by default]
[19:42] <cone-561> ffmpeg.git 03Stefano Sabatini 07master:22b06103f019: lavfi/overlay: add format option
[19:46] <ubitux> huh strange i didn't spot it
[19:48] <cone-561> ffmpeg.git 03Clément BSsch 07master:9b9d996b34ae: lavfi/subtitles: fix recently introduced warning about discarded const qualifier.
[19:49] <ubitux> saste: thx
[20:05] <cone-561> ffmpeg.git 03Michael Niedermayer 07master:039f55c80333: h264: call alloc_scratch_buffers() only once linesize is known
[20:05] <cone-561> ffmpeg.git 03Michael Niedermayer 07master:da7672b20a74: h264: Fix handling of changing reference counts between slices
[21:04] <durandal_1707> ubitux: why you do not use table in swf?
[21:05] <ubitux> i could
[21:05] <ubitux> but the 16-bits bit flag is not part of the passed format
[21:06] <ubitux> i could construct one and extrand the tab
[21:06] <ubitux> not sure that's a better idea
[21:06] <ubitux> afaict the s16 in the tab are supposed to be "pcm" by the specs
[21:07] <ubitux> and the bit flag i'm using somehow describe it
[21:08] <durandal_1707> ubitux: you could use ff_get_pcm_codec_id()
[21:09] <ubitux> mmh interesting
[21:09] <ubitux> maybe a bit overkill
[21:11] <durandal_1707> ubitux: yea, not really important...
[21:11] <durandal_1707> how you found flag?
[21:12] Action: durandal_1707 wonders why this was not already done
[21:12] <ubitux> web & specs
[21:12] <ubitux> i'll re-send a patch soon though
[21:16] <ubitux> durandal_1707: i wonder if it doesn't affect the adpcm
[21:16] <ubitux> but i don't have any adpcm 8bit in swf samples
[21:16] <ubitux> only pcm 8bit, and very few
[21:19] <durandal_1707> ubitux: you cant encode one?
[21:20] <ubitux> no idea how i'm supposed too
[21:20] <ubitux> to*
[21:21] <ubitux> sounds like i likely need some windows-specific black magic tools
[21:21] <ubitux> i really don't like the dup of code in swfdec
[21:22] <ubitux> not sure to understand it very well too..
[21:22] <durandal_1707> ubitux: adpcm not mentioned in spec?
[21:24] <ubitux> the specs says (about the 16-bit flag): "This parameter only pertains to uncompressed formats"
[21:25] <ubitux> do we consider adpcm uncompressed?
[21:32] <durandal_1707> no
[21:39] <ubitux> ok
[21:39] <ubitux> i'll send a new patch set soon
[22:06] <cone-561> ffmpeg.git 03Clément BSsch 07master:9a0076f50cfc: lavf/swfdec: factorize the creation of a new stream.
[22:06] <cone-561> ffmpeg.git 03Clément BSsch 07master:624fb5f96544: lavf/swfdec: support 8-bits PCM audio.
[22:44] <jojva> Hi, I'm a student of a french university, and I'd be happy to know about how far the need to implement the MVC decoder has been brought to. Is there any decoder ? A partly functional, or even a beginning of implementation of MVC ? Cheers.
[22:47] <ubitux> D.VIL. mvc1 Silicon Graphics Motion Video Compressor 1
[22:47] <ubitux> D.VIL. mvc2 Silicon Graphics Motion Video Compressor 2
[22:47] <ubitux> ?
[22:48] <ubitux> (Decode, Video, Intra-only, Lossy)
[22:48] <jojva> Oh, sorry I should have been more precise. It's about H.264 / MVC (Multiview Video Codec), not Motion Video Compressor.
[22:50] <jojva> I can see on this thread for example, that I'm not the only one hoping for this : http://ffmpeg.org/pipermail/ffmpeg-devel/2012-June/126139.html
[22:56] <cone-561> ffmpeg.git 03Carl Eugen Hoyos 07master:c4dc6c4c86a0: Fix bits_per_coded_sample when encoding png with frame-level multithreading.
[22:56] <burek> im not sure if this is a bug, so i dont know should i put it on the bug tracker or what..
[22:56] <burek> when using -acodec aac (without any other audio encoding options), and when the audio input is like this: Stream #0:1: Audio: pcm_u8 ([1][0][0][0] / 0x0001), 8000 Hz, mono, u8, 64 kb/s
[22:57] <burek> ffmpeg i guess adds default -ab 128k
[22:57] <burek> and final stream ends up like this: Stream #0:1: Audio: aac, 8000 Hz, mono, fltp, 128 kb/s
[22:57] <burek> which causes: [aac @ 0360f720] Too many bits per frame requested
[23:00] <burek> for more reference: http://ffmpeg.gusari.org/viewtopic.php?f=11&t=825&p=1627
[23:09] <ubitux> is there a way, from a muxer, to get the packet pts exactly as it was stored originally in the demuxer?
[23:10] <ubitux> it seems i'm able to get it back with something like av_rescale(ts, stream->time_base, (AVRational){1,25})
[23:10] <ubitux> but the 1/25 is a lucky guess
[23:11] <nevcairiel> shouldnt the stream time base match the timebase used for storage in the source format?
[23:12] <nevcairiel> depends on the demuxer, i guess
[23:12] <ubitux> where is the "timebase used for storage" stored?
[23:13] <ubitux> (did i misunderstood what you said?)
[23:13] <nevcairiel> for some files, its always the same, say mpegts is always 1/90000, mkv has it in its headers somewhere, and in both cases its put in the avstream timebase and the file timestamp just read and not changed at all
[23:14] <nevcairiel> i'm sure there are exceptions that do change the timestamps
[23:37] <burek> geez, forum really got big..
[23:37] <burek> a lot of people is using it now
[23:38] <burek> i wasnt there for a couple of days and now i need to spend couple of hours to reply to all the questions :/
[23:38] <ubitux> (ok get it, i needed the tb from the stream->codec)
[23:39] <ubitux> burek: it will only get worse!
[23:39] <burek> that's the (murphy's) spirit :)
[23:41] <wm4> so, you don't have a machine where you'd be able to run a VDA FATE test or so?
[23:41] <wm4> same for other hwaccel stuff
[23:41] <wm4> though these would be harder
[23:41] <wm4> as the vda pseudo decoder works like a normal decoder
[23:52] <burek> what is ffmbc?
[23:53] <ubitux> a gpl fork of ffmpeg for broadcasting
[23:53] <burek> something like embedded ffserver or worse? :)
[23:53] <ubitux> just a ffmpeg specialized for broadcasters needs
[23:54] <burek> ok, thx :)
[00:00] --- Thu Feb 21 2013
1
0
[00:15] <ezekiel> saste: v4l2 capture with out-of-sync audio to MP4: http://ezekiel.endpoint.com/vid/sync-test-video1.txt and resulting video: http://ezekiel.endpoint.com/vid/sync-test-video1.mp4
[00:16] <ezekiel> I understand if you don't personally have time to troll through my logs
[00:17] <ezekiel> seems like I _may_ be a pretty good test case, though. Willing to rebuild and test.
[00:17] <saste> ezekiel, check that audio timestamps start with 0 as well
[00:17] <saste> video timestamps at least should be good
[00:17] <saste> -af ashowinfo
[00:19] <ezekiel> is it fairly significant that a 30fps capture would show: n:30 pts:1000068 pts_time:1.00007
[00:24] <ezekiel> wow - now that I'm looking at the ashowinfo output, I can see that the above is not nearly as significant as I might've imagined
[00:29] <saste> ezekiel, audio ts start from 0?
[00:30] <saste> also ffmpeg has a host of tools for dealing with timestamps
[00:30] <saste> check for example resample, setpts, asetpts filters just to name a few
[00:31] <saste> and the v4l2 ts option
[00:36] <stephanedev> i wrote some code to stream a video encoded with vp8 over rtp using ffmpeg C API and i play it using ffplay. when packets are dropped, many messages related to missed RTP packets are written to the console
[00:36] <stephanedev> i noticed those 2 in particular:
[00:36] <stephanedev> [vp8 @ 00a6e960] RTP: missed 1 packets
[00:36] <stephanedev> [NULL @ 00a6e960] RTP: missed 1 packets
[00:37] <stephanedev> i thought the first part (e.g. vp8) was extracted from AVClass and the part after '@' was an address
[00:37] <stephanedev> am i correct?
[00:46] <stephanedev> if my understanding is correct, what could explain the fact that the information extracted from the AVClass ("vp8") turn into "NULL"?
[00:47] <stephanedev> turns*
[00:47] <stephanedev> as obviously the address is the same in the 2 log messages above
[01:49] <defaultro> saste
[01:53] <ezekiel> still investigating, comparing the first 12 ashowinfo of out-of-sync git master: http://pastebin.com/ibCGHt76 with in-sync v1.0.4: http://pastebin.com/ymxh8jxK
[02:05] <ezekiel> recording audio and video using separate ffmpeg processes, then combining the videos, still totally out of sync
[02:05] <ezekiel> maybe an alsa issue, not a v4l2 issue
[02:08] <ezekiel> ffprobe seems to pretty regularly show "start: 0.032000" - which I know is mere tenths of a second...
[02:24] <stephanedev> is pthreads support required to stream over rtp?
[02:24] <stephanedev> i am asking because of this message when running ffmpeg on windows:
[02:24] <stephanedev> [udp @ 00645b80] 'circular_buffer_size' option was set but it is not supported on this build (pthread support is required)
[02:25] <stephanedev> using the zeranoe build
[06:51] <jagginess> hi
[06:51] <jagginess> there good guidelines for merging timing information from a video file to an audio one?
[13:24] <kvz> if i set bitrate, is that per channel, or do channels share that bitrate?
[13:24] <durandal_1707> share
[13:24] <kvz> durandal_1707: thank you!
[13:49] <robertbuhren> Hi there, i'm using a product which uses fmmpeg to transcode video files to a lower resolution/codec/container. I'm having the problem that audio and video is out-of sync, could this be anyhow related to running ffmpeg inside a virtual machine (qemu-kvm) I know the information is very vague, but maybe someone here has a hint for me
[13:50] <robertbuhren> the reason for asking is that the fact that it's run from a virtualized evinvironment is the only thing which differs from other setups that work well
[14:18] <creep> how do you like this? :) http://images.puszta.com/hazi_kifli/Hazi%20kifli%201000.jpg
[16:03] <Ziarkaen> I am running an ffmpeg screen capture command in a Makefile, for conveinience. I can run the command separately with no problem, but have to use Ctrl-C to stop recording. When this is put in the Makefile, I believe make captures the ctrl-C and exits with error. Is there a way to pass Ctrl-C directly into FFMPEG, or another way to stop recording?
[16:22] <Ziarkaen> I have solved the problem by spawning a graphical terminal running the command, although this isn;t too nice.
[16:26] <sacarasc> What do you need the capture for?
[16:50] <teratorn> Ziarkaen: Make does not intercept SIGINT
[16:50] <teratorn> Ziarkaen: the process exits with non-zero on SIGINT, so make fails at that point
[16:51] <teratorn> Ziarkaen: you can add "| true" at the end of the command in your Makefile, probably
[16:51] <teratorn> err, || true
[17:03] <n1n0> which is the command line, first to set a certain video to be used , then to set ffmpeg to extract frames at the original frame rate?
[17:36] <pedro_sland> should it make a difference if i switch the output order when i'm doing output to 2 files? eg ffmpeg -i file -filter:v ... out1.mp4 -filter:v ... out2.webm
[17:39] <relaxed> pedro_sland: using -map sure wouldn't hurt
[17:41] <relaxed> also -filter:v:$stream_id
[17:42] <pedro_sland> I appologise for not being more explicit. here is what i'm using http://pastebin.com/zFRtCeqE
[17:43] Action: pedro_sland try again
[17:44] <pedro_sland> i'll go to the beginning :) i am trying to convert one video into 2 suitable for the web. i'm happy with just webm and mp4.
[17:45] <pedro_sland> sure, i was going to give the story first?
[17:45] <relaxed> durandal_1707 loves stories
[17:46] <pedro_sland> i thought some context might help. @relaxed guessing that was sarcasm
[17:49] <pedro_sland> i checked my command http://pastebin.com/ThZ5WBk2
[17:50] <relaxed> not really, I just like messing with him.
[17:50] <relaxed> (or her?)
[17:51] <pedro_sland> k. my problem is that the webm freezes for a second at the beginning (at about 1s in).
[17:51] <pedro_sland> i started cutting down the parameters
[17:52] <pedro_sland> eventually i tried switching round the output files (and their associated commands) and the problem seems to be fixed
[17:53] <pedro_sland> The difference in output is this "[Parsed_overlay_1 @ 0x2e99260] Buffer queue overflow, dropping." line only shows when it jumps (when webm is last)
[17:53] <relaxed> pedro_sland: try, ffmpeg -i input -map 0 (first output options) $first_output -map 0 (second output options) $second_output
[17:55] <relaxed> oh yeah, remove 99% of those options and use "-t 5" for both outputs to get a 5 second sample.
[17:56] <relaxed> until you iron this out
[17:56] <pedro_sland> yep, that's what i was doing hence the first paste.
[17:56] <pedro_sland> it was fine without the filter but the filter broke it
[18:02] <durandal_1707> does it happen if you write only webm file?
[18:03] <pedro_sland> i think i tested that but i'll test again just now
[18:03] <pedro_sland> http://pastebin.com/GQYwvpBA
[18:03] <relaxed> pedro_sland: did you check out ffmpeg from an old repo? Your version is much older than mine: ffmpeg version N-49946-ga08ebf0
[18:03] <pedro_sland> oh
[18:03] <pedro_sland> hmm, i'll check
[18:03] <relaxed> git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg
[18:04] <relaxed> they switched at some point
[18:04] <relaxed> once you update, look at -filter_complex in the man page.
[18:06] <pedro_sland> i've moved that repo and downloaded another. compiling now zzzz
[18:08] Action: relaxed sticks a beer under burek's nose
[18:09] <durandal_1707> relaxed: how you found out that pedro_sland is using old version?
[18:10] <relaxed> his pastebin, http://pastebin.com/ThZ5WBk2 , shows "ffmpeg version N-38907-g39b0393"
[18:10] <pedro_sland> i downloaded a new one anyway but looking at it i don't think it is old. last commit i have is 9d9420d2 at Wed Feb 20 11:54:06 2013 +0100
[18:11] <relaxed> 11,000 commits behind
[18:11] <pedro_sland> :O
[18:12] <pedro_sland> something fishy here. .git/config says [remote "origin"] url = git://source.ffmpeg.org/ffmpeg
[18:12] <pedro_sland> nvm almost done compiling
[18:14] <durandal_1707> relaxed: how you get that number?
[18:14] <pedro_sland> latest version should say "ffmpeg version git-2013-02-20-2ccaab9" right?
[18:15] <durandal_1707> yes
[18:15] <pedro_sland> :)
[18:15] <durandal_1707> and previous build you used was just few commits older
[18:17] <pedro_sland> that's what i thought i was using - a build from earlier today but nvm
[18:17] <relaxed> durandal_1707: hmm, I thought the number in N-38907-g39b0393 was the number of commits. Am I mistaken?
[18:18] <durandal_1707> relaxed: yes you are
[18:20] <pedro_sland> i ran the broken command again and same behaviour http://pastebin.com/UqKXRBgv. checking with just webm
[18:21] <pedro_sland> that worked fine http://pastebin.com/ya30Zgvk
[18:22] <relaxed> durandal_1707: then what is it?
[18:25] <durandal_1707> relaxed: last commit hash
[18:25] <durandal_1707> (partial one)
[18:25] <relaxed> isn't this the commit hash --> g39b0393
[18:29] <durandal_1707> relaxed: without g
[18:37] <pedro_sland> relaxed: thanks for the -filter_complex hint. i imagine that this way only loads the overlay once and "movie=" loads it 2x? other than looking nicer on the command line, are there more benefits of this way?
[18:41] <relaxed> I thought it was required with 2 or more inputs, but I guess the old way still works for now.
[18:42] <pedro_sland> well, i just tested, this way works properly unlike the other way :)
[18:42] <pedro_sland> ty much for your help. i really appreciate it.
[18:43] <relaxed> unless you read the man page from time to time they'll change things right under your nose.
[18:43] <pedro_sland> :)
[18:43] <relaxed> you're welcome
[18:45] <pedro_sland> if they're going to remove that other way, does that mean that we won't be able to have different filters on different streams? not that i want that but just curious.
[18:56] <relaxed> pedro_sland: no, -filter_complex is just a better implementation for use with multiple inputs.
[18:57] <durandal_1707> s/better/red
[18:58] <relaxed> red?
[18:59] <pedro_sland> i don't get it either, unless he/she's playing with you
[19:00] <relaxed> I think she meant "rad"
[19:00] <relaxed> which it totally is
[19:00] <pedro_sland> totally. i love it already
[19:03] <pedro_sland> thanks again and bye all
[19:45] <Emmanuel_Chanel> hi
[19:45] <Emmanuel_Chanel> Someone is alive?
[19:46] <Emmanuel_Chanel> I want to encode Columbo: Friend in Deed aired on BSP. But the audio is one track with ual monaural.
[19:47] <Emmanuel_Chanel> It has only 1 Stereo sound track that has Japanese "left" and English "right". How can I split the track into two, Japanese and English?
[19:55] <relaxed> Emmanuel_Chanel: look at -map_channel in the man page
[19:57] <relaxed> Emmanuel_Chanel: there are example commands
[20:06] <someperson> anyone happen to know anything about video editing ? particularly automated skin softening/retouching ? just wondering what tools i need to look at
[20:07] <Emmanuel_Chanel> Yeah, I did. But I couldn't get a good result. Anyway, ok, I try. Thanks.
[20:09] <durandal_1707> Emmanuel_Chanel: use filters
[20:12] <durandal_1707> Emmanuel_Chanel: see documentation for channelsplit
[20:19] <Aaronds> Hi, I'm trying to convert a video to .mp4 using ffmpeg CLI. However, at the moment my video plays normally in Firefox but in Chrome I only get audio, no video. I'm using -pix_fmt yuv420p as I've seen on the net that this is needed for chrome, but otherwise I have no idea what is causing the problem. Can anyone help?
[20:19] <Aaronds> I'll paste command/output shortly
[20:23] <durandal_1707> Aaronds: pasted anything?
[20:23] <Aaronds> Command & output for the aforementioned question: http://pastebin.com/fE51jgVX
[20:24] <Aaronds> IRC bugged out on me briefly
[20:25] <durandal_1707> does chrome plays aac?
[20:25] <Aaronds> durandal_1707, sorry aac?
[20:25] <JEEB> yeah
[20:25] <JEEB> H.264 + AAC
[20:25] <JEEB> in mp4
[20:26] <Aaronds> I don't know... AFAIK others have done the same conversion and had everything worked, the only problem I picked up on was the yuv240p thing.
[20:26] <Aaronds> obviously H.264 is fine
[20:29] <durandal_1707> maybe it support only certain samplerates?
[20:30] <durandal_1707> it is little strange because AFAIK chrome use libavcodec
[20:30] <JEEB> but he is getting the audio :D
[20:30] <Aaronds> durandal_1707, are you able to provide a set of parameters which you'd expect to work?
[20:31] <JEEB> that should in all theory work
[20:31] <Aaronds> >_<
[20:31] <JEEB> are you testing on chromium or chrome_
[20:31] <JEEB> ?
[20:31] <Aaronds> chrome
[20:31] <JEEB> weird
[20:33] <Emmanuel_Chanel> durandal_1707: How shall I fix this command for that? ffmpeg -i ${FILENAME}.ts -t 00:03:00 -acodec libfaac -aq 100 -vcodec libx264 -preset ultrafast -crf 24 -vf hqdn3d,crop=in_w:in_h-8:0:0,scale=1280:720,unsharp -aspect 16:9 -threads 0 -map 0:0 -map 0:1 -map 0:2 -metadata:s:1 language=jpn -metadata:s:2 language=eng -async 1000 ${FILENAME}test1.mp4
[20:36] <Aaronds> Oh, the issue is with chrome... Going to try and have to use a different file format for now then -.-
[20:39] <durandal_1707> Emmanuel_Chanel: try to add "-af channelsplit" before -vf
[20:40] <Emmanuel_Chanel> ok.
[20:42] <Emmanuel_Chanel> I got "Cannot map stream #0:2 - unsupported type." with "ffmpeg -i ${FILENAME}.ts -t 00:03:00 -acodec libfaac -aq 100 -vcodec libx264 -preset ultrafast -crf 24 -af channelsplit -vf hqdn3d,crop=in_w:in_h-8:0:0,scale=1280:720,unsharp -aspect 16:9 -threads 0 -map 0:0 -map 0:1 -map 0:2 -metadata:s:1 language=jpn -metadata:s:2 language=eng -async 1000 ${FILENAME}test1.mp4"
[20:44] <durandal_1707> Emmanuel_Chanel: replace "-af" with "-filter_complex"
[20:48] <Emmanuel_Chanel> ffmpeg -i ${FILENAME}.ts -t 00:03:00 -acodec libfaac -aq 100 -vcodec libx264 -preset ultrafast -crf 24 -filter_complex channelsplit -vf hqdn3d,crop=in_w:in_h-8:0:0,scale=1280:720,unsharp -aspect 16:9 -threads 0 -map 0:0 -map 0:1 -map 0:2 -metadata:s:1 language=jpn -metadata:s:2 language=eng -async 1000 ${FILENAME}test1.mp4
[20:48] <Emmanuel_Chanel> says: Cannot map stream #0:2 - unsupported type.
[20:48] <Emmanuel_Chanel> ...
[20:51] <Emmanuel_Chanel> ok.
[21:10] <Emmanuel_Chanel> durandal_1707: thx. It seems that I can encode TS with Dual Monaural now.
[21:12] <durandal_1707> Emmanuel_Chanel: how? what happened?
[21:12] <Emmanuel_Chanel> It seems that -filter_complex channelsplit helps.
[21:15] <Emmanuel_Chanel> durandal_1707: Where on the document can I see channel_split , the option?
[21:20] <durandal_1707> Emmanuel_Chanel: http://ffmpeg.org/ffmpeg-filters.html
[21:24] <Emmanuel_Chanel> thx.
[22:53] <burek> beer! :D \o/
[22:53] Action: burek bows to relaxed :)
[23:00] <burek> wieshka, trouble in paradise? :)
[23:02] <burek> does anyone use ffplay extensively enough to answer this q: http://ffmpeg.gusari.org/viewtopic.php?f=26&t=826
[23:02] <ubitux> burek: hey long time no see.
[23:02] <burek> it's related to controlling ffplay programatically
[23:02] <burek> ubitux hello :)
[23:02] <burek> it was a busy holiday week :)))
[23:02] <Mavrik> holiday week? O.o
[23:03] <burek> couple of holidays merged together in a whole week :D
[23:03] <ubitux> burek: ffplay has no "slave" mode like MPlayer has
[23:03] <ubitux> burek: redirect him to MPlayer/mplayer2/mpv
[23:03] <burek> ok thanks
[23:04] <burek> is there any interesting news around here :)
[23:05] <beastd> ubitux: didn't mpv drop slave-mode? or has it something different now?
[23:05] <ubitux> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=Changelog;hb=HEAD ?
[23:05] <ubitux> beastd: mmh good question
[23:06] <ubitux> beastd: yeah right seems they dropped it
[23:06] <ubitux> so MPlayer/mplayer2 then
[23:30] <burek> halp :) again :)
[23:30] <burek> "How can i change the "scan type" into interlaced? How can i change the "scan order" (Bottom Field First - Top Field First) fot interlaced encoding?"
[00:00] --- Thu Feb 21 2013
1
0
[00:29] <wm4> there are no FATE tests for seeking, right?
[00:43] <ubitux> wm4: fate/seek.mak?
[00:48] <wm4> ok
[00:51] <wm4> that doesn't seem to test decoding
[00:51] <wm4> i.e. whether the packets are valid?
[00:52] <wm4> or maybe the packet size is always good enough for this purpose?
[01:14] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:c6f59b95c529: h264: avoid calling get_format() multiple times
[01:14] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:56b6909b395e: movenc: hotfix, dont store fiel for h264 / mpeg4-asp / dnxhd
[01:14] <cone-967> ffmpeg.git 03Diego Biurrun 07release/1.1:7acfa7758c7f: configure: Make warnings from -Wreturn-type fatal errors
[01:25] <cone-967> ffmpeg.git 03Diego Biurrun 07release/1.1:1ca25bc38775: libopencore-amrnb: cosmetics: Group all encoder-related code together
[01:25] <cone-967> ffmpeg.git 03Diego Biurrun 07release/1.1:e492818d8925: libopencore-amr: Conditionally compile decoder and encoder bits
[01:25] <cone-967> ffmpeg.git 03Diego Biurrun 07release/1.1:a23d6ea1e443: libopencore-amrwb: Make AMR-WB ifdeffery more precise
[01:25] <cone-967> ffmpeg.git 03Diego Biurrun 07release/1.1:6c62098827d3: build: The libopencore-amrnb encoder depends on audio_frame_queue
[01:25] <cone-967> ffmpeg.git 03Luca Barbato 07release/1.1:b9a287f23700: build: make audio_frame_queue a stand-alone component
[01:25] <cone-967> ffmpeg.git 03Matti Hamalainen 07release/1.1:d61c6ebccfde: svq3: unbreak decoding
[01:25] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:0f5a0a415510: Merge remote-tracking branch 'qatar/release/9' into release/1.1
[01:35] <cone-967> ffmpeg.git 03Hendrik Leppkes 07release/1.1:e14564b9262d: lavfi/kerndeint: use av_pix_fmt_desc_get instead of directly accessing the table
[01:35] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:f5955d9f6f9f: targa: Fix y check in advance_line
[01:35] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:1ea5bbc5940d: sanm: add forgotten check for decoded_size in old_codec37()
[01:35] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:1400f1a1e46d: sanm: Use the correct height variable in the decoded_size checks
[01:35] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:f6687bbb6464: pngdec/filter: dont access out of array elements at the end
[01:35] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:75211f2b8cfb: tiff: Check buffer allocation and pointer increment more carefully in shorts2str() and double2str()
[01:36] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:811a504c6bc2: shorten: dont leave invalid channel counts in the context.
[01:36] <cone-967> ffmpeg.git 03Michael Niedermayer 07release/1.1:7c40a0449b47: swr: check channel layouts before using them.
[01:50] <BBB> michaelni: ok, fixed for that one crash at least
[01:50] <BBB> michaelni: it's hard for me to run complete fate, but if you find more crashers, let me know and I'll fix them too
[01:50] <BBB> michaelni: seems to work for me now
[03:01] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:a3a97129e79d: mpegvideo: make ff_print_debug_info independant of mpegvideo
[03:01] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:d41efc1f267c: h264: put visualization and debug support back
[03:41] <BBB> michaelni: where do I find svq3_watermark.mov?
[03:51] <Compn> should be listed in allsamples.txt
[03:51] <Compn> http://samples.ffmpeg.org/mov/watermark/svq3_watermark.mov
[03:51] <Compn> BBB : ^^
[03:53] <BBB> ty
[03:53] <BBB> I don't know how allsamples.txt works
[03:56] <michaelni> http://samples.ffmpeg.org/allsamples.txt should list all samples so one can easily grep in it
[03:59] <Compn> ehe
[03:59] <Compn> BBB : you dont have a copy of the samples repo on your hd ?
[03:59] <Compn> michaelni : how many tests do you run on the samples? i mean, how did you find out svq3_watermark.mov had a problem? :)
[04:00] <BBB> Compn: how much HD space do you think I have?
[04:00] <BBB> I have a little laptop that is easily portable at home, that's all
[04:00] <Compn> BBB : last i checked it was only ~60gb
[04:00] <Compn> 64G unified/
[04:01] <BBB> my hd generally is full of pictures, music and stuff
[04:01] <BBB> you know, I have kids and stuff, I have ... baggage, as they'd say
[04:01] <Compn> no problem :)
[04:01] <Compn> ssd arent there yet with the size
[04:01] <BBB> I'll just download them as needed
[04:05] <BBB> hm right the 8x8 only clears one half of the block data
[04:05] <BBB> let me try to fix that
[04:07] <BBB> ok, that's fixed now, now idct8_add4 seems broken also
[04:07] <BBB> blegh
[04:11] <BBB> ok h264 fixed
[04:11] <BBB> will test svq3 in a bit, may well be fixed by the same thing
[04:12] <BBB> yeah looks mostly the same to me now
[04:12] <BBB> michaelni: care to test again?
[04:22] <BBB> oh nm the crcs are different
[04:22] <BBB> lemmecheck
[05:45] <highgod> Hi, I want to ask a question. When will the next ffmpeg release version be published? where can I get the time schedule about ffmpeg release version be published? thanks
[05:50] <Compn> no idea :)
[05:51] <Compn> maybe michaelni knows release schedule
[05:51] <Compn> usually after a bunch of bugs are fixed, and new bugs are merged in
[05:59] <highgod> @Compn:thanks, does ffmpeg will publish a version on march? I have read the comments, and I have some quesions, and I have sent the email.
[06:04] <BBB> michaelni: nice bug you detected there in svq3_watermark.mov
[06:04] <BBB> congrats on that
[06:27] <highgod> michaelni:do you know when the next ffmpeg release version be published?
[06:27] <highgod> Thanks
[10:10] <ubitux> highgod: please fix your mailer and reply below each comment
[10:10] <ubitux> currently you're duplicating the whole mail, without quoting, so the previous mail is mixed with your answer
[10:10] <ubitux> it's unreadable
[10:52] <durandal_1707> ubitux: should i sent new blend patch or just commit?
[10:53] <ubitux> saste is more familiar than me with the multiple input thing
[10:53] <ubitux> you may want to ask him before
[10:53] <ubitux> or maybe Nicolas
[12:05] <michaelni> highgod, a new major release every 3 month (more or less) minor releases randomly
[12:16] <cone-841> ffmpeg.git 03Paul B Mahol 07master:f2d200d460ab: lavfi/overlay: yuv444p & yuva444p support
[12:16] <cone-841> ffmpeg.git 03Paul B Mahol 07master:03b711d95e28: tmv: initialize unused pallete entries with 0
[12:21] <highgod> Thanks
[12:21] <highgod> @ubitux:I got it. thanks
[12:24] <cone-841> ffmpeg.git 03Vicente Jimenez Aguilar 07master:202b5f6deb65: doc: Fix some obsolete references to av* tools as ff* tools
[12:24] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:4e469842b525: Merge commit '202b5f6deb65e405b07b9b5c20f97c8cb925cf49'
[12:36] <cone-841> ffmpeg.git 03Daniel Kang 07master:7a03145ed7cb: x86: dsputil: int --> ptrdiff_t for ff_put_pixels16_mmxext line_size param
[12:36] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:cf10616cc0db: Merge commit '7a03145ed7cb4f1ce794b5126559dd6f38029243'
[12:41] <highgod> @ubitux:should I just write the replies under the comments of the orignal mail?
[12:54] <mateo`> hello, i'm trying to fix the frame duplication issue i have with interlaced jpeg2000 patch. Since the decoder output a single frame for 2 video packets, is there a proper way to correct the pts/dts of the decoded frame so that ffmpeg won't duplicate frames ? (-vsync 0 does the trick)
[12:55] <michaelni> mateo`, sounds like something sets wrong timestamps
[12:55] <michaelni> did you look at the demuxer timestamps ? and the framerate choosen ?
[12:56] <mateo`> yes pkt_pts and pkt_dts seems ok within the decoder
[12:58] <mateo`> i tried to devide by 2 the pkt_pts and pkt_dts within the decoder when it outputs a full frame but it does nothing
[12:58] <mateo`> (maybe i'm not supossed to do that in the decoder)
[12:59] <cone-841> ffmpeg.git 03Daniel Kang 07master:9acd23d655b5: x86: dsputil: Fix h263 loop filter link error in some configurations
[12:59] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:fa09ad5c9e68: Merge remote-tracking branch 'qatar/master'
[12:59] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:b9237aa7b090: x86/h263_loopfilter: Fix author attribution after code has been moved/splited around
[15:12] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:b97d61f924ee: avfilter/ff_merge_formats: only merge if doing so does not loose chroma or alpha
[15:16] <Compn> this bug looks invalid, AWS problem not ours ... https://ffmpeg.org/trac/ffmpeg/ticket/1486
[15:26] <Compn> yay i closed a bug :)
[15:31] <durandal_1707> nooooooooooooooooooooooooooooooooooooo
[16:02] <ubitux> durandal_1707 ?
[16:21] <cone-841> ffmpeg.git 03Ronald S. Bultje 07master:a1f1ca96b426: svq3: fix decoding residual blocks of b-frames.
[16:43] <cone-841> ffmpeg.git 03Ronald S. Bultje 07master:1acd7d594c15: h264: integrate clear_blocks calls with IDCT.
[16:50] <Compn> ubitux : i think he was making a joke about me closing a bug
[16:51] <ubitux> oh. my bad.
[17:02] <wm4> why does linking to libav* add -SDLmain?
[17:03] <durandal_1707> because of sdl in lavd?
[17:03] <wm4> but why -SDLmain?
[17:03] <wm4> this is completely retarded
[17:04] <wm4> I wonder why you'd ever need -lSDLmain at all
[17:04] <wm4> it's a crap hack
[17:04] <wm4> I almost feel like adding a horrible hack to configure to filter it out
[17:04] <durandal_1707> i don't see it anywhere
[17:05] <durandal_1707> perhaps that is what pkgconfig returns
[17:05] <wm4> yes, it is
[17:05] <wm4> hm maybe it's recursively added from the SDL .pc
[17:05] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:633f9974790e: bmp: check available space when reading palette
[17:06] <wm4> indeed it is
[17:06] <wm4> got to report this as SDL bug
[17:06] <wm4> because... seriously.
[17:08] <durandal_1707> i don't have that here with old SDL (one from FreeBSD ports though)
[17:11] <wm4> I get it with 1.2.15
[17:11] <wm4> but well, it's a windows-only thing
[17:11] <wm4> so it won't happen with a unix native libsdl
[17:11] <wm4> just mingw packages
[17:12] <durandal_1707> same version here
[17:12] <durandal_1707> how would you describe lavfi in few words?
[17:13] <wm4> "new ways of hiding semi-useful functionality under awkward APIs"
[17:15] <durandal_1707> that is too much words, i need 5 max
[17:21] <Compn> lol
[17:22] <Compn> durandal_1707 : powerful filters with confusing syntax
[17:22] <Compn> :D
[17:22] <Compn> sorry, i am too honest :)
[17:22] <Compn> i love the filters :)
[17:22] <Compn> i dont want to spend 20 minutes to figure how to write the right command line tho
[17:23] <Compn> the examples are good, i want more of them.
[17:23] <wm4> even Compn (mplayer and mencoder veteran) thinks the syntax is confusing
[17:24] <Compn> by confusing i mean, most people only know 8th grade math (and probably lower than that) so creating equations is difficult
[17:25] <Compn> the filters are complex, so they need to have the ability to provide details like that. but most users wont be able to use them as-is
[17:25] <Compn> without some kind of frontend, script, examples, etc
[17:26] <Compn> i never understood custom matrices in mpeg encoding either...
[17:26] Action: Compn not the smartest
[17:28] <wm4> it just means it's not userfriendly
[17:31] <Compn> j-b : is there someone working on the g2m decoder?
[17:31] <Compn> j-b : kostya updated his blog http://codecs.multimedia.cx
[17:45] <durandal_1707> what about those cinepak patches?
[17:51] <divVerent> Compn: what annoys me most about the filters is how the tags work
[17:51] <divVerent> as long as your filters are a single chain, you can use comma to avoid most tag use
[17:52] <divVerent> but it is quite annoying that to insert a filter into the middle of e.g. [in] foo [mid]; [mid] [in2] bar [out]
[17:52] <divVerent> you suddenly have to RENAME a tag
[17:53] <divVerent> and e.g. turn it into [in] foo [new]; [new] added [mid]; [mid] [in2] bar [out]
[17:53] <divVerent> this is especially bad if you want to programmatically edit filter chains
[17:53] <divVerent> of course, in this case, comma syntax helps too ([in foo, added [mid]; [mid] [in2] bar [out]
[17:53] <divVerent> )
[17:53] <divVerent> but that can't be used in the general case
[17:54] <divVerent> so to add a filter before [tag], you need to rename existing OUTPUT uses of [tag] to [newtag], then add [newtag] addedfilter [tag]... the "rename" part here is annoying
[17:56] <divVerent> however, having said that this syntax sucks... I have no constructive idea about doing it better :(
[17:56] <ubitux> it looks like you are complaining in a situation where you have a variable foo and when wanting to add a similar variable you need to rename it to foo0 and add foo1
[17:56] <ubitux> :))
[17:56] <durandal_1707> using single line for just this is wrong
[17:56] <divVerent> ubitux: in C this issue is simple to solve :P
[17:57] <divVerent> just add functioncall(...) around the expression
[17:57] <divVerent> but C syntax doesn't help us for filters, as a function can only have one return value there
[17:57] <divVerent> I wonder if a Haskell-style "let" would actually help :P
[17:58] <durandal_1707> script reader
[17:59] <wm4> and then there's this idea to use Lua to build the filtergraph
[17:59] <divVerent> https://gist.github.com/4987734
[17:59] <divVerent> actually, the concept of Haskell's "let" would help
[17:59] <divVerent> wm4: does Lua fix the issue, is the question
[18:00] <divVerent> basically, the idea of this "let" use is to have a way to insert filters at a point by JUST inserting text
[18:00] <durandal_1707> wm4: using single line to describe filtergraph is silly
[18:00] <wm4> divVerent: it seems vsynth is doing fine with using Python for building graphs
[18:00] <divVerent> durandal_1707: that is mostly the issue
[18:00] <divVerent> wm4: wonder if it has the same issue, for dynamically generated graphs
[18:00] <divVerent> or mkanually edited graphs
[18:01] <durandal_1707> where are devs when you need them
[18:02] <divVerent> haha, C/Perl-like syntax WOULD help a great deal already
[18:02] <divVerent> https://gist.github.com/4987753
[18:02] <BBB-work> michaelni, opinions on https://github.com/rbultje/ffmpeg/commit/be3d3b499bd241b7d6f7d9b4b6d91cb2dc… ?
[18:02] <divVerent> i.e. clear distinction between input and output args
[18:02] <divVerent> in C done by &, in perl done by creating the variables using "my", ...
[18:02] <divVerent> this way, the filter insertion can be done by pure text operations
[18:02] <divVerent> which is basically my goal
[18:02] <wm4> well, Lua and python can return tuples
[18:03] <divVerent> right, but we still need a way to easily programmatically build a filter graph
[18:03] <divVerent> e.g. to add the subtitle renderer only if subtitles are present
[18:03] <Compn> durandal_1707 : well the patch author just said the code for rejecting strips could be moved to detecting incomplete blocks... i think someone said we shouldnt make a regression by not playing some files due to his patch
[18:03] <Compn> durandal_1707 : so it would be nice if his patch doesnt affect the playing of (possibly broken) files
[18:03] <saste> divVerent, lua scripting will be possibly a gsoc task i'd like to mentor
[18:04] <Compn> will the patch author do that or will someone else have to is the question
[18:04] <saste> otherwise i'll tackle it myself as soon as i'll have some time (not before a few months)
[18:04] <Compn> saste : should i start writing gsoc 2013 page ?
[18:04] <saste> Compn, wait for llogan
[18:04] <divVerent> wm4: BTW, actually found now that my "ffstuff" can't handle my test case :P
[18:04] <saste> he asked mike if we should use multimediawiki.cx
[18:04] <divVerent> or... it can? why did it work at all then
[18:04] <BBB-work> actually let's send that to the ml
[18:05] <Compn> saste : thats what we've always used ....
[18:05] <divVerent> AH, because one tag is defined later :P
[18:05] <saste> if we use trac we should make sure that can't be hacked by a random user
[18:05] <divVerent> that's the sole reason why it even works+
[18:05] <Compn> trying to avoid conflict or something ?
[18:05] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:a34418c28e0a: tiff: check bppcount
[18:05] <BBB-work> if you want to see h264 w/o mpegvideo and dsputil, that three patches finish it
[18:05] <divVerent> because in ffmpeg filter graphs, it's hard to detect by text checks whether a tag is an input or an output
[18:05] <saste> Compn, i don't know if mike is still actively maintaining the wiki
[18:06] <saste> if he's not, he could be non responsive for granting access to new contributors
[18:06] <Compn> oh
[18:06] <saste> so i'd say let's wait a day or two
[18:06] <Compn> who needs new wiki contribs ? :P
[18:06] <Compn> students claim on the ml, devels change it on wiki
[18:06] <Compn> but ok
[18:06] <merbanan> BBB-work: did you really untie the Gordian knot ?
[18:07] <saste> Compn, i don't know who has access to the (multimedia.cx) wiki
[18:07] <Compn> merbanan : he also fixed svq3 breaking too :)
[18:07] <saste> also we don't want to depend on it if mike is not maintaining it
[18:07] <Compn> saste : http://wiki.multimedia.cx/index.php?title=Special:Listusers
[18:07] <Compn> hehe
[18:07] <Compn> yeah i got it
[18:08] <saste> e.g. in case of attacks or spamming
[18:08] <Compn> somewhat trolling
[18:08] <BBB-work> merbanan, it doesn't compile yet if you actually make dsputil files selectable, but it's close
[18:08] <saste> that's why we should wait his reply
[18:08] <BBB-work> merbanan, what's left is the constants in dsputil_mmx (pw_N)
[18:08] <Compn> BBB-work : did you run benchmarks on it ?
[18:09] <Compn> oh i guess that doesnt change too much, nm
[18:10] <michaelni> BBB-work, lots of ifdefs in be3d3b499bd241b7d6f7d9b4b6d91cb2dce5f5fe
[18:11] <michaelni> not really pretty
[18:11] <michaelni> does it have some users / usecase ?
[18:12] <ubitux> i guess the reason is to have a config without the dspcontext need
[18:16] <BBB-work> michaelni, chrome would use it
[18:16] <BBB-work> michaelni, note how it removes dspcontext from h264
[18:16] <BBB-work> michaelni, that's a major size reduction
[18:17] <BBB-work> merbanan, and secondly, the zigzag scan table permutators; thirdly, hpeldsp is still in there (used by vp3); fourthly, all the rectangular fullpel mc functions are currently in dsputil, but shared between chromamc (bilin), h264/mpeg4qpel, vp8mc, hpeldsp; all this needs to be made more generic for it to be truly finished
[18:17] <BBB-work> merbanan, but it's almost there
[18:18] <BBB-work> michaelni, might be worth doing the same for mpegvideo-derived codecs also; a lot of cases, you don't want er, you just want to error out
[18:19] <nevcairiel> why would you want to error out instead of trying to mask an error? too risky that there are other bad errors in there?
[18:21] <BBB-work> showing a corrupt frame only makes sense if the transport mechanism somehow allows that to have any meaning
[18:21] <BBB-work> e.g. multi-slice frame, one slice corrupt: all is fine
[18:21] <BBB-work> single-slice frame, one slice corrupt: yeahright
[18:21] <BBB-work> vp8 same with partitions
[18:21] <nevcairiel> doent er also cover single macroblock corruptions?
[18:21] <BBB-work> over tcp, if the file is corrupt, it means the source was corrupt, so you want to error out if you're chrome
[18:22] <BBB-work> over udp, the stream commonly becomes corrupt because of natural transport issues
[18:22] <BBB-work> so it all depends on the use case
[18:22] <BBB-work> for chrome, we don't want er; error means error out
[18:22] <nevcairiel> not that ffmpegs ER is the best in the world anyway, my GPUs hardware decoder does a better job at covering things up in some clips =p
[18:23] <BBB-work> plus er is slow, introduces extra security risks (more code to cover), is not as well tested, doesn't support >8bpp, I'm not even sure it supports non-420
[18:23] <BBB-work> isn't integrated with ffvp8
[18:23] <BBB-work> there's tons of issues with it
[18:23] <BBB-work> we'd prefer to not use it
[18:23] <michaelni> is there a big disadvantage over making ER optional while leaving most things in h264.c so there are few ifdefs?
[18:24] <BBB-work> no, that should be fine
[18:24] <BBB-work> I just did the ifdef route so I know I fixed all users
[18:24] <BBB-work> but I can just add if (CONFIG_ER) around the relevant calling blocks and leave the assignments that don't require er symbols in place
[18:24] <BBB-work> I don't care about 100bytes of unused ERContext memory in H264Context
[18:25] <michaelni> if(CONFIG_ER) sounds alot better than ifdefs
[18:25] <BBB-work> sure, can adapt
[18:25] <BBB-work> we might need to make ER explicitely settable in configure than
[18:25] <BBB-work> since currently, --disable-everything --enable-decoder=h264 will disable er
[18:25] <BBB-work> with no way to enable it other than --enable-decoder=vc1 or so
[18:26] <BBB-work> which isn't quite ideal
[18:26] <nevcairiel> cant you make it enabled by default if a decoder is selected which uses it, unless overriden at command line?
[18:26] <BBB-work> I don't know enough about configure to know how to do that
[18:27] <ubitux> with the _select vars maybe
[18:27] <michaelni> about lack of >8bpp support, it also lacks support for interlaced concealment, i suspect these are why the GPU is better, our error concealmentz was actually quite good in the cases that it supports when i last compared to other decoders but thats long ago
[18:28] <michaelni> interlace, >8bpp support should of course be added ...
[18:28] <nevcairiel> my clip was 420 8bpp
[18:28] <nevcairiel> looks like a slice was failing
[18:29] <nevcairiel> ffmpeg overwrote the previous reference frame with a flat color more or less, while the gpu decoder just seemed to try to interpolate some motion vectors on top of it, which looked OKish because there wasnt much movement
[18:29] <ubitux> (btw, it would be nice to fix all the uninitialized read issues recently introduced...)
[18:29] <ubitux> libav might have not realized it because their valgrind+undef is broken
[18:30] <michaelni> nevcairiel, probably makes sense to open a ticket with sample file and screenshoot
[18:30] <nevcairiel> i'll put it on my list of things todo for a rainy day
[18:31] <nevcairiel> at least i still know which sample it was
[18:33] <michaelni> nevcairiel, also try with threads 1, could be a thread issue
[18:34] <michaelni> on that rainy day thart is
[18:34] <nevcairiel> i suppose
[18:34] <nevcairiel> i'll try to remember this
[18:34] <nevcairiel> lately i'm happy to have time to sleep
[18:37] <BBB-work> it's certainly true er is broken wiht threading
[18:38] <BBB-work> mpeg-er has ff_thread_await_progress() indicators
[18:38] <BBB-work> h264-er doesn't, they're explicitely marked as FIXME
[18:38] <BBB-work> lack of a fate test means I couldn't be sure it worked or not so I just left it out
[18:38] <BBB-work> (i.e. yes we need a h264-er test)
[18:39] <nevcairiel> just out of curiosity, does the spec outline how missing slices for example should be handled?
[18:41] <BBB-work> no
[18:41] <BBB-work> hm does implement something for it, I believe
[18:41] <BBB-work> but you're not required to follow that, from what I understand
[18:41] <BBB-work> jm, not hm
[18:41] <BBB-work> lol
[18:43] <durandal_1707> stupid spammer
[18:43] <BBB-work> I also wonder if I should make ercontext embed dspcontext, so the caller doesn't have to set it anymore
[18:45] <durandal_1707> how to permanently remove files uploaded to trac?
[18:46] <durandal_1707> saste: gonna comment on blend filter?
[18:47] <saste> durandal_1707, yes in a few hours, please wait
[18:59] <michaelni> slavon888 Smartmil8 smartmil888(a)gmail.com
[18:59] <michaelni> IP address: 93.84.19.159
[19:05] <michaelni> BBB-work, for frame threading with EC not only does the EC code need to wait for its references to be available subsequent frames also need to wait as soon as a slice is missing or damaged, i think they dont currently
[19:06] <saste> i'm going to implement an annoying feature in overlay
[19:06] <michaelni> slice threading should work with EC though
[19:08] <durandal_1707> saste: what that means?
[19:11] <saste> durandal_1707, that people will suffer because of that
[19:11] <durandal_1707> saste: what feature?
[19:11] <saste> durandal_1707, surprise, but you can easily guess what
[19:12] <durandal_1707> i lost my pil
[19:12] <saste> no more teasers
[19:13] <BBB-work> michaelni, no they do (at least for mpeg1/2)
[19:13] <BBB-work> michaelni, for h264 they probably don't, you may be right
[19:20] <cone-841> ffmpeg.git 03Paul B Mahol 07master:76415a942051: configure: fix libavfilter pkgconfig description
[19:21] <michaelni> mpeg1/2 do only slice threads
[19:23] <michaelni> also theres the problem that a error in MB x,y can produce the first detectable error a row or more later
[19:23] <michaelni> (not in mpeg2 which is a special case here)
[19:25] <BBB-work> maybe it's mpeg4
[19:25] <BBB-work> it's the one covered by the -error fate test
[19:26] <BBB-work> I don't know which mpeg it is, they all look so much alike (and are - honestly - all equally useless)
[19:31] <michaelni> yes, seems mpeg4 has a simple check
[20:37] <saste> ubitux, durandal_1707: how hard would be to make showspectrum output a frame only when it is completely filled?
[20:38] <saste> this should allow to compose a global spectrum image
[20:54] <durandal_1707> saste: aren't you lavfi expert?
[20:55] <durandal_1707> saste: fix your connection
[20:55] <durandal_1707> by calling ff_filter_frame only once
[20:56] <durandal_1707> lol
[20:57] <saste> damn
[20:59] <durandal_1707> saste: got any of my replies?
[21:01] <saste> i got: by calling ff_filter_frame only once
[21:02] <saste> i tried to look at the code, and changing the logic for outputting just one filled frame doesn't look trivial
[21:02] <saste> that's why i asked
[21:03] <nevcairiel> michaelni: i took two screenshots of the EC behaviour on this clip http://screenshotcomparison.com/comparison/8966 .. while you can see that there is something wrong in the GPU decoded image, its by far not as obvious (and even less so when the image is moving) .. and the second one is even worse: http://screenshotcomparison.com/comparison/8968
[21:03] <nevcairiel> its single threaded, so not related to threading in this case
[21:04] <durandal_1707> saste: why it is not trivial?
[21:06] <nevcairiel> a user actually asked me once for this particular clip if avcodec was broken, because he couldnt see the corruption with the gpu decoder, and thought it might be a software issue =p
[21:06] <nevcairiel> i will upload the clip and open a ticket, i guess
[21:07] <saste> durandal_1707, why? because of the caching logic, which is not immediate to grasp
[21:14] <ubitux> between two timeouts, feel free to tell him "no" to answer his question
[21:15] <ubitux> damn it's not answering the question
[21:15] <ubitux> then "hard at 5%"
[21:16] <durandal_1707> ^ what that last one means?
[21:16] <ubitux> it means it's not hard
[21:17] <durandal_1707> than do it before saste comes ;)
[22:04] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:ce153eef8ffd: vc1dec: dont decode slices when the slice header failed to decode.
[22:22] <durandal_1707> saste: fixed connection?
[22:23] <saste> durandal_1707, don't know, it doesn't depend on me
[22:23] <saste> let's see if it lasts
[22:24] <durandal_1707> saste: what caching logic you see in showspectrum?
[22:25] <saste> durandal_1707, in showwaves i send the frame only when I fill the frame
[22:25] <durandal_1707> so?
[22:25] <saste> with showspectrum it is different, since the frame is updated at each audio frame, right?
[22:25] <saste> so it is different
[22:26] <durandal_1707> i dont think updating frame is good idea (even you could store frame data in another buffer)
[22:26] <durandal_1707> generally see what sox does
[22:27] <saste> durandal_1707, sox is not meant for audio visualization
[22:27] <saste> i want that feature mostly to be able to generate a single image containing the audio spectrum
[22:29] <durandal_1707> saste: sox does just that
[22:30] <durandal_1707> for some duration - can be whole file
[22:30] <saste> durandal_1707, will you implement that feature of should I look at it?
[22:30] <durandal_1707> but lavfi do not know what is duration of file yet, but this does not stop you to implement it
[22:31] <durandal_1707> saste: i plan to work on exr and filters (blend, stereo3d, noise, etc ...)
[22:32] <saste> durandal_1707, i don't mind, as long as I can script it and use tile and scale if necessary
[22:32] <saste> my problem is that don't want overlapping frames
[22:32] <durandal_1707> you return frame at EOF
[22:35] <durandal_1707> saste: those doxygen issues you mentioned are present in geq/overlay too
[22:35] <durandal_1707> i just copy-pasted it
[22:36] <saste> durandal_1707, i know
[22:36] <durandal_1707> saste: can av_image_copy() copy only one plane?
[22:37] <saste> durandal_1707, av_image_copy_plane
[22:37] <durandal_1707> nice
[22:39] <saste> durandal_1707, i was sleeping when reviewing your overlay patch
[22:40] <saste> working on a fix right now
[22:40] <durandal_1707> saste: i had conversation on issue with michaelni
[22:40] <durandal_1707> there are two solution: elegant and less elegant one
[22:40] <saste> an ugly fix
[22:41] <saste> yes I'm working on the less elegant/simpler
[22:41] <durandal_1707> elegant is very simple
[22:41] <durandal_1707> and makes filter simpler
[22:42] <saste> durandal_1707, how?
[22:42] <saste> does it involve scaling?
[22:42] <durandal_1707> what you mean with more explicit check for inputs dimensions?
[22:42] <durandal_1707> saste: no, using single pix fmt list
[22:42] <saste> i meant a more explicit *error message*
[22:42] <saste> like the one in concat
[22:43] <saste> durandal_1707, i don't understand about overlay
[22:43] <saste> what I'm doing now is to select the "format", either yuv420, yuv444, rgb
[22:44] <saste> it is a generalization of the rgb option
[22:44] <durandal_1707> removing slow rgba code handling different rgbs in inputs
[22:45] <durandal_1707> i see no point to allow inputs to be of different pix fmts
[22:45] <saste> durandal_1707, the overlay is usually alpha
[22:45] <saste> why do you need to add alpha to the main input?
[22:47] <durandal_1707> than API have limitations
[22:48] <durandal_1707> there should be way to force inputs to have same subsampling and etc
[22:48] <durandal_1707> but anyway rgba handling is funny
[22:48] <saste> how?
[22:49] <durandal_1707> converting both inputs to same one would be faster than conversion code in filter
[22:49] <durandal_1707> because you would do just memcpy in filter
[22:50] <durandal_1707> saste: also why if i give single images as inputs to blend filter i get nothing in output?
[22:50] <durandal_1707> looks like there is of by 1 error somewhere
[22:51] <durandal_1707> eg, using 2 images for inputs give 1 output frame
[22:51] <saste> durandal_1707, two images for *both* inputs?
[22:51] <saste> or two images in total?
[22:52] <saste> also I see no synchronization mechanism in your code
[22:52] <durandal_1707> two images in total = 0
[22:52] <saste> that's what makes overlay more complex
[22:52] <durandal_1707> saste: why would i need to add one (same code in overlay)?
[22:52] <durandal_1707> i used alphamerge, and it does not have this issue
[22:53] <saste> alphamerge is supposed to be used on synched inputs
[22:53] <saste> but i don't mind, that can be done on top of your patch
[22:53] <saste> ideally the synching code should be shared
[22:53] <durandal_1707> so i just add setpts?
[22:54] <saste> durandal_1707, i suppose you have this situation
[22:54] <saste> 1, 1, EOF, 2
[22:54] <saste> 1, EOF, 2
[22:54] <saste> or something like that
[22:55] <saste> when the first EOF is found, it exists
[22:55] <saste> *exits
[22:55] <saste> so you have no chance to blend the two frames
[22:55] <durandal_1707> could be
[22:56] <durandal_1707> so if EOF is found on first input but not on second i just return EAGAIN?
[22:56] <saste> setpts can't help in your case since it depends on how frames are cached on the filtergraph
[22:56] <saste> durandal_1707, the complete logic is in overlay, so that should be ideally the model to follow
[22:57] <saste> see if nicolas wants to help, since he proposed to write common utilities for input synching
[22:59] Action: llogan should have picked a less generic name
[23:16] <cone-841> ffmpeg.git 03Michael Niedermayer 07master:a5153b1d1692: shorten: Fix signedness of comparission
[23:39] <wm4> how do you force a video codec with ffplay?
[23:41] <saste> wm4: -vcodec?
[23:41] <wm4> Unrecognized option 'vcodec'
[23:41] <wm4> Failed to set value 'h264' for option 'vcodec'
[23:41] <wm4> I tried that first
[23:46] <michaelni> wm4 -codec:v h264
[23:53] <wm4> runtime CPU detection is enabled by default, right?
[23:54] <BBB-work> not for libpostproc I think
[23:55] <BBB-work> then again, libpostproc...
[23:55] <wm4> yeah, nobody cares about libpostproc
[23:55] <wm4> and isn't it full of 2001 code anyway?
[23:58] <saste> wm4, how old is your ffplay?
[23:58] <wm4> saste: oh, that's the distro version, probably a bit older
[23:59] <saste> -vcodec should work, i fixed it some months ago
[23:59] <wm4> yeah, it works with the git version
[00:00] --- Wed Feb 20 2013
1
0
[01:00] <teratorn> is command like this possible? ffplay -f rawvideo -pixel_format yuv420p -s 352x288 '/tmp/%04d.yuv420p.raw'
[01:24] <dionoea> /win 0
[03:45] <relaxed> teratorn: I think only the image2 format can use printf's %d naming as input.
[03:46] <relaxed> but, you can cat raw yuv in as input.
[03:49] <defaultro> hey guys, how can we mimic a warm filter?
[03:50] <defaultro> it's different from hue filter
[03:51] <defaultro> or maybe another solution is to superimpose a png image with 50% or less for alpha
[03:51] <defaultro> is that possible?
[08:53] <mista_D> anyway to run libass on same file using two srt files (eng and spa)? without retranscoding twice?
[09:14] <juanfont> hi. Is there any way to reduce max_analyze_duration at runtime? or discard those 5M of data? I'm trying to reduce the lag between server and client as much as I can
[09:14] <juanfont> thanks :)
[09:29] <eli___> hello
[09:33] <eeli_> hello
[09:33] <eeli_> would you please help me `md?
[09:34] <eeli_> what can i do for can not chatting freenode channel?
[10:52] <jjido> Hello, I learnt that .VOB files in a VIDEO_TS folder are MPEG2. Do you know if it is possible to replace the video and audio streams in it to occupy less space? I want to do what DVDShrink does to fit a 2H DVD on a 4.7GB disk
[11:51] <invariant> I have a wav file which is the result of a cdrip. I want to turn that in a sequence of flac files. How can that be done?
[11:53] <relaxed> you need to rip the cd into a sequence of wav files
[11:55] <invariant> relaxed, alternatively, how can I make the wav file available as a rippable device?
[11:56] <invariant> I don't get why all the rippers work on a device file myself.
[11:56] <relaxed> Does it have a cue sheet?
[11:59] <invariant> relaxed, yes
[11:59] <invariant> relaxed, I have a cue file.
[12:00] <relaxed> Then you need software to extract the tracks using that file. ffmpeg can't do it.
[12:01] <durandal_1707> ffmpeg can do it but not in friendly way like some tools designed to do just that
[12:02] <relaxed> invariant: mplayer2 has cue://$file support, use that with -dumpstream to extract a wav for each track.
[12:04] <durandal_1707> http://www.etree.org/shnutils/shntool/
[12:04] <durandal_1707> it have cue mode
[12:05] <relaxed> real men use mplayer
[12:05] Action: relaxed flexes
[12:15] <durandal_1707> this one can create cue files
[12:41] <Aktau> Hey guys. I've been using ffmpeg to decode videos very well, but I'd like to know if it does (maybe through libavcodec) support HW accelerated decode through VA-API on linux?
[12:41] <Aktau> Does anyone know about that?
[14:00] <JEEB> WmA, sending me private messages for nothing that needs to be private or is specifically related to me will only lead to /ignore :P
[15:35] <otters> being a total noob, what option do I pass to map_metadata to convert a FLAC to Apple Lossless and keep per-track metadata intact (like album name etc.)
[15:37] <durandal_1707> they are usually completly copied and encoder is added
[15:38] <otters> so adding map_metadata is redundant?
[15:38] <durandal_1707> dunno, did you tried it ? does it work as expected?
[15:38] <otters> yeah, I tried it, and the tracks don't have *any* metadata when I open them in itunes
[15:39] <otters> oh, hey, yeah, omitting the option works.
[15:40] <otters> I must be some sort of moron
[15:46] <otters> thanks everyone
[15:53] <e2b> Is there any plan to support the Raspberry Pi GNU-acceleration for en-/decoding? The CPU itself is pretty slow, but thanks to the GPU it can already play HD videos witch optimized players.
[15:55] <durandal_1707> no plans, people usually come with patch that implements such funcionality - that is how FOSS works
[15:58] <divVerent> if "nm -a binary" shows an entry:
[15:58] <divVerent> 000000010087ca70 T _avcodec_default_execute
[15:58] <divVerent> does that mean I have the right to the source code of the binary? ;)
[15:59] <divVerent> I found this in "iTap Mobile RDP.app" for Mac OS X
[15:59] <divVerent> *hint* want to bring that Hall of Shame back?
[16:00] <durandal_1707> only if you have enough money to sue them
[16:00] <divVerent> putting them on the HoS costs no money :P
[16:00] <durandal_1707> and pay lawyers and etc
[16:00] <divVerent> also, they violated none of MY copyright
[16:01] <divVerent> so I wouldn't have a case there
[16:01] <divVerent> but I mainly am wondering, did I read the "nm" output right and this is not an external reference?
[16:01] <divVerent> I THINK I did
[16:02] <divVerent> also, can I somehoe figure out if they used ffmpeg or libav?
[16:02] <divVerent> ah, it's PROBABLY libav
[16:02] <divVerent> I see an URL with upload.libav.org
[16:38] <MIG-> e2b: raspbian includes source code for x264 encoding using the gpu
[16:42] <jeje34> Hi to all ;-)
[16:43] <MIG-> hello
[16:44] <jeje34> I'm want to use FFMPEG library in a windows application to decode H264 video streams from camera. I have a question about the memalign-hack for the configure compilation. I'm not sure if it needed on Windows OS
[16:46] <JEEB> if it is enabled by default in the configuration for your way of building for windows on the current ffmpeg, then it is either needed or not
[16:46] <JEEB> you shouldn't have to set it manually
[16:46] <jeje34> yes but in the configure file there: --enable-memalign-hack emulate memalign, interferes with memory debuggers
[16:47] <jeje34> So if I use Visual Studio VC++ debuggers, it can make me memory problems?
[16:47] <JEEB> I repeat
[16:47] <JEEB> you shouldn't have to enable it manually
[16:48] <JEEB> configure should look it up by the way you are compiling it and for which arch
[16:48] <jeje34> yes but you tell me it is enables by default
[16:48] <JEEB> DEPENDS ON THE CONFIGURATION AND WHAT YOU ARE BUILDING WITH
[16:48] <JEEB> AS I SAID
[16:48] <jeje34> I use msys and mingw to compile FFMPEG on my windows
[16:48] <JEEB> look at the gawd damn configure script's output
[16:48] <JEEB> seriously
[16:49] <JEEB> and it shouldn't make the thing any worse debug-wise
[16:49] <JEEB> if it is needed, it is needed
[16:49] <JEEB> if it isn't, it isn't
[16:49] <JEEB> that's it
[16:49] <JEEB> if you are building with mingw MSVS's debugger won't be able to load the damn debug symbols anyways, if you are building with MSVC then you are getting the debug symbols if you enable debug
[16:50] <jeje34> I use Visual Studio to make my application, but I use the library of FFMPEG I built and don't try to debug "inside" FFMPEG
[16:51] <jeje34> just use debugger to debug my application
[16:52] <JEEB> go over what I said, the answers to your question(s) are there
[17:14] <ozone89> hello :)
[17:21] <jeje34> I'm sorry but there is something I can't explain in my code. I have an AVPacket m_lpPacket and an AVPicture m_lpPict. In order, I call m_lpPict.data[0] and after decoding I use avpicture_alloc( &m_lpPict, AV_PIX_FMT_RGB32, dwWidth, dwHeight); to use swscale to convert my YUV420 to RGB32. But at the next frame, when I call av_init_packet(&m_lpPacket), m_lpPict.data[0] become NULL.
[17:41] <mpfundstein> jeje34: didnt know people still use hungarian notation
[17:44] <jeje34> I find it more clear when writing classes
[17:58] <fenduru> How can I avoid artifacts between segments when segmenting an audio file?
[17:59] <fenduru> I need to be able to play them back sequentially without gaps/clicks between them
[18:36] <sabton> hey guys, just got caught up and built for Android at head release/1.1, after making necessary adjustments for deprecated API everything is running stable... except certain codecs (ogg/mp3) are decoding to garbled audio; however FLAC for example is decoding perfect
[18:36] <sabton> does this immediately indicate anything to anyone?
[18:48] <JEEB> sabton, sounds like you're expecting the decoder to output interleaved int while it is giving you planar float?
[18:48] <JEEB> although if flac works then it's the float vs int difference
[18:48] <JEEB> most lossy decoders were switched to output float by default, as that's how it basically gets encoded
[18:49] <JEEB> lossless stuff like flac naturally should output exactly what is put in
[18:49] <durandal_1707> flac did not changed sample format
[18:49] <JEEB> yes
[18:49] <durandal_1707> so it is GrEaT sample format switch
[18:49] <JEEB> I already noted it
[18:49] <JEEB> if you didn't see my second line :P
[18:50] <JEEB> and then the third+fourth noting the fact that the sample format was changed for most lossy formats
[18:50] <JEEB> sabton, ffmpeg does provide you a library to then convert the float audio to integer audio that you probably want
[18:51] <durandal_1707> and from planar to interleaved
[18:51] <JEEB> yup, if that is needed
[18:53] <sabton> ooh
[18:55] <sabton> that sure does explain things
[18:55] <sabton> the sound of my garbled audio is consistent with that explanation
[18:56] <sabton> so in addition to converting the samples, i guess the codecs will tell me what sample format is used?
[18:56] <sabton> i would need to find that API as well
[18:56] <JEEB> yes, I would think the avframe tells you what the output format is
[18:57] <sabton> ah, thank you
[18:57] <sabton> man, im playing with compiler settings and such.. =\
[18:57] <sabton> thanks :)
[19:13] <Dmole> is there a repo rpm with a recent ffmpeg with all libs enabled or is everyone compiling for some reason?
[19:17] <sophie_> hi
[19:17] <Dmole> what's p?
[19:17] <sophie_> do you know it's possible to run ffmpeg with ubuntu 11.04 ?
[19:18] <Dmole> ye it is.
[19:18] <Dmole> *yes
[19:18] <sophie_> i have a problem with yasm
[19:18] <sophie_> i have difficulties to upgrade it
[19:19] <JEEB> there shouldn't really be anything hard in upgrading it
[19:19] <JEEB> build it and put it somewhere in your PATH
[19:20] <Dmole> might need to purge from apt before did you try compiling yasm?
[19:20] <sophie_> yes
[19:20] <Dmole> so what exactly is the issue?
[19:20] <sophie_> terminal tell me :
[19:20] <sophie_> Found yasm 1.1.0.2352
[19:20] <sophie_> Minimum version is yasm-1.2.0
[19:20] <sophie_> If you really want to compile without asm, configure with --disable-asm.
[19:21] <JEEB> that just means that you have 1.1.0 installed
[19:21] <sophie_> but i've already done apt-get upgrade yasm
[19:21] <JEEB> yes, because the repositories don't have 1.2.0
[19:21] <JEEB> you have to grab the tarball
[19:21] <JEEB> extract it
[19:21] <sophie_> so, i can also unabled i ?
[19:21] <sophie_> it
[19:21] <JEEB> what
[19:21] <Dmole> disable
[19:22] <JEEB> no, don't disable asm
[19:22] <JEEB> that will be SLOW
[19:22] <sophie_> ok
[19:22] <sophie_> the tarball ?
[19:22] <JEEB> http://www.tortall.net/projects/yasm/releases/yasm-1.2.0.tar.gz
[19:22] <JEEB> wget that
[19:22] <JEEB> tar xvf yasm[tab]
[19:22] <JEEB> cd yasm-[tab]
[19:23] <JEEB> do you want it to be available to everyone in the system automatically?
[19:23] <JEEB> or are you OK with a local install?
[19:23] <sophie_> local install it's ok
[19:24] <JEEB> ./configure --prefix=${HOME}/ownapps
[19:24] <JEEB> make
[19:24] <JEEB> make install (no sudo needed)
[19:24] <sophie_> ok, i try
[19:24] <JEEB> export PATH=${HOME}/ownapps/bin:${PATH}
[19:24] <JEEB> then try what yasm --version gives
[19:25] <JEEB> if that works, then you can add that thing into your dot-bashrc file
[19:25] <JEEB> so that directory gets added to the PATH every time
[19:35] <sophie_> thank you JEEB
[19:35] <sophie_> it has worked!
[19:41] <sophie_> JEEB what do you mean by "add that thing into your dot-bashrc file"
[19:41] <sophie_> ?
[19:41] <sophie_> you mean these phrase :
[19:41] <sophie_> export PATH=${HOME}/ownapps/bin:${PATH}
[19:42] <JEEB> yes, that thing. dot-bashrc is a file that gets loaded every time bash gets loaded :P
[19:42] <JEEB> basically so you don't have to run it every time you start a terminal
[19:43] <JEEB> manually
[19:43] <sophie_> and where it is ?
[19:44] <JEEB> /home/your_username/ is where .bashrc should reside
[19:44] <JEEB> (that file name is dot-bashrc)
[19:45] <JEEB> although you generally should have one I guess
[19:45] <sophie_> when i want to ./configure ffmpeg
[19:45] <sophie_> it tells me :
[19:45] <sophie_> :~/ffmpeg-source/x264$ ./configure
[19:45] <sophie_> ./configure: ligne 241: rm : commande introuvable
[19:45] <sophie_> ./configure: ligne 243: dirname : commande introuvable
[19:45] <sophie_> ./configure: ligne 245: grep : commande introuvable
[19:45] <JEEB> uh-oh
[19:45] <JEEB> you did something wrong :P
[19:45] <sophie_> Unknown system , edit the configure
[19:45] <sophie_> yes
[19:45] <sophie_> i know
[19:45] <JEEB> your original ${PATH} got derped
[19:46] <sophie_> i just pick some command on the net
[19:46] <sophie_> when it wasn't good
[19:46] <sophie_> what can i do ?
[19:46] <sophie_> reinstall my all system ?
[19:47] <JEEB> no...
[19:47] <JEEB> you can just close the terminal and open a new one :P
[19:47] <sophie_> :)
[19:49] <sophie_> i was on this site :
[19:49] <sophie_> http://buildall.wordpress.com/2011/05/09/how-to-compile-and-install-ffmpeg-…
[19:50] <sophie_> perhap's it will be more easy by installing new version of ubuntu ?
[19:50] <JEEB> no
[19:50] <JEEB> also that "guide" is a few years old :P
[19:51] <JEEB> https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide use this if you want to use some damn guide :P
[19:51] <JEEB> also this installs yasm on the whole system
[19:52] <sophie_> yes, it's ok too
[19:53] <sophie_> This guide supports Ubuntu 12.10 (Quantal Quetzal), Ubuntu 12.04 (Precise Pangolin), and Ubuntu 11.10 (Oneiric Ocelot).
[19:54] <sophie_> but i have version 11.04
[19:54] <sophie_> natty
[19:54] <JEEB> shouldn't matter too much in most cases
[19:54] <sophie_> it will work either ?
[19:54] <sophie_> ok
[19:54] <JEEB> there are some things you actually will have to think about, yes
[19:54] <JEEB> now you're not afraid of thinking, right?
[19:55] <JEEB> just try to understand what the fuck is going on instead of mindless copy-pasting shit and you will be effing fine
[19:56] <sophie_> yeah!
[19:56] <sophie_> you know how to speak my godness!
[20:00] <JEEB> Do pardon me on my french, but I was kind of trying to think how to explain some very basic *nix things to you, but ended up just not thinking about it because it made my head hurt, and I'm not here to teach basics of *nix
[20:12] <Dmole> or if you don't want to think stick to apt
[20:29] <reactormonk> how do I drop the video stream? -vcodec null doesn't work
[20:30] <JEEB> if you want no video at all in the output, -vn
[20:30] <reactormonk> thanks
[20:45] <WmA> can anyone tell me what I am doing wrong please?
[20:45] <WmA> http://pastebin.com/zZpSwrEq
[20:46] <WmA> I would like to pipe a http ts stream into libx264
[20:46] <WmA> to record it straight to mp4
[20:49] <relaxed> WmA: why wouldn't you use the stream as ffmpeg's input?
[20:58] <stephanedev> hi everybody. i wrote a program using ffmpeg C API which streams a video encoded with vp8 over rtp. when streaming the video with ffmpeg and playing it with ffplay and simulating a 5% datagram loss, the video still plays ok and most frames are not too degraded.
[20:59] <stephanedev> but when using my program, it works fine if no datagram is lost, but with a 5% loss, ffplay shows a much more degraded video with most frames showing big pixel blocks or sometimes unrecognizable images
[21:00] <stephanedev> of course i am not surprised that ffmpeg is doing a much better job than my program, but i would like to understand what it does to achieve that
[21:01] <stephanedev> i tried to play with gop size, bitrate and other settings, but to no avail
[21:04] <stephanedev> does anyone have any hint on what i could try to improve my streaming code?
[21:10] <stephanedev> is there any parameter in AVCodecContext or AVFormatContext which would help?
[22:15] <vulkanr> hello. need some help with libavfilter. anyone in the house?
[22:15] <saste> vulkanr, don't ask to ask, just ask
[22:16] <vulkanr> trying to transcode video and pass the raw frames though libavfilter. i am getting really bad quality output. i think i am doing something wrong with the frame pts.
[22:17] <vulkanr> looking at the output file - the frames seem to be ordered by the pts which is weird and not the case when not using libavfilter
[22:19] <vulkanr> i followed the example code and am converting the pts between timebases as showed
[22:34] <ezekiel> in recent versions of ffmpeg, has the "lowpass"(http://ffmpeg.org/ffmpeg-filters.html#lowpass) filter been removed? Or perhaps replaced with something better?
[22:35] <saste> ezekiel, it has been added one week ago or so
[22:35] <ezekiel> okay, so only master ?
[22:35] <saste> yes
[22:35] <ezekiel> saste: much appreciated!
[22:36] <ezekiel> I've been building from git, but I've had to step back a few versions to achieve good audio+video sync from v4l2
[22:36] <saste> ezekiel, how old? we recently fixed a few v4l2 issues
[22:36] <ezekiel> 1.0.4 works great - 1.1.X is out of sync :(
[22:37] <saste> ezekiel, try master
[22:38] <ezekiel> I will try again and hopefully come back with some better info
[23:19] <ezekiel> eck, bad news for me - build from master still doesn't capture synchronized audio
[23:33] <ezekiel> to AVI: http://pastebin.com/HHddvm8W
[23:36] <rsumi> hello, i'm trying to build mediainfo on my box but i'm getting the following errors - http://pastebin.com/ZEJ9sQNA - how can i fix this?
[23:38] <llogan> rsumi: this is #ffmpeg. mediainfo is not from the FFmepg project.
[23:38] <llogan> ezekiel: N-36755-gce153ee does not appear to be current git head
[23:39] <rsumi> llogan: yes i know but i thought it's likely that someone from ffmpeg has ran into this issue. thanks anyways
[23:39] <llogan> we usually just use ffprobe instead
[23:39] <ezekiel> hmm, git log tells me - commit ce153eef8ffdd10ead74ad4646eb784be1c5b107
[23:40] <saste> ezekiel, how can be sure that the timestamps are synched?
[23:40] <llogan> current is N-50103-ga5153b1
[23:41] <saste> use setpts/showinfo to make sure
[23:41] <llogan> rsumi: did you run ./autogen before configure?
[23:41] <ezekiel> llogan: yeah, that commit was pushed while I was building - so now my build is one commit behind
[23:42] <llogan> close enough, but your N- number is odd
[23:42] <llogan> i mean weird
[23:43] <rsumi> llogan: no. i just did now and i get these errors
[23:43] <rsumi> http://pastebin.com/gXNcxc94
[23:44] <llogan> i don't know. make sure you have libtool, automake, autoconf as build deps and libmediainfo as a depends.
[23:45] <llogan> other than that i don't know, especially for PPC.
[23:47] <rsumi> llogan: yeah i have all that stuff except for libmediainfo. i suspect libmediainfo is what i get after building
[00:00] --- Wed Feb 20 2013
1
0
[00:05] <BBB> michaelni: have you tested h264 w/ and w/o edges (i.e. w/ and without draw_edges) lately (since I optimized ff_emulated_edge_mc)?
[00:06] <BBB> michaelni: it's the last dsp function (not counting er)
[00:07] <BBB> michaelni: I'd very much fancy killing it
[00:09] <michaelni> Any tests i did where in the distant past i think
[00:10] <BBB> hm ok I'll do some testing
[00:10] <BBB> on vp8, not using draw_edge was never slower and sometimes faster
[00:10] <BBB> (only tested on x86 ofc)
[00:10] <BBB> I think the same would hold for arm also though
[00:11] <ubitux> yay blend filter.
[00:11] <ubitux> Paul is on steroids recently
[00:12] <nevcairiel> i think he is on a mission to kill mpcodecs
[00:12] <ubitux> yes, that's good, but afaict blend is not related to it :)
[00:12] <ubitux> same for all the awesome audio filters
[00:21] <Compn> gasp!
[00:21] <Compn> hes gone!
[00:21] <Compn> :p
[00:31] <cone-485> ffmpeg.git 03Paul B Mahol 07fatal: ambiguous argument 'refs/tags/n0.8.13': unknown revision or path not in the working tree.
[00:31] <cone-485> Use '--' to separate paths from revisions
[00:31] <cone-485> refs/tags/n0.8.13:HEAD: lavfi/noise: switch to AVLFG noise generator
[00:34] <BBB> yeah h264 is definitely faster with -flags +emu_edge on x86
[00:34] <BBB> (on cathedral)
[00:34] <BBB> so same as vp8
[00:34] <BBB> I'll kill draw_edges - fun fun fun
[00:53] <cone-485> ffmpeg.git 03Michael Niedermayer 07master:54b2bddd22fb: v4l2: try to fix build on BSD
[01:09] <cone-485> ffmpeg.git 03Mans Rullgard 07release/0.7:b102d5d97dae: h264: allow cropping to AVCodecContext.width/height
[01:09] <cone-485> ffmpeg.git 03Mans Rullgard 07release/0.7:0054d70f23ed: mov: set AVCodecContext.width/height for h264
[01:09] <cone-485> ffmpeg.git 03Carl Eugen Hoyos 07release/0.7:c497d71a026a: Clarify that -passlogfile has a different syntax when used with -vcodec libx264.
[01:10] <cone-485> ffmpeg.git 03Ronald S. Bultje 07release/0.7:9a5e81235e62: dxva2: include dxva.h if found
[01:10] <cone-485> ffmpeg.git 03Carl Eugen Hoyos 07release/0.7:8582e6e9a3ce: Fix muxing mjpeg in swf. (cherry picked from commit 7680d99b4302e476076cc1b8f2567f47c2aaef4d)
[01:10] <cone-485> ffmpeg.git 03Anton Khirnov 07release/0.7:a60eb6ef12df: ffmpeg: fix -force_key_frames
[01:10] <cone-485> ffmpeg.git 03Kostya Shishkov 07release/0.7:0173a7966b33: vc1dec: add flush function for WMV9 and VC-1 decoders
[01:10] <cone-485> ffmpeg.git 03Janne Grunau 07release/0.7:f31170d4e7f9: nuv: check RTjpeg header for validity
[01:10] <cone-485> ffmpeg.git 03Janne Grunau 07release/0.7:8812b5f16410: imgconvert: avoid undefined left shift in avcodec_find_best_pix_fmt
[01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:899d95efe12f: mpegvideo: Don't use ff_mspel_motion() for vc1
[01:10] <cone-485> ffmpeg.git 03Anton Khirnov 07release/0.7:77d43bf42d76: lavf: don't segfault when a NULL filename is passed to avformat_open_input()
[01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:b6ba39f931a8: alsdec: check opt_order.
[01:10] <cone-485> ffmpeg.git 03Mina Nagy Zaki 07release/0.7:b6c5848a1f8f: lavfi: avfilter_merge_formats: handle case where inputs are same
[01:10] <cone-485> ffmpeg.git 03Justin Ruggles 07release/0.7:61ece4137298: vorbisenc: check all allocations for failure
[01:10] <cone-485> ffmpeg.git 03Alex Converse 07release/0.7:d6e250abfc36: vorbis: Validate that the floor 1 X values contain no duplicates.
[01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:e28814e0e16c: Merge remote-tracking branch 'qatar/release/0.7' into release/0.8
[01:10] <cone-485> ffmpeg.git 03Diego Biurrun 07release/0.7:7b91e52eb9fd: x86: Require an assembler able to cope with AVX instructions
[01:10] <cone-485> ffmpeg.git 03Clément BSsch 07release/0.7:80b8dc30dc96: lavc/ass_split: check for NULL pointer in ff_ass_split_override_codes().
[01:10] <cone-485> ffmpeg.git 03Mans Rullgard 07release/0.7:a2ae183a382f: h264: allow cropping to AVCodecContext.width/height
[01:10] <cone-485> ffmpeg.git 03Kostya Shishkov 07release/0.7:e39fc137aeac: vc1dec: add flush function for WMV9 and VC-1 decoders
[01:10] <cone-485> ffmpeg.git 03Anton Khirnov 07release/0.7:d3e2f35f7add: bmpdec: only initialize palette for pal8.
[01:10] <cone-485> ffmpeg.git 03Max Lazarov 07release/0.7:0892a6340f86: eval: fix swapping of lt() and lte()
[01:10] <cone-485> ffmpeg.git 03Martin Storsjö 07release/0.7:a81c1ea2eb66: h263: Add ff_ prefix to nonstatic symbols
[01:10] <cone-485> ffmpeg.git 03Janne Grunau 07release/0.7:99008ba3667a: rv34: use AVERROR return values in ff_rv34_decode_frame()
[01:10] <cone-485> ffmpeg.git 03Mina Nagy Zaki 07release/0.7:10c244cc89e8: lavfi: avfilter_merge_formats: handle case where inputs are same
[01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:7a0ff7566b66: avsdec: Set dimensions instead of relying on the demuxer.
[01:10] <cone-485> ffmpeg.git 03Aneesh Dogra 07release/0.7:42c3a3719b05: bytestream: add a new set of bytestream functions with overread checking
[01:10] <cone-485> ffmpeg.git 03Anton Khirnov 07release/0.7:4c849c69910c: dfa: check that the caller set width/height properly.
[01:10] <cone-485> ffmpeg.git 03Kostya Shishkov 07release/0.7:2d63f9b4effc: dfa: add some checks to ensure that decoder won't write past frame end
[01:10] <cone-485> ffmpeg.git 03Kostya Shishkov 07release/0.7:c0df6a24ce9e: indeo: check custom Huffman tables for errors
[01:10] <cone-485> ffmpeg.git 03Kostya Shishkov 07release/0.7:601fa565823b: indeo: clear allocated band buffers
[01:10] <cone-485> ffmpeg.git 03Kostya Shishkov 07release/0.7:3c0f84402b2b: indeo: check for invalid motion vectors
[01:10] <cone-485> ffmpeg.git 03Janne Grunau 07release/0.7:8148833193c6: indeo5: prevent null pointer dereference on broken files
[01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:aa097b4d5fd4: indeo5: check tile size in decode_mb_info().
[01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:44da556815fc: lagarith: check count before writing zeros.
[01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:4a636a5e4368: wmaprodec: check num_vec_coeffs for validity
[01:10] <cone-485> ffmpeg.git 03Anton Khirnov 07release/0.7:05f5a2eb62db: avidec: use actually read size instead of requested size
[01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:6996a2f79689: cavsdec: check for changing w/h.
[01:10] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:e3e369f6962c: alsdec: Check that quantized parcor coeffs are within range.
[01:10] <cone-485> ffmpeg.git 03Thilo Borgmann 07release/0.7:1b48a426a96f: alsdec: Fix out of ltp_gain_values read.
[01:10] <cone-485> ffmpeg.git 03Mans Rullgard 07release/0.7:7e070cf2025f: alsdec: remove dead assignments
[01:10] <cone-485> ffmpeg.git 03Thilo Borgmann 07release/0.7:9474c9302844: alsdec: fix number of decoded samples in first sub-block in BGMC mode.
[01:11] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:aa45b90804ab: alsdec: Check k used for rice decoder.
[01:11] <cone-485> ffmpeg.git 03Reinhard Tartler 07release/0.7:642d758a2d82: Update RELEASE file for 0.7.7
[01:11] <cone-485> ffmpeg.git 03Piotr Bandurski 07release/0.7:fe9cbf582b09: tiffdec: Use the correct height field.
[01:11] <cone-485> ffmpeg.git 03Anton Khirnov 07release/0.7:9f8071245491: ivi_common: check that scan pattern is set before using it.
[01:11] <cone-485> ffmpeg.git 03Luca Barbato 07release/0.7:700fb8c8dd06: vp56: make parse_header return standard error codes
[01:11] <cone-485> ffmpeg.git 03Luca Barbato 07release/0.7:7fd7950174f9: vp56: release frames on error
[01:11] <cone-485> ffmpeg.git 03Sami Pietila 07release/0.7:bfbff1c7483c: vp8: reset loopfilter delta values at keyframes.
[01:11] <cone-485> ffmpeg.git 03Luca Barbato 07release/0.7:f3f22f183fa2: yuv4mpeg: reject unsupported codecs
[01:11] <cone-485> ffmpeg.git 03Justin Ruggles 07release/0.7:3d0c9c9af687: flacenc: ensure the order is within the min/max range in LPC order search
[01:11] <cone-485> ffmpeg.git 03Reinhard Tartler 07release/0.7:ce8910d861c1: h264: Fix parameters to ff_er_add_slice() call
[01:11] <cone-485> ffmpeg.git 03Luca Barbato 07release/0.7:4ede95e69cf9: vp6: properly fail on unsupported feature
[01:11] <cone-485> ffmpeg.git 03Janne Grunau 07release/0.7:10ff052c6013: lavf: avoid integer overflow in ff_compute_frame_duration()
[01:11] <cone-485> ffmpeg.git 03Alex Converse 07release/0.7:b143844ea0f6: aacdec: Fix an off-by-one overwrite when switching to LTP profile from MAIN.
[01:11] <cone-485> ffmpeg.git 03Janne Grunau 07release/0.7:5fa739e685bc: h264: enable low delay only if no delayed frames were seen
[01:11] <cone-485> ffmpeg.git 03Luca Barbato 07release/0.7:08d9fd611eac: ppc: always use pic for shared libraries
[01:11] <cone-485> ffmpeg.git 03Janne Grunau 07release/0.7:4457e6137d83: h264: check sps.log2_max_frame_num for validity
[01:11] <cone-485> ffmpeg.git 03Victor Lopez 07release/0.7:884a9b0d298a: h264: fix sps parsing for SVC and CAVLC 4:4:4 Intra profiles
[01:11] <cone-485> ffmpeg.git 03Justin Ruggles 07release/0.7:a39c6bf1b878: alacdec: do not be too strict about the extradata size
[01:11] <cone-485> ffmpeg.git 03Martin Storsjö 07release/0.7:808187965570: rtsp: Recheck the reordering queue if getting a new packet
[01:11] <cone-485> ffmpeg.git 03Dale Curtis 07release/0.7:55065315caf1: Fix uninitialized reads on malformed ogg files.
[01:11] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:391e0fc6c90c: roqvideodec: check dimensions validity
[01:11] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:af343f5cddc5: eamad: fix out of array accesses
[01:11] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:2cac35086c9e: vqavideo: check chunk sizes before reading chunks
[01:11] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:e6ac11e41734: aacdec: check channel count
[01:11] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:41eda870483f: pngdec/filter: dont access out of array elements at the end
[01:11] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:377fabc9e687: Update for 0.8.13
[01:11] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:7378101d4102: Merge branch 'release/0.8' into release/0.7
[01:14] <cone-485> ffmpeg.git 03Michael Niedermayer 07release/0.7:f86da592c339: update for 0.7.14
[01:23] <llogan> saste: are you interested in administering an application to GSoC this year?
[01:24] <saste> llogan, if there are no other volunteers
[01:25] <saste> possibly, i'd prefer to just act as co-administrator
[01:26] <ubitux> @_@
[01:31] <llogan> i can help format/cleanup/prettify the ideas list at least...
[01:34] <llogan> should we continue to use multimedia.cx, or use the trac wiki this time? .cx is more of an actual wiki, but it might look nice to use our own domain
[01:35] <saste> llogan, you can already start to do it
[01:35] <saste> i'm biased towared multimedia.cx, because of stricter access policy
[01:35] <llogan> shit. responsibilities...
[01:35] <saste> we don't want to let random j. hacker to edit the page just before review
[01:36] <llogan> multimedia.cx is fine with me as long as whoever needs to edit it can get access. does it still depend on mike?
[01:36] <saste> llogan, i think so
[01:36] <saste> we should probably ask him before
[01:37] <llogan> good idea. i can do that.
[01:37] <saste> then we could simply copy the 2012 page to a new one, and let candidate mentors edit the proposed tasks
[01:37] <llogan> ok
[01:38] <llogan> i'll try to do that later today or tomorrow if i hear back from mike
[01:39] <saste> llogan, thanks
[01:46] <cone-485> ffmpeg.git 03Ronald S. Bultje 07master:c63f9fb37a7b: h264: don't store intra pcm samples in h->mb.
[02:08] <ubitux> what is cathedral?
[02:09] <BBB> a sample clip often used for profiling in this community
[02:10] <ubitux> where can i find it?
[02:10] <Compn> oh
[02:10] <Compn> that clip
[02:10] <Compn> raw yuv
[02:10] <BBB> google for cathedral-beta2-400extra-crop-avc.mp4
[02:10] <ubitux> thx
[02:11] <BBB> it's somewhere on mplayersamples or whatever that site is called
[02:11] <BBB> but I'm too lazy to find it
[02:12] <ubitux> http://samples.ffmpeg.org/V-codecs/h264/cathedral-beta2-400extra-crop-avc.m…
[02:12] <ubitux> thx :)
[02:12] <BBB> tadaah
[02:13] <Compn> what?
[02:13] <Compn> dont use the compressed version :P
[02:14] <ubitux> oh, never saw this diablo-elephant-dream animation
[02:14] Action: ubitux looking for a higher res now
[02:16] <Compn> http://samples.ffmpeg.org/raw-video/
[02:16] <Compn> well foremans in there
[02:18] <ubitux> looks like there are some other interesting stuff on platige.com
[02:55] <ubitux> oh, seems libav realized they broke another codec with cosmetics
[02:56] <ubitux> mmh, already fixed in ffmpeg.
[02:59] <ubitux> actually, michaelni merged it properly
[03:02] <Compn> we're not supposed to talk/troll/complain about libav, apparently
[03:02] <Compn> :P
[03:02] <ubitux> who told this?
[03:02] <Compn> that guy daemon404 who complains about ffmpeg development
[03:02] <ubitux> yes but he ragequited so that's ok
[03:03] <Compn> lol
[03:03] <ubitux> anyway, it's just that i find very funny when Diego is saying Michael is merging blindly, while he is actually fixing the crap he introduce with his cosmetics madness
[03:05] <Compn> i'm curious how they arent breaking fate ?
[03:09] <BBB> you know, svq3 is a very elaborate format
[03:09] <BBB> it looks simple, but isn't
[03:10] <BBB> it uses hpel, tpel, for example
[03:10] <BBB> and has many coding modes
[03:10] <BBB> code coverage of the sample in fate is likely in the 50% range
[03:10] <BBB> in terms of functional coverage, not lines of code
[03:11] <ubitux> in term of LoC it's about 70%
[03:11] <ubitux> http://lucy.pkh.me/ffmpeg-coverage/ffmpeg/libavcodec/svq3.c.gcov.html
[03:12] <ubitux> and it seems the broken code was exactly in the non-covered area
[03:13] <ubitux> (~L564)
[03:13] <BBB> there you go
[03:13] <BBB> so one could say that we need more fate coverage
[03:13] <BBB> (not that I care)
[03:15] <Compn> svq3 is tied into h264 heavily
[03:15] <Compn> or was
[03:16] <Compn> BBB : i like your .2percent speedups :)
[03:20] <BBB> it still is
[03:20] <BBB> this is more cleanup than speedup, strictly speaking
[03:20] <BBB> but thanks anyway
[03:56] <BBB> isn't draw_horiz_band utterly broken for h264?
[03:56] <BBB> (unless SLICE_FLAG_CODED_ORDER is set)
[03:56] <BBB> michaelni: ^^
[04:04] <michaelni> BBB, it does look broken, i wonder why noone reported an issue
[04:38] <BBB> michaelni: I can only imagine nobody is using it
[04:38] <BBB> can we unshare that between mpeg and h264?
[04:38] <BBB> it makes no sense to share after my last patch, there's virtually no code shared at all
[05:50] <highgod> Hi,all,I want to ask a question. I use the hwaccel and software both decode the same h264 video to yuv. I want to ask that are the both yuv file are the same? totally?
[05:56] <Compn> you want to compare each file ?
[05:57] <Compn> use ffmpeg with the framemd5 , i think
[05:57] <Compn> if they are exact, the md5 should match
[05:57] <Compn> ffmpeg -i file.yuv -f framemd5
[06:02] <Compn> ffmpeg -i file.yuv -f framemd5 out.txt that is
[06:02] <Compn> highgod : it work for you ?
[06:02] <Compn> michaelni : are you going to review the dxva2 patch? i dont think anyone else is
[06:02] <highgod> No,I use dxva2 and ffmpeg software decode the same h.264 video to yuv,but the two files is mismatch in some bits
[06:03] <Compn> highgod : if you are using ffmpeg to decode using dxva2, cant you directly use -f framemd5 ? just to make sure the yuv part isnt adding additional bug
[06:04] <Compn> are some frames the same ? is it one type of frame that causes problems? like b-frame or p-frame or i-frame ?
[06:04] <Compn> and are you sure you are using single thread h264 software decoder ? i think multithreaded h264 decoder will cause slight differences when decoding
[06:05] <BBB> no it won't
[06:05] <Compn> oh ok, was that fixed ?
[06:05] <BBB> multithreaded h264 is entirely bitexact for all test sequences
[06:05] <Compn> good :)
[06:05] <BBB> I fixed that before it was ever merged in either tree
[06:05] <Compn> its hard to remember all of these bugs
[06:06] <BBB> you can test it locally
[06:06] <BBB> make THREADS=2 fate-h264
[06:06] <Compn> yeah well i could do that
[06:06] <BBB> if that passes, all sequences are bitexact
[06:06] <Compn> or i could go to sleep
[06:06] Action: Compn sleeps
[06:06] <BBB> fate.ffmpeg.org does it every day
[06:06] <BBB> http://fate.ffmpeg.org/report.cgi?time=20130218001421&slot=x86_64-archlinux…
[06:12] <highgod> I just use the merge tool to compare the two yuv
[06:31] <BBB> you'd want to use jm to decode the h264 file and check which is correct
[06:32] <highgod> sorry,I tested use ffmpeg -s 1920x1080 -i ./testfile/output.yuv -f framemd5 out.txt, and ffmpeg -s 1920x1080 -i ./testfile/output_dxva2.yuv -f framemd5 out_dxva2.txt.all the same
[06:32] <highgod> 10frames
[06:32] <highgod> I will test more frames
[06:38] <highgod> I test all the 912 for my video,all the md5 code is the same
[06:39] <BBB> then they are identical
[06:44] <highgod> @BBB: the orignal code I implement modified the pthread code, is it can cause the difference? the difference happened on my orignal code. the patch version I tested just now is OK
[06:44] <BBB> I don't know, I didn't see the patch
[06:44] <BBB> messing with pthread is a bad idea in general, unless you know what you're doing
[06:44] <BBB> (I know what I'm doing, so I know how much a bad idea it is)
[06:46] <highgod> hehe,yes.Thanks
[11:11] <ubitux> durandal_1707: about the ffmpeg -help full, i was wondering if it didn't go in infinite loop or something, because of the cross references
[11:11] <ubitux> like it does for instance when you have several filters with the same class
[11:16] <durandal_1707> ubitux: doesn't happen
[11:17] <ubitux> okay, good
[11:17] <durandal_1707> i dont see cross reference
[11:17] <ubitux> seems valgrind is not happy since a little while
[11:17] <durandal_1707> ?
[11:17] <ubitux> durandal_1707: i was refering to multiple options refering to the same flags
[11:17] <ubitux> (my valgrind remark is unrelated)
[11:17] <ubitux> http://fate.ffmpeg.org/report.cgi?time=20130218023828&slot=x86_64-archlinux…
[11:17] <ubitux> http://fate.ffmpeg.org/history.cgi?slot=x86_64-archlinux-gcc-valgrindundef
[11:18] <ubitux> i guess d2a25c4032ce6ceabb0f51b5c1e6ca865395a793 is the cause
[11:19] <ubitux> i wonder if libav finally added a real valgrind instance to spot these
[11:20] <ubitux> ah yeah, there is one, in the yellow as well.
[11:20] <michaelni> Compn, did you ask fenrir ? he is the author of libavcodec/dxva2* ...
[11:31] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:39bec05ed42e: qdm2: check array index before use, fix out of array accesses
[11:31] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:f7d18deb73d1: vqavideo: check chunk sizes before reading chunks
[11:31] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:488f87be8735: roqvideodec: check dimensions validity
[11:31] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:96008082dbd3: Merge commit '488f87be873506abb01d67708a67c10a4dd29283'
[11:36] <durandal_1707> ubitux: i gonna write lua filter if none will, using eval is slow and unfriendly
[11:39] <durandal_1707> ubitux: guess what this does: ffmpeg -f lavfi -i color=c=white:size=720x480 -i INPUT -filter_complex "blend=all_expr=B*(if(gte(T\,10)\,1\,mod(T\,10)/10))+A*(1-(if(gte(T\,10)\,1\,mod(T\,10)/10)))"
[11:59] <ubitux> fade? :)
[11:59] Action: ubitux trying
[12:05] <ubitux> yeah e
[12:06] <cone-967> ffmpeg.git 03Luca Barbato 07master:0b70fb1d518c: dsputil: convert remaining op_pixels_func
[12:06] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:f88f7b86a962: Merge commit '0b70fb1d518cbd796545fd6eef02772cd0d892c7'
[12:14] <cone-967> ffmpeg.git 03Diego Biurrun 07master:56632fef65c0: libopencore-amrnb: cosmetics: Group all encoder-related code together
[12:14] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:4ff0e63a015a: Merge commit '56632fef65c0cb6946ed3648ded3d7b82e5c5c17'
[12:19] <cone-967> ffmpeg.git 03Diego Biurrun 07master:e6bda9a9fd86: libopencore-amr: Conditionally compile decoder and encoder bits
[12:19] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:ab213b5360e8: Merge commit 'e6bda9a9fd86505927a2e095e495eae104860701'
[12:26] <cone-967> ffmpeg.git 03Diego Biurrun 07master:8837f4396a1a: libopencore-amrwb: Make AMR-WB ifdeffery more precise
[12:26] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:2220f13d8227: Merge commit '8837f4396a1a458a0efb07fe7daba7b847755a7a'
[12:34] <cone-967> ffmpeg.git 03Diego Biurrun 07master:870a0c669e53: build: The libopencore-amrnb encoder depends on audio_frame_queue
[12:34] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:4b7c81c3e059: Merge commit '870a0c669e536d56c6325d84f65e34c53792398e'
[13:00] <cone-967> ffmpeg.git 03Luca Barbato 07master:aa11cb79318b: build: make audio_frame_queue a stand-alone component
[13:00] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:82e82fa2e5bb: Merge commit 'aa11cb79318baa3415d553424ba378f6c62e1f9b'
[13:11] <cone-967> ffmpeg.git 03Matti Hamalainen 07master:7f3624dc81a1: svq3: unbreak decoding
[13:11] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:0789374ed39a: Merge remote-tracking branch 'qatar/master'
[13:19] <divVerent> haha, I have the very oppposite bug... so maybe this regression fixes my issue :P
[13:19] <divVerent> I just couldn't figure out why -vn doesn't prevent album art from being copied, and also could not find a -map_metadata option to do this
[13:22] <divVerent> BTW, I have four features I cannot "really" enable in ffmpeg... is this normal?
[13:22] <divVerent> http://paste.debian.net/235197/
[13:22] <divVerent> frei0r and libopencv only really can be used with ffmpeg if they are built without ffmpeg support, or we get a cycle :P
[13:22] <divVerent> (yes, I know, cycles can actually be worked around, but not when packaging software)
[13:23] <divVerent> libflite and libilbc apparently don't work, any idea about these? as in, which version of these is ffmpeg known to work with?
[13:24] <divVerent> I noticed these when making an arch linux PKGBUILD that tries to enable virtually everything :P
[13:25] <divVerent> the cycle can be resolved when dynamically linking by first building ffmpeg without the dependency, building the dependency with ffmpeg support, then building ffmpeg with it
[13:25] <divVerent> just can't apply this to packaging
[13:25] <divVerent> just find it funny that ffmpeg supports using libraries that in turn depend on ffmpeg in the usual configurations :P
[13:27] <ubitux> libflite is very recent afaik
[13:27] <ubitux> ask saste when he's back
[13:28] <ubitux> divVerent: btw, don't want your name in the showspectrum copyright?
[13:28] <ubitux> not that is has much legal value in comparison to the git history, but well
[13:30] <divVerent> ubitux: don't care much for that, my changes aren't that big
[13:30] <divVerent> the infrastructure in there is not by me, I only did the drawing part
[13:31] <divVerent> so for my taste, the git history's info alone is ok
[13:31] <divVerent> if you want to add me to the file, go ahead, though
[13:35] <ubitux> ok :)
[13:36] <cone-967> ffmpeg.git 03Clément BSsch 07master:f894246304b3: lavf/showspectrum: add divVerent in the (c) for his recent work on the filter.
[13:36] <divVerent> yes, in retrospect, I should have added that... it's better for ffmpeg to have clean copyright notices, even on minor changes... one too many is better than one too little
[14:09] <durandal_1707> ubitux: so you have no opinion on lua filter?
[14:11] <ubitux> i would be happy to have a lua filter, but actually i think what is lacking is not a lua filter, but a lua filtergraph architecture
[14:11] <ubitux> typically if you want to script an editing scenario
[14:12] <ubitux> basically, a "lua binding" for the lavfi api
[14:12] <durandal_1707> and how it should be implemented?
[14:12] <ubitux> i have no idea
[14:15] <durandal_1707> ubitux: not as filter that does filtergraph?
[14:18] <ubitux> might be interesting
[14:18] <durandal_1707> or ffmpeg script
[14:19] <durandal_1707> ?
[14:20] <ubitux> i'm not sure
[14:20] <ubitux> you should look for common workflow in the anime ripping world
[14:20] <ubitux> we talked about a while ago about that stuff
[14:21] <ubitux> it was things like detect the interlacing (so calling filters and analyze results), then do some processing, maybe do some ad cutting in the middle, or change the de-interlacing method for the op/ed part
[14:22] <ubitux> whatever the solution, imo, this kind of operation should be solvable easily
[14:22] <ubitux> you can imagine processing some filters on part of the video too
[14:23] <ubitux> anyway, that's what the lua scripting should be used for
[14:23] <ubitux> that might be different from what you originally had planned though (aka a geq-like in lua)
[14:23] <durandal_1707> what? no
[14:23] <ubitux> ah i misunderstood then, sorry
[14:24] <durandal_1707> lua filter would just create filtergraph using own stuff
[14:24] <durandal_1707> because using eval for if/else is slow and ugly
[14:25] <durandal_1707> and idea is that "lua filter" would take input as file script
[14:25] <durandal_1707> (i do not use ed for programming ....)
[14:26] <ubitux> a script.lua?
[14:26] <ubitux> that reminds me there was some security concerns about this
[14:27] <ubitux> (about accessing any FFmpeg C functions from a user PoV)
[14:27] <durandal_1707> isn't that implementation problem
[14:27] <ubitux> maybe
[14:28] <durandal_1707> doesn't avisynth/vapoursynth do just that?
[15:14] <ubitux> no idea, i never used that
[15:26] <cone-967> ffmpeg.git 03Ronald S. Bultje 07master:51513b98d62d: h264/svq3: stop using draw_edges.
[15:26] <cone-967> ffmpeg.git 03Ronald S. Bultje 07master:71ae8d50b283: x86/dsputil: fix compilation when h263 decoder/encoder are disabled.
[15:31] <durandal_1707> so in the end ffmpeg logo on web page is strange
[15:48] <Compn> durandal_1707 : you should see mplayer's scriptable 'run' command if you want security trouble ... :)
[15:50] <durandal_1707> that just execute shell?
[15:59] <Compn> yea run sudo rm -rf /
[16:02] <Compn> michaelni : thanks for looking out for mplayer compilation :
[16:02] <Compn> :)
[16:14] <durandal_1707> michaelni: ffmpeg -ss 3 -i /tmp/1.nut gives assertion if nut is ffv1 video with single frame
[17:26] <cone-967> ffmpeg.git 03Paul B Mahol 07master:1f4ab61b746b: lavc/cdxl: clear palette before reading it
[17:31] <BBB> does anyone have qemu-system-arm set up on his system and want to test a series of patches for me?
[17:37] <av500> or a real ARM system
[17:37] <av500> they exist....
[17:37] <BBB> that would be nice also
[17:37] <BBB> anyone?
[17:42] <michaelni> I still have that old beagle board that neither thilo nor me could get fully working with fate, its not fun to use but should do for a single fate test (compared to all fate tests which it just doesnt seem to pass without failing somehow)
[17:43] <michaelni> i dont recommand that though if theres any other solution
[17:44] <cone-967> ffmpeg.git 03Paul B Mahol 07master:85921499c747: cdgraphics: set palette to zero too
[17:44] <michaelni> but if you or another ffmpeg devel wants the board iam happy to put it in a box and send it ...
[17:47] <BBB> I have enough arm devices if I wanted to run fate myself
[17:47] <BBB> I dont' :)
[17:47] <BBB> they're all a pain
[17:49] <BBB> e.g. current version (http://pastebin.com/x4Rcd4RD) may fix arm, but I'm not sure
[17:49] <BBB> it would take me many hours to hook up any arm device and run a test; so if anyone has arm set up, that would be a lot quicker
[18:07] <cone-967> ffmpeg.git 03Paul B Mahol 07fatal: ambiguous argument 'refs/tags/n0.7.14': unknown revision or path not in the working tree.
[18:07] <cone-967> Use '--' to separate paths from revisions
[18:07] <cone-967> refs/tags/n0.7.14:HEAD: cdgraphics: set palette to zero too
[18:09] <durandal_1707> ^ can something be done with this failure?
[18:10] <michaelni> durandal_1707, talk with thresh
[18:11] <BBB> yeah and qemu setup doesn't work either on mac
[18:11] <BBB> bah
[18:12] <BBB> slightly better version: http://pastebin.com/EDT4BWV8
[18:12] <BBB> still untested, but maybe it works
[18:12] <av500> one will never know...
[18:12] <BBB> I s'pose
[18:13] <BBB> I believe one of my computers has an android sdk installed somewhere, and I was able to connect to my phone at least at some point for some testing
[18:15] <durandal_1707> michaelni: anyway, I did not touched 0.7.14 and i dot plan to, also i dont see it in web interface
[18:18] <michaelni> durandal_1707, yes, cone has a bug with tags it posts nosnense, thresh and j-b are cones masters, ask them what is wrong with code / if it can be fixed
[18:19] <michaelni> s/code/cone/
[18:20] <michaelni> the repo is always fine / unaffected its just what is printed to IRC thats a bit off
[19:04] <durandal_1707> are there files/samples where stereo3d filter is useful
[19:04] <durandal_1707> i plan to port this filter
[19:08] <durandal_1707> michaelni: gonna fix that assert in nut or should i fill bug?
[20:14] <durandal_1707> -filter_complex "showwaves=mode=line:size=976x240[out3],split[one][two],[one]histogram=mode=waveform:waveform_mode=row:display_mode=overlay[out1],[two]pad=976:720[out2],[out2][out1]overlay=720:0[x],[x][out3]overlay=0:480"
[20:19] <ubitux> does it work?
[20:19] <ubitux> (doesn't seem so here for unknown reason)
[20:20] <durandal_1707> what error you get?
[20:20] <ubitux> "Open inputs in the filtergraph are not acceptable"
[20:21] <ubitux> but i'm not using filter_complex here
[20:21] <durandal_1707> you should
[20:21] <durandal_1707> thought i dunno what that error means
[20:21] <durandal_1707> i display it with ffmpeg sld muxer
[20:21] <durandal_1707> *sdl
[20:22] <ubitux> oh good idea the ffmpeg sdl
[20:22] <durandal_1707> ffplay is in bad shape
[20:22] <ubitux> will try, after i figure out the problem
[20:27] <ubitux> it's strange..
[20:27] <durandal_1707> blend=all_expr=if(eq(mod(X\,2)\,mod(Y\,2))\,A\,B)
[20:33] <ubitux> ok get it.
[20:45] <ubitux> Open inputs in the filtergraph are not acceptable. Orphan inputs: "in" "(null)"
[20:45] <ubitux> erm.
[20:46] <ubitux> actually, just "(null)"
[20:50] <ubitux> durandal_1707: ./ffplay -f lavfi "amovie=niea7-op.mp3,showwaves=mode=line:size=976x240,split[waves1][waves2]; [waves1]split[a][b]; [a]histogram=mode=waveform:waveform_mode=row:display_mode=overlay[hist]; [b]pad=976:720[padded]; [padded][hist]overlay=720:0[over]; [over][waves2]overlay=0:480"
[20:50] <ubitux> is that the same?
[20:50] <ubitux> maybe a split=3 would make more sense
[20:52] <ubitux> so ./ffplay -f lavfi "amovie=niea7-op.mp3,showwaves=mode=line:size=976x240,split=3[a][b][c]; [a]histogram=mode=waveform:waveform_mode=row:display_mode=overlay[hist]; [b]pad=976:720[padded]; [padded][hist]overlay=720:0[over]; [over][c]overlay=0:480"
[21:01] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:695a766bff4c: ff_read_timestamp: check stream_index before using it as array index
[21:01] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:6cd650dbd2be: ff_gen_search: Fix finding the maximum timestamp in a really small file
[21:01] <cone-967> ffmpeg.git 03Michael Niedermayer 07master:e39821a65e42: nutenc: only write an index if there are syncpoints
[21:04] <durandal_1707> michaelni: what? seek does not return -1 any more for lavf-ogg?
[21:05] <durandal_1707> :)
[21:11] <michaelni> yes it matches asf/mpg in that now
[21:11] <michaelni> or nut
[21:18] <cone-967> ffmpeg.git 03Paul B Mahol 07master:480ddf2bc9c0: lavfi/histogram: overlay display mode for levels histogram mode
[21:50] <Compn> durandal_1707 : youtube outputs some 3d stuff ,like side-by-side samples
[21:50] <Compn> stereo3d can convert between side by side, top/bottom, and red/blue (amongst a bunch of other colors for analgylph 3d)
[21:50] <Compn> also frame interleaved or something
[21:50] <Compn> probably
[21:53] <durandal_1707> and there is sample, with which filter is used?
[21:55] <durandal_1707> i can use any video, but would like to have one where output will make more sense
[21:57] Action: durandal_1707 should probably google it
[22:05] <Compn> durandal_1707 : oh, well you take a side by side and use stereo3d to convert it to red/blue i guess
[22:05] <Compn> i think i wrote examples on the wiki
[22:05] <Compn> one sec
[22:06] <Compn> oh no
[22:06] <Compn> i wrote this one
[22:06] <Compn> http://wiki.multimedia.cx/index.php?title=How_to_make_a_3d_movie_with_ffmpeg
[22:06] <Compn> this page has mencoder options > http://www.mtbs3d.com/phpbb/viewtopic.php?f=27&t=4521
[22:07] <Compn> er page 3 of that does > http://www.mtbs3d.com/phpbb/viewtopic.php?f=132&t=4521&start=30
[22:31] <cone-967> ffmpeg.git 03Ronald S. Bultje 07master:e2789d3e3399: split out ff_hwaccel_pixfmt_list_420[] over individual codecs.
[23:05] <wm4> what is the state of a demuxer after a failed av_seek_frame()?
[23:05] <wm4> in particular, what does av_read_frame() return?
[23:10] <wm4> I think it_'s reproducible with ffplay and a .flac file
[23:11] <wm4> hit the cursor down and cursor left keys at the same time a few times at the beginning of the file
[23:11] <wm4> and you get errors like [flac @ 0x9106220] invalid sync code
[23:11] <wm4> and it only happens after "error while seeking" is printed
[23:12] <saste> llogan, any news from mike?
[23:14] <llogan> saste: no reply yet.
[23:14] <llogan> although i would like to get started on it soon
[23:15] <durandal_1707> on what?
[23:17] <wm4> hm I guess a ticket will generate more attention
[23:17] <llogan> i asked him about using his wiki again for the GSoC idea list and if he didn't mind keeping up with any new requests for edit permissions
[23:18] <michaelni> wm4, the state after failed seek is undefined (that is how it is currently, not neccesarily what we want it to be)
[23:18] <durandal_1707> llogan: you need permisison to edit gsoc page?
[23:18] <wm4> michaelni: well, it appears for some formats it works, and for some (flac) it doesn't
[23:19] <wm4> michaelni: and by "works" I mean unconditionally resetting the decoder, and then calling av_read_frame() and feeding that to the decoder
[23:19] <durandal_1707> flac use generic seek
[23:19] <llogan> durandal_1707: you need an account to edit http://wiki.multimedia.cx ...or maybe an account with editing permissions, either way mike has to manually do something
[23:19] <michaelni> wm4, reset is easy the hard part is to be in the same state as before the failed seek was attempted
[23:20] <wm4> michaelni: so I would suggest: seeking before start of the file -> clamp seek position to 0 (and always succeed seeking), and seeking past the end of the file => return EOF (and let av_read_frame() return EOF too)
[23:20] <michaelni> llogan, you can just use trac wiki
[23:21] <llogan> i know, and i like the trac wiki for the most part, but saste mentioned that a more strict permission might be good in this case
[23:21] <llogan> last minute troll vandalism, etc
[23:22] <durandal_1707> trac wiki can't have finer permissions?
[23:22] <michaelni> can be done somehow AFAIK but i dunno how
[23:23] <llogan> probably, but michael has better things to deal with
[23:23] <llogan> if i don't hear from him by tomorrow i'll just use our wiki.
[23:25] <durandal_1707> i could do it...
[23:32] <saste> http://trac.edgewall.org/wiki/TracFineGrainedPermissions
[23:32] <saste> so yes it is possible
[23:32] <wm4> how can I seek with "ffmpeg" again?
[23:32] <durandal_1707> wm4: only once -ss
[23:33] <saste> i have no objection into using our wiki if we can implement dev-only access on the gsoc page
[23:33] <wm4> ok... that's not enough
[23:34] <durandal_1707> saste: it appears doable, but from friendly
[23:34] <durandal_1707> *far
[23:36] <saste> durandal_1707, what do you mean, unfriendly towards whom?
[23:36] <durandal_1707> the trac
[23:37] <saste> gsoc page should be edited only by candidate mentors or developers
[23:37] <saste> there is no much point into allowing a random user to edit it
[23:38] <saste> we never did like that in the past years
[23:41] <durandal_1707> i mean cofiguring finer permissions for trac are unfriendly.
[23:42] <durandal_1707> llogan: why you plan to put for gsoc?
[23:43] <durandal_1707> *what
[23:43] Action: durandal_1707 needs to sleep
[23:44] <llogan> i was going to copy most of last year's and let interested devs/mentors edit it. then i was going to make sure it is formatted well and in a good layout before google views it
[23:44] <llogan> scrolling marquees, page counter, some flash, gifs, etc.
[23:45] <durandal_1707> flash?
[23:45] <llogan> bad joke
[23:45] <llogan> durandal_1707 needs to sleep more
[00:00] --- Tue Feb 19 2013
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[02:36] <llogan> relaxed: were you able to work on your compile guide?
[03:09] <t4nk422> hi all, I'm trying to use ffmpeg to convert an FLV file to MP4. I managed to do this successfully for some FLV files, but other seem to be failing. The encoding on the ones that fail are saying "flv1, yuv420p". I'm running on Ubuntu and followed the following link to compile ffmpeg: https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide. Am I missing a flag or a codec that I need to add, any assistance or pointers would be very
[03:09] <t4nk422> My command line output is: http://pastebin.com/pJVyfirD
[03:16] <llogan> t4nk422: never seen that before. can you supply a sample file?
[03:17] <llogan> also please include the command and output for one that does work
[03:19] <t4nk422> Yeah sure, let me put that together
[03:26] <t4nk422> It looks like the one that was working is encoded differently (see pastebin). The working file plays in VLC & Movie Player on Ubuntu. The notworking file ONLY plays in VLC, would it be a codec I'm missing on my system that might be causing this? http://pastebin.com/Gv0eLS7N
[03:27] <t4nk422> The links listed in the pastebin, should make those random sample files playable
[03:30] <Tvilling> speex and flv1 vs. vp6 and mp3
[03:32] <llogan> t4nk422: looks like the audio stream is the source of the issue, AFAIK. probably worthy of a bug report.
[03:33] <llogan> ah, i see. i think you need --enable-libspeex.
[03:34] <t4nk422> ah thanks, I'll give that a try
[03:34] <llogan> i assumed ffmpeg can decode speex natively, but i was wrong and that error isn't very informative
[03:34] <llogan> https://ffmpeg.org/trac/ffmpeg/ticket/1668
[03:35] <llogan> so install libspeex-dev and recompile ffmpeg with --enable-libspeex
[03:35] <Tvilling> Try running it more verbosely next time.
[03:36] <llogan> the verbose output was not very useful in this case.
[03:36] <t4nk422> ah ok, thanks very much for the help to both of you, installing that now
[03:38] <llogan> t4nk422: refer to the "Upgrading FFmpeg" section of the guide
[03:38] <Tvilling> llogan: The debug level weren't helpful as well?
[03:39] <llogan> not for the general user.
[03:39] <Tvilling> Eh.
[03:43] <llogan> t4nk422: i updated the guide so it will now install libspeex.
[08:04] <defaultro> llogan, the slowmovideo result was awesome :)
[14:30] <jarno> ...using libav* and libx264 to encode and mux a video stream received from a camera and I get "nvalid DTS: PTS is less than DTS"...I'm setting PTS in my code and it indeed is monotonically increasing....
[14:30] <jarno> ...so what's the cure?
[14:32] <jarno> ...anyone?
[14:44] <Mavrik> jarno: obviously you need to set DTS on output packets as well.
[14:47] <jarno> ...I took a quick peak related to this issue and I (like others) thought that the DTS was something generated by the libx264
[14:47] <jarno> but, then again, I must have been wrong
[14:47] <Mavrik> it is
[14:47] <Mavrik> PTS and DTS is generated by libx264
[14:48] <Mavrik> so if you're getting that error
[14:48] <Mavrik> you're doing something wrong with them after encode
[14:48] <jarno> in my code I'm setting PTS for the frame but not the DTS....
[14:48] <jarno> ...since I thought it was not necessary to set the DTS....
[14:49] <Mavrik> I see,
[14:49] <Mavrik> well PTS and DTS of output packets are set by libx264 encoder
[14:50] <Mavrik> you need to make sure your frames going to the encoder have proper PTS
[14:50] <Mavrik> are you by any chance trying to set PTS on encoded frames?
[14:50] <jarno> I'm setting PTS on the frame that I give to the encoder
[14:51] <jarno> ...i.e. the frame that I want to be encoded...
[14:52] <Mavrik> well, step through the code with a debugger and find out where do your PTS change
[14:53] <jarno> ...okis...will do...however, I'm pretty much following the decoding_encoding example....
[14:54] <jarno> ...in any case, thanx for the hint...
[14:54] <JEEB> also, the reason for the error is simple -- PTS never can be smaller than DTS
[14:54] <JEEB> DTS = when is this frame decoded
[14:54] <JEEB> PTS = when is this frame shown
[14:54] <JEEB> you cannot show it before it's decoded :P
[14:55] <JEEB> but yes, if you give libx264 valid PTS values, it should give you valid PTS and DTS
[14:55] <Mavrik> yeah
[14:55] <Mavrik> jarno: basically, libx264 should never give you invalid PTS/DTS pairs, so you're breaking them somewhere in your code before passing them to the muxer
[14:56] <Mavrik> unless you have broken/buggy versions of libav*/x264
[14:56] <Mavrik> but that's extremely unlikely
[14:58] <jarno> ...ok...I'm pretty sure that I'm the cause of the problem since I'm new to this...
[14:58] <Mavrik> so I suggest you first just output PTS/DTS pairs right after encoding to see if the order there is ok (as JEEB said, DTS is ALWAYS <= PTS) and then follow your code through for changes
[14:59] <JEEB> remember to output both nominator and the denominator (timescale and the pts value) for all cases
[14:59] <jarno> ...ok...thanx guys!!!!!!!
[14:59] <JEEB> printf debugging om nom nom
[15:00] <Mavrik> ^^
[15:00] <jarno> ...okis...I'll output the whole stuff.... :) with a bit of luck I can get the initial version of this thing running today....!!!!!
[15:06] <brx_> im using an ffmpeg compiled for arm
[15:06] <brx_> on one device: ffmpeg -i a.jpg returns ifo on the file
[15:07] <brx_> on another device the same command returns "No such file or directory"
[15:07] <brx_> it does the same for png files, why is this? its the same ffmpeg binary on both devices
[15:07] <JEEB> sounds like a file rights issue
[15:07] <brx_> ahh i see
[15:10] <brx_> JEEB, i did chmod 777 on a.png, same error
[15:10] <brx_> im running the process as root btw
[15:10] <JEEB> is the ffmpeg binary run at all btw?
[15:11] <brx_> yes, it runs fine ffmpeg outputs the "file not found" error
[15:11] <JEEB> ok
[15:11] <brx_> also, when i try to overay a png using -VF it compains about the file
[15:11] <brx_> the same command works on another device
[15:11] <JEEB> I do remember someone on ARM/android having fun with rights and group ownership and other stuff
[15:12] <brx_> this is a major setback for me, the app im writing depends on overlaying an image onto a video
[15:13] <JEEB> see the code for what exactly is being checked for and how far the whole thing gets?
[15:13] <JEEB> and why it errors out
[15:13] <JEEB> that's all I can think of
[15:14] <brx_> ok thanks for the suggestions JEEB
[15:14] <Mavrik> usually strace helps with those problems
[15:16] <brx_> Mavrik, ok i will copy the console output now
[15:28] <brx_> Mavrik, JEEB http://pastebin.com/SA8ea0rF
[15:28] <brx_> both those commands work on nexus 7 but not my htc desire. both devices are rooted and im in the terminal as root
[15:30] <JEEB> first try more verbose output from ffmpeg, second of all, start tracing to th error through the source code :P
[15:30] <JEEB> also do note that there have been at least two major releases of ffmpeg since that version
[15:30] <JEEB> or three
[15:30] <brx_> ! ahh ok
[15:30] <JEEB> no, four
[15:30] <brx_> shit
[15:31] <JEEB> 0.10, 0.11, 1.0, 1.1
[15:31] <Mavrik> also, make sure your sdcard is actually mounted.
[15:31] <Mavrik> and not plugged into the computer
[15:31] <Mavrik> (which would cause difference in behaviour from Nexus 7 to Desire)
[15:31] <brx_> it is mounted, i can do cat /sdcard/image.png and it outputs the file
[15:31] <brx_> cable in or out, same errors
[15:32] <JEEB> I'm pretty sure that this is similar to that one other person's problems with rights on android devices before, but that possibly was already sorted over at the newer releases
[15:32] <JEEB> also without this person actually going there and checking where and why that error occurs we can't say anything
[15:32] <brx_> it was a bitch getting ffmpeg binary in the first place, myb thats why i went with an older version :s
[15:33] <brx_> thanks for the help ayway guys, im going to try and build the latest version of ffmpeg
[15:39] <Mavrik> well as I said
[15:39] <Mavrik> check strace and see what goes wrong
[15:46] <brx_> ok im on it now thanks
[15:58] <shevy> I have a .flv file here, which has Audio codec aac
[15:59] <shevy> I would like to snatch the audio stream and dump it somehow to a file
[15:59] <shevy> it however tells me: Encoder (codec id 86017) not found for output stream #0.0
[15:59] <shevy> is there perhaps a simplified way how to just split up audio and video data into separate streams?
[15:59] <shevy> I don't even know what aac exactly is, whether there could be a .aac file or not
[16:15] <shevy> relaxed hmm ok
[16:16] <shevy> I first tried to convert a .flv into a .mp3 actually
[16:16] <shevy> Here is the raw pastie: http://pastie.org/pastes/6213332/text
[16:19] <relaxed> Your ffmpeg version is very old. Try, ffmpeg -i test.flv -vn -acodec copy output.m4a
[16:22] <shevy> aha interesting, that created a file called output.m4a
[16:23] <shevy> now I seem to have an audio stream, thanks relaxed
[17:23] <terr1> Are there any x264 best practices for calculating which profile (baseline/main/high) and level (1-5.1) to use from a given input video source file?
[17:26] <klaxa> mhh i think profile depends more on what level of compression you want to have in your output file
[17:27] <klaxa> i don't know about the level parameter that much myself, i think this should give a general idea though? http://en.wikipedia.org/wiki/H.264/MPEG-4_AVC#Levels
[17:28] <klaxa> preset and crf are important factors for the resulting output too, in general have a look at this guide http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide
[17:29] <terr1> klaxa, ok thanks I will read thoes
[21:32] <Sander^home> DOes it exist a standar for recording video in a browser yet?
[21:32] <Fjorgynn> recording video in a browser?
[21:33] <Sander^home> Fjorgynn, Yes.. To do it without flash.
[21:34] <Fjorgynn> no
[21:34] <Fjorgynn> different for each browser
[21:34] <Fjorgynn> webm is one of them
[21:36] <Sander^home> Fjorgynn, Do you have any link on how to do it in html5?
[21:38] <Sander^home> Fjorgynn, in webkit eg?
[22:41] <teratorn> anyone can recommend an image viewer application that can handle raw formats?
[22:43] <ubitux> ffplay
[22:43] <teratorn> ubitux: I had a sneaking suspicion...
[22:44] <durandal_1707> raw formats?
[22:45] <durandal_1707> ffplay is bad because of sdl only output ...., and image viewer should try hard to display correct color
[22:45] <teratorn> durandal_1707: YUV420P, specifically... assuming adjacent plane alignment
[22:46] <teratorn> perhaps I can use image magick
[22:46] <durandal_1707> ffplay can display yuv420p just fine (though i think it messes up color range sometimes, i dunno what is workaround, -vf colormatrix?)
[22:47] <durandal_1707> image magick is so memory hungry, last time i used it ...
[22:47] <durandal_1707> generaly there is rawvideo demuxer
[22:48] <durandal_1707> and with it you can play raw frames with mplayer1/2/mpv
[22:49] <teratorn> i really wish there was some kind of text-based standard for YUV images
[22:49] <teratorn> like there are for RGB
[22:49] <durandal_1707> there is, it just some codecs use different colors, so you use colormatrix then ....
[22:50] <llogan> i wonder if graphicsmagick is a memory hog too
[22:50] <durandal_1707> so you will get correct colors with mplayer2/mpv
[22:50] <teratorn> durandal_1707: there is? what?
[22:51] <durandal_1707> teratorn: different color spaces
[22:51] <durandal_1707> BT.709, BT.601, SMPTE-240M, FCC, ...
[22:52] <durandal_1707> it really depends on your source
[22:54] <durandal_1707> and i dunno if imagemagick viewer displays correct color
[22:55] <durandal_1707> it certainly does not have opengl output IIRC
[22:55] <durandal_1707> _if_ opengl is relevant at all in your scenarion...
[22:57] <teratorn> nah, i'm just debugging a thing that makes test frames
[22:57] <teratorn> and need to dump/view them
[22:57] <durandal_1707> but there is bunch of application that use lavc/lavf/swscale to display images ...
[00:00] --- Tue Feb 19 2013
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