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September 2013
- 1 participants
- 58 discussions
[00:06] <cone-47> ffmpeg.git 03Vignesh Venkatasubramanian 07master:1d0d2201b4f4: Adding a line to Changelog about Opus Matroska Support
[00:06] <cone-47> ffmpeg.git 03Vignesh Venkatasubramanian 07master:b7bd68853d67: lavf/matroska: Removing experimental suffix for Opus
[07:39] <ubitux> wtf...
[10:56] <durandal_1707> michaelni: how you got crash that fixes 51dab60c7b91a21ec280c7d3042aa8350aae048f ?
[10:57] <durandal_1707> i always get same numbers with bytewidth and linesize
[11:18] <ubitux> Daemon404: fyi, vid.stab merged my work for linesize fixes
[11:18] <ubitux> and i fixed the green artefacts
[11:18] <ubitux> so it's usable now
[11:21] <Daemon404> ubitux, "usable"
[11:21] <Daemon404> maybe by hand
[11:22] <ubitux> ?
[11:22] <Daemon404> but it's e.g. cropping is still busted
[11:22] <Daemon404> unless you tweak it
[11:22] <Daemon404> youll see random black borders appear, depending on how unstable it is
[11:22] <Daemon404> it fails to calculate how to much to rop properly
[11:22] <Daemon404> not useful in a pragmatic waya :P
[11:22] <Daemon404> -a
[11:22] <ubitux> probably
[11:47] <cone-915> ffmpeg.git 03Georg Martius 07master:ad96482d67b2: avfilter/vidstabtransform: allow negative zoom.
[12:07] <Daemon404> ubitux, i maybe look into it one day
[12:07] <Daemon404> i mean.. in theory it's possible to calculate exactly how much to crop
[12:07] <Daemon404> given a certain transform/shear
[12:10] <ubitux> i'm working on it currently
[12:10] <Daemon404> oh
[12:10] <Daemon404> well then.
[12:10] Action: Daemon404 goes back to being laz
[12:10] <Daemon404> y
[12:11] <ubitux> it will probably be merged within ffmpeg
[12:12] <ubitux> in an undefined future
[12:15] <Daemon404> ubitux, i hope you also mean to drop 99% of the code
[12:15] <Daemon404> which is all available in e.g. swscale
[12:15] <Daemon404> ;)
[12:15] <ubitux> yeah probably
[12:15] <ubitux> and benefit from our optims...
[12:17] <Daemon404> ubitux, swscale is actually very slow in man instances :P
[12:17] <Daemon404> compaed to how it could be
[12:17] <Daemon404> e.g. extra conversions, etc
[12:17] <durandal_1707> what to drop from swscale?
[12:17] <Daemon404> durandal_1707, it's an architecture problem
[12:17] <Daemon404> not easily fixed
[12:17] <ubitux> Daemon404: vid.stab code is really naive
[12:18] <ubitux> so it will be hardly slower
[12:18] <ubitux> note that the vid.stab performances are particularly bad
[12:18] <ubitux> not*
[12:18] <Daemon404> oh
[12:18] <Daemon404> i know
[12:18] <Daemon404> it's super slow
[12:18] <ubitux> not that much
[12:18] <ubitux> IMO
[12:18] <Daemon404> i mean a copy is bundled with opencv.... kind of slowness
[12:18] <Daemon404> king*
[12:18] <Daemon404> ;)
[12:19] <ubitux> huh? what's this opencv thing about?
[12:20] <Daemon404> ubitux, opencv includes a copy of vid.stab
[12:20] <Daemon404> bundled
[12:20] <Daemon404> is all.
[12:20] <ubitux> wtf
[12:20] <Daemon404> opencv is horribly managed...
[12:20] <ubitux> ok..
[12:20] <wm4> what's opencv again?
[12:21] <durandal_1707> we have opencv wrapper, so lets dicht both filters
[12:21] <Daemon404> used for stuff liek object tracking
[12:21] <Daemon404> facial recog
[12:21] <Daemon404> etc
[12:21] <nevcairiel> open library of video algorithms
[12:21] <nevcairiel> rather generic collection of everything :p
[12:21] <Daemon404> e.g. recognize a banana in a photo
[12:21] <wm4> sounds useful
[12:21] <Daemon404> it is
[12:21] <Daemon404> but to do so you have to train it
[12:21] <Daemon404> with 100s or 1000s of photos of bananas
[12:21] <Daemon404> :D
[12:22] <wm4> I don't have that many bananas
[12:22] <durandal_1707> lool
[12:22] <wm4> speaking of external stuff, how useful is that frei0r stuff?
[12:23] <Daemon404> not useful
[12:23] <wm4> and why doesn't lavfi have wrappers for ladspa and sox?
[12:23] <Daemon404> sox wrapper is in swresample
[12:23] <Daemon404> iric.
[12:23] <Daemon404> er, iirc.
[12:23] <wm4> only for resampling, isn't it
[12:23] <Daemon404> yeah
[12:24] <Daemon404> there was a vf_sox.c on the ML from.. stefano
[12:24] <Daemon404> i think
[12:24] <Daemon404> one time.
[12:24] <durandal11707> lavfi does not have ladspa wrappers because nobody wrote it
[12:25] <wm4> I mean, after all mplayer has a ladpsa wrapper
[12:25] <wm4> and isn't all mplayer filter code considered holy to lavfi devs?
[12:26] <ubitux> 12 filters left
[12:26] <Daemon404> wm4, holy?
[12:26] <Daemon404> nah
[12:26] <ubitux> well, 12 .c
[12:26] <ubitux> less then
[12:26] <Daemon404> perhaps "hole-y"
[12:26] <Daemon404> :)
[12:26] <durandal11707> the problem is only libmpcodecs is considered holy, af2 not
[12:26] <durandal11707> libaf2 was never merged, ask michaelni why
[12:27] <durandal11707> ah, perhaps because lavfi did not have audio support then?!
[12:27] <wm4> don't give him bad ideas
[12:28] <ubitux> :D
[12:45] <wm4> why does AVPicture exist?
[12:45] <wm4> it duplicates AVFrame stuff
[12:45] <wm4> historic reasons?
[12:47] <durandal11707> the big question is what from libaf2 is considered worthy enough to be ported/reimplemented?
[12:49] <wm4> not much
[12:49] <wm4> it's a big loads of crap, basically
[12:50] Action: wm4 looks at af_sweep
[12:56] <durandal11707> ubitux: add contrast and gamma to hue and it will be 2 less
[12:57] <durandal11707> perhaps i should finally write adelay filter?
[12:57] <durandal11707> and port pullup
[12:59] <durandal11707> but i want to make transpose 6x faster
[13:10] <BoR0> I have a C++ library that wraps around libffplay, and instead of doing while (position != y...) fetch audio from ffmpeg I need to fill packets to directx as soon as ffplay has data for me
[13:10] <BoR0> does anyone have any experience with DirectSound? basically I need a queue management with audio buffer rather than setting notifications on position and filling buffer accordingly. is it possible to do this with plain DirectSound API or does there exist any nice wrapper for this?
[13:16] <wm4> libffplay? what
[13:19] <BoR0> yes, I have converted ffplay to a libffplay
[13:19] <BoR0> and exposing methods that will give me raw video and audio data
[13:28] <durandal11707> why this does not work: ffmpeg -f lavfi -i "aevalsrc=0|0|0|0" -lavfi "channelsplit=channel_layout=4.0:[A1][A2][A3][A4];[A1][A2] amerge=inputs=2 [o0];[A3][A4] amerge=inputs=2 [o0]" -f null -
[13:29] <wm4> durandal11707: it hates you
[13:42] <durandal11707> anyone?
[13:43] <wm4> well, maybe you should say WHAT "not works" means
[13:43] <wm4> because "it doesn't work" is the worst problem description you could possibly give
[13:46] <durandal11707> it gives me: Output pad "default" with type audio of the filter instance "Parsed_amerge_1" of amerge not connected to any destination
[13:48] <durandal11707> ah, i need to add -map '[o0]' -map '[o1]'
[13:49] <cone-915> ffmpeg.git 03Michael Niedermayer 07master:53102fa2f044: Revert "avfilter/vf_noise: dont corrupt the picture outside width x height"
[13:49] <cone-915> ffmpeg.git 03Michael Niedermayer 07master:126ba688267e: avfilter/vf_noise: rename linesize to bytewidth
[13:49] <cone-915> ffmpeg.git 03Christian Schmidt 07master:d9cdb7d8d6d8: pcm-dvd: Support channel configuration changes
[13:49] <cone-915> ffmpeg.git 03Michael Niedermayer 07master:3c89ec281695: Merge commit 'd9cdb7d8d6d828bb5497ea3f0fd7edd2f3f6cc30'
[13:50] <durandal11707> michaelni: but that is saste ideas to name it linesize and not stride/or bytewidth
[13:51] <durandal11707> have bounties be added to trac?
[13:52] <durandal11707> gmake: *** No rule to make target `doc/avtools-common-opts.texi', needed by `doc/ffmpeg.pod'. Stop.
[13:53] <cone-915> ffmpeg.git 03Christian Schmidt 07master:aca25104ea9c: pcm-dvd: Minor leftovers
[13:53] <cone-915> ffmpeg.git 03Michael Niedermayer 07master:fc477263fadd: Merge remote-tracking branch 'qatar/master'
[15:10] <durandal11707> hmm channelsplit filter clones input frames, thus it wastes memory
[15:40] <durandal11707> hmm that is correct only if reference counting is disabled
[17:56] <durandal_1707> how to make one filter calls another one ?
[18:02] <wm4> use avisynth
[18:02] <wm4> </troll>
[18:03] <gnafu> Seems legit.
[18:07] <durandal_1707> well on the other hand calling reconfiguration from filter itself
[18:08] <durandal_1707> so that auto filter insertion like scale kicks in
[18:30] <durandal_1707> one more filter under way
[20:38] <llogan> "buttracker"
[20:38] <Daemon404> with opencv?
[20:58] <cone-25> ffmpeg.git 03Michael Niedermayer 07master:a25585bb5072: avutil/pixdesc: Prevent minor array overread in ff_check_pixfmt_descriptors()
[20:58] <cone-25> ffmpeg.git 03Michael Niedermayer 07master:f3ba91a3f1d9: avutil/pixdesc: dont try to use av_read_image_line() with bayer formats
[21:18] <cone-25> ffmpeg.git 03Paul B Mahol 07master:b8b398027cc2: avfilter/video: remove commented out cruft
[21:29] <cone-25> ffmpeg.git 03Paul B Mahol 07master:3dcd3e08f6a4: avcodec/proresdec2: remove duplicated tables
[21:40] <durandal_1707> llogan: how this sounds "If number of given delays is smaller than number of channels all remaining channels will not be delayed."
[21:48] <cbsrobot_> durandal_1707 and what happens if it's bigger ?
[21:48] <durandal_1707> ignored
[21:49] <cbsrobot_> then add a comma between channels and all
[21:50] <durandal_1707> why?
[21:58] <cbsrobot_> because there are two verbs and two subjects
[21:59] <cbsrobot_> but I'm not native speaker
[22:08] <durandal_1707> what ffv1 doc change i need to update?
[22:09] <llogan> durandal_1707: sounds good to me
[22:22] <durandal_1707> llogan: adding twitter link to main page?
[22:29] <llogan> next to the g+?
[22:31] <durandal_1707> whatever
[22:32] <durandal_1707> all icons could be on right side....
[22:33] <durandal_1707> then you also need to create fb one
[22:34] <durandal_1707> or better not...
[22:37] <llogan> yes, icons would be nicer on right, but i don't really care about the fb one.
[22:37] <llogan> maybe i'll make patches over the weekend
[22:41] <cone-25> ffmpeg.git 03Paul B Mahol 07master:23bd0335af70: avcodec/g2meet: remove redundant log message
[23:30] <cone-25> ffmpeg.git 03Michael Niedermayer 07master:6c07e41f4302: avcodec/mjpegdec: check for malloc failure of blocks[] and last_nnz[]
[00:00] --- Sat Sep 14 2013
1
0
[00:00] <fazias> it was included in the pastebin, for how I found out the point where to split, well...
[00:01] <fazias> that involves a script that I copied from a stackoverflow answer
[00:39] <willwh> hi guys - I am curious, I've been testing with ffprobe: http progressive delivered content I can get bitrate data from FLVs
[00:39] <willwh> no dice with mp4 files though
[00:40] <willwh> and I've tried quite a few
[00:40] <willwh> runnin' 'em through mp4box they are valid
[00:40] <willwh> is that a codec thing, or, where would I be able to get some more info?
[00:40] <willwh> I'm hunting - if anyone has any experience - give me a yell :) (I'll idle for a few days)
[00:42] <willwh> ah............ scratch that :)
[00:42] Action: willwh slaps himself
[01:17] <chrisballinger> hey everybody
[01:18] <chrisballinger> other than reading the source, are there any examples on how to do the equivalent of "ffmpeg -i input.mp4 -f mpegts -vcodec copy -acodec copy -vbsf h264_mp4toannexb output.ts" with libavformat?
[01:19] <chrisballinger> ive been studying the muxing.c and demuxing.c examples in the docs but it's a little hard to wrap my head around those examples
[01:27] <vl4kn0> Hi, does format of audio frame after decoding it depend on codec used? I want to detect silent frame in video and I guess I just count all values that are close to zero and then compute percentage of the values and if the percentage is big enough (>98%) then the frame could be considered silent. I'm just not sure whether this algorithm can be applied to all codes?
[01:38] <llogan> vl4kn0: http://ffmpeg.org/ffmpeg-filters.html#silencedetect
[01:40] <vl4kn0> llogan: aha! I tried to find this but I was looking at the libav documentation
[01:40] <llogan> JoeyJoeJo: Junior Shabadoo
[01:41] <llogan> to me, libav means the FFmpeg libraries.
[01:43] <vl4kn0> except, there is no silecedetect filter in libavfilter
[05:56] <praveenmarkandu> hi. is there a way to transcode apple prores to h264
[05:56] <praveenmarkandu> but directly to main profile
[05:57] <praveenmarkandu> when doing prores > x264 it only allows me high422
[06:03] <relaxed> praveenmarkandu: use -profile main
[06:04] <relaxed> but keep in mind that the profile is also determined by the frame size/rate and other factors.
[06:05] <relaxed> http://en.wikipedia.org/wiki/H.264/MPEG-4_AVC#Profiles
[06:07] <relaxed> er, I'm thinking of levels, but the profile does take color space and bit depth
[06:09] <relaxed> praveenmarkandu: pastebin you're command and output, then I'll be able to see what's going on
[06:09] <relaxed> your*
[06:18] <praveenmarkandu> ok
[06:20] <praveenmarkandu> relaxed: http://pastebin.com/XHfQRNdj
[06:25] <relaxed> praveenmarkandu: add -pix_fmt yuv420p
[06:26] <praveenmarkandu> oh. is the colour space not letting me encode at main profile?
[06:26] <relaxed> color space and bit depth
[06:27] <relaxed> your source is 10bit and yuv422p
[06:28] <relaxed> main only supports 8bit yuv420p
[06:29] <praveenmarkandu> ok. gotcha
[06:29] <praveenmarkandu> thanks
[06:34] <praveenmarkandu> relaxed: btw, for an h264 10mbps source file ,if i do a -vcodec copy, and also use -b:v 2000k will it transcode it to 2mbps?
[06:36] <relaxed> -vcodec copy is a stream copy
[06:37] <relaxed> it's one or the other
[06:38] <praveenmarkandu> oh okay. so no
[06:52] <chrisballinger> heya are there any good examples of doing bitstream copies (and with h264_mp4toannexb) with just libavformat?
[08:54] <praveenmarkandu> hi guys
[08:54] <praveenmarkandu> is there a way to fix the bitrate on a prores file?
[08:54] <praveenmarkandu> or do i just rely on -profile:v and let that do the work?
[09:33] <cbsrobot_> praveenmarkandu: in prores_ks you can set bits_per_mb and/or mbs_per_slice
[09:33] <praveenmarkandu> oh. thanks
[09:34] <praveenmarkandu> i assume the difference between prores and prores_ks is minimal
[09:34] <praveenmarkandu> in terms of quality
[09:35] <cbsrobot_> honestly I don't know exactly, but prores_ks supports 4444
[09:36] <praveenmarkandu> oh okay
[09:36] <praveenmarkandu> thanks
[09:36] <praveenmarkandu> got questions answered in #ffmpeg. today was a good day
[09:39] <cbsrobot_> praveenmarkandu: well then just do ma a favor and help someone else today too ;-p
[09:41] <praveenmarkandu> my knowledge of ffmpeg is very rudimentary
[10:44] <rooty> i know this is likely not the right channel but i was wondering if anyone know of a method to take an input stream (icecast) convert it to lower quality stream rates and rebroadcast ?
[11:12] <Jenser> hello everyone. is there a way to write the scale=xxx:yyy command only once when merging serveral files together with -filter_complex?
[11:14] <durandal_1707> depends what filters you use
[11:21] <Jenser> i am trying to combine several videos (already made to the same format), for not losing to much quality the first step was to convert them to a common intermediate format (19020x1080@10MBit), now i want to combine them to one long clip. concat: ... worked fine with -vf yadif (... ),scale=xxx:yyy as output, but with -filter_complex this seams not to work. for me it looks like i have to define the size every video input file, although it's all
[11:26] <kein> Hi there
[11:27] <durandal_1707> Jenser: it get cut of, could you pastebin both commands?
[11:28] <kein> I got some troubles with audio filers "channelsplit" end "amerge". When processing 4channels -> 4 streams it's fine. When 4 channels -> 2 streams. the second stream is mixed ? First stream is fine ???
[11:30] <BoR0> does anyone have any experience with DirectSound? basically I need a queue management with audio buffer rather than setting notifications on position and filling buffer accordingly. is it possible to do this with plain DirectSound API or does there exist any nice wrapper for this?
[11:31] <BoR0> I have a C++ library that wraps around libffplay, and instead of doing while (position != y...) fetch audio from ffmpeg I need to fill packets to directx as soon as ffplay has data for me
[11:32] <kein> ffloger > there you go http://pastebin.com/HmWySybD
[11:34] <kein> I got some troubles with audio filers "channelsplit" end "amerge". When processing 4channels -> 4 streams it's fine. When 4 channels -> 2 streams. the second stream is mixed ? First stream is fine ??? Find details here http://pastebin.com/HmWySybD
[11:45] <kein> I got some troubles with audio filters "channelsplit" end "amerge". When processing 4channels -> 4 streams it's fine. When 4 channels -> 2 streams. the second stream is mixed ? First stream is fine ??? Find details here http://pastebin.com/HmWySybD
[11:45] <durandal_1707> that is not COMPLETE output
[11:53] <kein> durandal_1707 http://pastebin.com/J6WaCFMQ
[11:54] <kein> I got some troubles with audio filters "channelsplit" end "amerge". When processing 4channels -> 4 streams it's fine. When 4 channels -> 2 streams. the second stream is mixed ? First stream is fine ??? Find details here http://pastebin.com/J6WaCFMQ
[11:54] <Jenser> @durandal http://pastebin.com/Es2pSFMF
[11:54] <Jenser> this command ignores the scale @ -vf filter
[11:55] <Jenser> thatswhy i want to move this command to the complex_filter for defining the output video size
[11:56] <kein> Jenser : put all the command "yadif, scale" in the filter complex. It should do the trick
[11:57] <kein> Jenser : -filter_complex "concat, yadif=... , scale=..."
[11:57] <Jenser> to which position? don't want to define this for each video input file
[11:57] <Jenser> the documentation for complex filter isn't the very best, i think...
[11:58] <kein> Jenser : first you "concat" so the scale and yadif are placed after, so it will process the output of concat
[11:59] <kein> Jenser : and i agree with the documentations, not enough examples :s
[11:59] <kein> [AFK] dinner time
[11:59] <Jenser> mahlzeit
[12:06] <durandal_1707> kein: i do not see how amerge could mix channels
[12:37] <kein> [back]
[12:38] <kein> Durandal_1707 : I do not see too, but in audition channels 3&4 are mixed. Don't no if bug or if I missed something
[12:39] <kein> Hi guys/ I got some troubles with audio filters "channelsplit" end "amerge". When processing 4channels -> 4 streams it's fine. When 4 channels -> 2 streams. the second stream is mixed ? First stream is fine ??? Find details here http://pastebin.com/J6WaCFMQ
[12:57] Last message repeated 1 time(s).
[12:58] <durandal11707> kein: i do not see how it could be mixed
[12:58] <durandal11707> by mixed, what you mean?
[13:00] <kein> durandal11707 : when listening with headphones and selecting audio "french", left and right sounds are different (test file). when selecting "english" left and right are mixed. Original file is not
[13:00] <kein> durandal11707 I got the files and a good upload if you want
[13:02] <kein> durandal11707 : by mixed I mean in left speaker I can ear both sounds (3 & 4) same with the right speaker
[13:03] <durandal11707> that looks like bug
[13:04] <kein> durandal11707 : i'm not the kind of guy to scream bug quickly. maybe you can test on your hardware ? ^^
[13:22] <kein> durandal11707 : when amerge with source (4 streams - mono) I don't get the bug...
[13:28] <durandal11707> perhaps you could use asplit and pan filter?
[13:28] <durandal11707> this could also be faster
[13:30] <kein> durandal11707 : asplit : copy input to multiple output... it's not the goal here. (if I get it)
[13:32] <durandal11707> no you split it to 2 outputs and use pan on all of them
[13:33] <durandal11707> you just do downmix from 4 to 2(stereo)
[13:33] <kein> durandal11707 : i'm not familier with this filter; I'll try it right now ;)
[13:37] <kein> durandal11707 : can we mp ?
[13:46] <Jenser> @kein can you propably tell me the correct positioning of the scale command? i didn't get it running yet. command is so far ... -filter_complex '[x:y] (...) concat=n=5:v=1:a=1 [v] [a],scale=xxx:yyy' -map
[13:46] <Jenser> this obviously isn't the right position for scale
[13:48] <kein> Jenser : -filter_complex '[0:0] [0:1] [1:0] [1:1] [2:0] [2:1] [3:0] [3:1] [4:0] [4:1] concat=n=5:v=1:a=1 [v] [a], yadif=1:-1:1,scale=720:576'
[13:50] <kein> oups : -filter_complex '[0:0] [0:1] [1:0] [1:1] [2:0] [2:1] [3:0] [3:1] [4:0] [4:1] concat=n=5:v=1:a=1 [v] [a], [V] yadif=1:-1:1,scale=720:576'
[13:53] <Jenser> output with label [v] does not exist in any defined filter graph or was already used elsewhere
[13:53] <Jenser> do i need to put this command behind the "-map" command?
[13:56] <kein> Jenser : mp
[13:58] <Jenser> urban dictionary tells me: mp = military police
[13:58] <Jenser> don't think you wanted to tell me this ;)
[14:01] <kein> privite message :p
[14:01] <kein> *private
[14:01] <kein> ffmpeg -i $FILES [audio-options] [video-options] -filter_complex '[0:0] [0:1] [1:0] [1:1] [2:0] [2:1] [3:0] [3:1] [4:0] [4:1] concat=n=5:v=1:a=1; [0:v] yadif=1:-1:1,scale=720:576' -y test.avi
[15:59] <vl4kn0> Hi, is it guaranteed that video and audio frame arrive in the same packet?
[16:52] <mcnulty_> hi
[16:54] <mcnulty_> I would like to change /tmp folder for recording/screencast...
[19:27] <chrisballinger> hi, are there any good examples on how to use libavformat to change the container of media without re-encoding?
[20:41] <ZaB|SHC|> hi there. i've got a problem applying two audio filters. i want to record from three sources, mix them into one stream and then increase the volume. my command at the moment looks like: ffmpeg -f pulse -i app1.monitor -f pulse -i app2.monitor -f pulse -i default -filter_complex 'amix=inputs=3[mixed]; [mixed]volume=volume=2[out]' test.mp3. this leads to: Output pad "default" with type audio of the filter instance "Parsed_volume_1" of
[20:41] <ZaB|SHC|> volume not connected to any destination
[20:53] <Barbariandude> Hi guys! I'm trying to stream to twitch.tv with both my computer's audio output and my microphone's input. All I'm getting if I put "-i default" is my microphone in really scratchy shitty quality. If I put "-i hw:0,0" I get alsa errors saying "Device in use"
[20:53] <Barbariandude> Anyone got any ideas?
[20:53] <llogan> ZaB|SHC|: add -map "[out]"
[20:54] <llogan> or remove [out]
[20:54] <llogan> but being explicit is probably better
[20:55] <Barbariandude> llogan: In the ffmpeg command itself, after "-f alsa"?
[20:55] <Barbariandude> llogan: Oh, responding to a diff guy
[20:55] <Barbariandude> llogan: nvm!
[20:55] <ZaB|SHC|> llogan: thx, removing [out] helped
[20:56] <llogan> ZaB|SHC|: you can shorten it: 'amix=inputs=3,volume=volume=2'
[20:57] <llogan> but again, it's usually a good idea to use named links
[20:58] <Barbariandude> fflogger: http://pastebin.com/hPqdMb8s
[20:58] <ZaB|SHC|> llogan: thx, i'll stick to the named links
[20:58] <llogan> fflogger is a bot
[20:58] <Barbariandude> llogan: Oops :P
[20:58] <llogan> the complete console output is missing
[20:58] <Barbariandude> llogan: Yup doing that now
[20:59] <Barbariandude> llogan: Ummm... this is a new error... http://pastebin.com/Ba2YXfdb
[21:00] <llogan> there is no context for the error without the actual, unscripted command
[21:01] <Barbariandude> llogan: I had an additional space
[21:01] <Barbariandude> llogan: Fixed that error
[21:02] <Barbariandude> llogan: http://www.twitch.tv/barbariandude
[21:02] <Barbariandude> I am both speaking into my mic and have music playing
[21:02] <Barbariandude> No alsa input is going through
[21:02] <Barbariandude> What's wrong with my command?
[21:03] <llogan> i don't know. i'm waiting for the console output.
[21:06] <Barbaria2dude> llogan: http://pastebin.com/BvTL1dzx
[21:08] <llogan> also show output of: arecord -l
[21:10] <Barbaria2dude> llogan: Ok, arecord -l showed card 0 device 0, so I changed "-i default" to "-i hw:0,0" and I got microphone output, but not actual audio. 1 second to put aplay -l and arecord -l into a pastebin
[21:10] <Barbaria2dude> http://bpaste.net/show/132196/
[21:11] <Barbaria2dude> http://bpaste.net/show/132197/
[21:12] <llogan> does it sound shitty if you use arecord? then you know it's not ffmpeg.
[21:13] <llogan> other than that...i don't know. burek knows more about alsa than i do. also see: https://trac.ffmpeg.org/wiki/Capturing%20audio%20with%20FFmpeg%20and%20ALSA
[21:13] <chrisballinger> hi, are there any good examples on how to use libavformat to change the container of media without re-encoding?
[21:15] <llogan> did you see docs/examples/demuxing.c and muxing.c?
[21:15] <llogan> i assume they may be useful, but I am only a cli tool user
[21:15] <Barbaria2dude> llogan: No, mic sounds fine in arecord. Mic also sounds fine in ffmpeg now that I put hw:0,0 instead of letting ffmpeg autoselect. Only problem is it's not recording the computer's audio output as well. aplay -l shows card 0 device 0 for output, which is the same as arecord -l which confuses me
[21:16] <chrisballinger> yeah i saw the examples but it is unclear how to do the equivalent of -vcodec copy
[21:16] <Barbaria2dude> llogan: So I'm not sure how to mix both into a stream
[21:17] <llogan> chrisballinger: try libav-user mailing list if you don't get an answer here or by looking at the code
[21:17] <llogan> Barbaria2dude: http://ffmpeg.org/ffmpeg-filters.html#amerge
[21:17] <chrisballinger> yeah ive been studying the code for the last few days
[21:18] <llogan> Barbaria2dude: https://trac.ffmpeg.org/wiki/AudioChannelManipulation
[21:20] <Barbaria2dude> llogan: Thank you for your help in narrowing this down. I'll ask #alsa why both my microphone and my audio output are both card 0 device 0.
[21:20] <llogan> good luck.
[21:58] <Barbaria2dude> llogan: Ok, apparently hw:0,0 is both my mic and my headphone output, it's just my headphone output is incredibly quiet. If both my mic and my headphone output are on the same subdevice, is there any way to alter the sound levels in the command?
[21:58] <Barbaria2dude> llogan: Or is it just a hard reality that I can't mix the audio levels?
[21:59] <Barbaria2dude> I've found numerous guides for mixing audio levels of different subdevices, nothing for the same subdevice
[22:16] <klaxa> Barbaria2dude: you will most likely need a mixer-board or two independent sources
[22:17] <klaxa> which would be two different subdevices then probably
[22:18] <Barbaria2dude> klaxa: I've managed to kludge the solution by just changing volume levels (increasing output to near-deafening levels and reducing the volume via a hardware thing on the headphones), now I'm facing a different problem
[22:18] <Barbaria2dude> klaxa: Video is fine on the stream, audio is cutting out every 20 seconds or so in time with "alsa buffer xrun"
[22:19] <klaxa> i never got those to stop :x
[22:19] <klaxa> maybe wrong sampling rate?
[22:19] <klaxa> that's the only thing that comes to mind
[22:20] <Barbaria2dude> klaxa: What sample rate would you recommend?
[22:20] <klaxa> the one alsa produces
[22:21] <klaxa> or maybe exactly half of it
[22:21] <klaxa> or something
[22:21] <klaxa> streaming 96khz is somewhat unncessary, don't you agree? :)
[22:22] <Barbaria2dude> klaxa: Absolutely, but where do I check alsa's sample rate?
[22:22] <Barbaria2dude> The one it produces, I mean
[22:22] <klaxa> ffmpeg will list it as an input
[22:23] <klaxa> but now that i think about it
[22:23] <klaxa> that shouldn't even matter
[22:23] <klaxa> alsa buffer xruns are mysterious to me
[22:24] <Barbaria2dude> klaxa: http://pastebin.com/egtP1cWe What should my sample rate be from that?
[22:25] <klaxa> 48khz right?
[22:26] <Barbaria2dude> klaxa: Giving me an incorrect parameter error on that
[22:27] <klaxa> pastebin the command line + complete output
[22:29] <Barbaria2dude> klaxa: http://pastebin.com/BnnUXq9B
[22:30] <klaxa> ah yes, good ol' flv container
[22:30] <klaxa> you will need to resample to 44.1khz
[22:30] <klaxa> no matter what
[22:30] <klaxa> because flv is weird like that
[22:31] <Barbaria2dude> klaxa: http://www.twitch.tv/barbariandude
[22:32] <Barbaria2dude> klaxa: Talking continuously, cutting out all the time
[22:33] <klaxa> new beer is always good
[22:33] <Barbaria2dude> Indeed.
[22:34] <Barbaria2dude> I need alcohol to prevent myself from tearing my computer into pieces
[22:35] <klaxa> i personally prefer to use pulseaudio for recording those things
[22:35] <Barbaria2dude> *sigh* I was hoping you wouldn't say that
[22:35] <Barbaria2dude> I'll use it if I have to
[22:35] <Barbaria2dude> But I hate it
[22:35] <klaxa> it's just not deterministic
[22:35] <klaxa> that is what i hate most about pulseaudio
[22:35] <klaxa> other than that, it's pretty nice
[22:35] <klaxa> leaving out the part where it eats CPU like a whore
[22:36] <Barbaria2dude> What I hate most is that it tries to be too clever and mutes random shit and I need to go to pavucontrol to unmute them
[22:36] <Barbaria2dude> No pulseaudio, I WANT both audio and teamspeak playing audio at the same time!
[22:36] <Barbaria2dude> *music and teamspeak
[22:37] <klaxa> there *should* be a setting for that
[22:38] <klaxa> i don't know what causes it either though
[22:44] <Barbaria2dude> klaxa: All this trouble to stream some ownage meepo dota 2 gameplay on gentoo :P
[22:45] <Barbaria2dude> Hopefully it'll be worth it
[22:48] <klaxa> now imagine all the "trouble" writing the code you are using :P
[22:55] <willwh> klaxa: whores eat cpu?
[22:55] <klaxa> maybe?
[22:55] <willwh> :]
[22:55] <willwh> hahaha
[22:56] <klaxa> i don't know any whores so i wouldn't know :V
[00:00] --- Sat Sep 14 2013
1
0
[11:18] <cone-47> ffmpeg.git 03Diego Biurrun 07master:5045f6bdfabf: configure: Fix wrong reference to user support mailing list
[11:18] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:ae77f06c691d: Merge commit '5045f6bdfabfa5d3374756d2c23f762fb042acae'
[11:26] <cone-47> ffmpeg.git 03Martin Storsjö 07master:48d6556dd46d: dsicin: Add some basic sanity checks for fields read from the file
[11:26] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:7e88dbe9358f: Merge commit '48d6556dd46d4f4fac10d0f4a819e314887cd50e'
[11:39] <cone-47> ffmpeg.git 03Martin Storsjö 07master:8d928023f953: smacker: Don't return packets in unallocated streams
[11:39] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:33cf70a8b88c: Merge commit '8d928023f953a28692ba27071a448259134b103b'
[11:48] <cone-47> ffmpeg.git 03Martin Storsjö 07master:710b0e270259: smacker: Avoid integer overflow when allocating packets
[11:48] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:c16cd5559349: Merge commit '710b0e27025948b7511821c2f888ff2d74a59e14'
[12:01] <cone-47> ffmpeg.git 03Martin Storsjö 07master:d002fce24a72: smacker: Check malloc return values
[12:01] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:d510d70df02b: Merge commit 'd002fce24a7206e95f9492031f45adc35711e183'
[12:52] <cone-47> ffmpeg.git 03Martin Storsjö 07master:0679cec6e880: smacker: Make sure we don't fill in huffman codes out of range
[12:52] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:f8932d990228: Merge commit '0679cec6e8802643bbe6d5f68ca1110a7d3171da'
[13:06] <cone-47> ffmpeg.git 03Martin Storsjö 07master:f3d57dc69145: smacker: Free memory properly if the init function fails
[13:06] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:af539319af3a: Merge commit 'f3d57dc69145f1b7acb4870da9ce60378190a1fd'
[13:38] <cone-47> ffmpeg.git 03Martin Storsjö 07master:49568851bf17: shorten: Use a checked bytestream reader for the wave header
[13:38] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:a44d35d90ffa: Merge commit '49568851bf1700e3d9ea9cda29208d0df3c2c38b'
[13:45] <cone-47> ffmpeg.git 03Martin Storsjö 07master:b26742cc3085: shorten: Break out of loop looking for fmt chunk if none is found
[13:45] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:4caf8c51de93: Merge commit 'b26742cc308552f242ee2bf93b07a3ff509f4edc'
[13:57] <cone-47> ffmpeg.git 03Martin Storsjö 07master:183b9d843a95: ape: Don't allow the seektable to be omitted
[13:57] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:fcfbc58a1cd7: Merge commit '183b9d843a9533774fabd3984a52f3987001acbc'
[14:07] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:0d61f2600107: zmbvdec: Check the buffer size for uncompressed data
[14:07] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:d6e650d88594: Merge commit '0d61f260010707f3028b818e8b24598e1a83d696'
[14:20] <cone-47> ffmpeg.git 03Martin Storsjö 07master:459f2b393a3f: mpc8: Check the seek table size parsed from the bitstream
[14:20] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:ad9a877a6d77: Merge commit '459f2b393a3f89ed08d10fbceb4738d1429f268e'
[14:32] <cone-47> ffmpeg.git 03Martin Storsjö 07master:17d57848fc14: mpc8: Make sure the first stream exists before parsing the seek table
[14:32] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:83d2b7e9a25e: Merge commit '17d57848fc14e82f76a65ffb25c90f2f011dc4a0'
[14:39] <cone-47> ffmpeg.git 03Martin Storsjö 07master:0f678c0214dc: aic: Validate values read from the bitstream
[14:39] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:72abfba88458: Merge commit '0f678c0214dccb355ed8955077a2bea46984fbc8'
[14:43] <durandal_1707> libavcodec/aic.c:226:43: warning: comparison of unsigned expression < 0 is always false [-Wtautological-compare]
[14:44] <durandal_1707> nice "fix" that does not fix anything
[14:45] <Daemon404> durandal_1707, it is not unsigned in libav
[14:45] <durandal_1707> than even worse
[14:57] <cone-47> ffmpeg.git 03Martin Storsjö 07master:b1e469885362: g2meet: Allocate cursor buffers large enough to fit the aligned width
[14:57] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:fa9e8bafc832: Merge commit 'b1e469885362febce3d9a4678624e44a92f77da9'
[15:02] <cone-47> ffmpeg.git 03Diego Biurrun 07master:e2cd766ca1b9: configure: Mention that icl does not build both static and shared libs
[15:02] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:07f6a3d78734: Merge commit 'e2cd766ca1b9bb2ba72aca5f6ba5d8be15743984'
[15:07] <cone-47> ffmpeg.git 03Vittorio Giovara 07master:c0779a67e85d: mp3: add .mpa extension
[15:07] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:44fb7de8ee6d: Merge commit 'c0779a67e85df856904ee6fab760c4233d4c2be5'
[15:17] <cone-47> ffmpeg.git 03Stefano Sabatini 07master:52ed1d6d5847: MAINTAINERS: move sections from lavc to lavu
[15:27] <cone-47> ffmpeg.git 03Alexandra Khirnova 07master:00a63bfb87af: matroskaenc: Fix stray pointers left over from av_reallocp_array refactoring
[15:27] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:ba5e77814e5c: Merge remote-tracking branch 'qatar/master'
[15:35] <JEEB> also related to this channel as well, so linking what I just posted on matroska-devel: http://lists.matroska.org/pipermail/matroska-devel/2013-September/004567.ht…
[16:16] <cone-47> ffmpeg.git 03Paul B Mahol 07master:b211607b5c97: avfilter: various cosmetics
[16:24] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:d2d96f9e47d6: vformat/smacker: use av_freep() to ensure no stale pointers remain
[16:24] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:59a08d193963: avcodec/smacker: use the correct field instead of a hardcoded value for the index check in smacker_decode_tree()
[16:24] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:fbab9a7c2267: avcodec/smacker: remove check that has become redundant
[16:24] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:f144168d9b7c: avcodec/smacker: fix some memleaks
[16:50] <durandal_1707> michaelni: can you please comment on noise patch?
[16:56] <cone-47> ffmpeg.git 03Paul B Mahol 07master:3eba83daedaf: avfilter/vf_noise: unbreak filter when inline assembly is not present
[17:21] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:51dab60c7b91: avfilter/vf_noise: dont corrupt the picture outside width x height
[17:21] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:bfe144587177: avcodec/aic: fix "warning: comparison of unsigned expression < 0 is always false"
[17:26] <durandal_1707> michaelni: your change to noise filter made n->linesize redundant/unused
[17:27] <durandal_1707> michaelni: how you managed to corrupt picture?
[18:18] <durandal_1707> fflogger: ?
[18:19] <durandal_1707> actually its cone, silly me
[18:37] <cone-47> ffmpeg.git 03Paul B Mahol 07master:1b99514dfdc1: avcodec/smacker: use init_get_bits8()
[19:10] <durandal_1707> michaelni: so, apropo noise commit, i'm still waiting for answer
[19:23] <durandal_1707> but not about that damn padding area but for bytewidth change which looks to me completly unnecessary
[19:25] <ubitux> durandal_1707: what about noise + pad + overlay on the right ?
[19:25] <durandal_1707> yes, what about it?
[19:25] <ubitux> i didn't follow but noise is writing in the linesize padding?
[19:26] <durandal_1707> no, it does not right now
[19:26] <durandal_1707> but spp does
[19:28] <ubitux> ah i misunderstood the issue then
[19:46] <michaelni> spp is intended to work on decoder output (well normally) that is 8x8 DCT based so the buffer normally wont be width%8 != 0
[19:47] <durandal_1707> ...normally ....
[19:48] <durandal_1707> but i see your point, so i moved on...
[19:48] <durandal_1707> i hope to conver that inline asm to yasm
[19:56] <tg2> ahoy
[19:56] <tg2> who maintains this? http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide
[19:58] <durandal_1707> people that use Ubuntu
[19:59] <tg2> needs some updating as it doesn't work
[19:59] <tg2> who should I email
[20:00] <tg2> "MarkZV"
[20:04] <tg2> k i'll push the change up
[20:05] <llogan> tg2: what does not work about it?
[20:05] <tg2> you are putting yasm in ~/bin
[20:05] <tg2> but when compiling x264 it doesn't find yasm
[20:05] <tg2> you need to export PATH=~/bin:$PATH
[20:05] <tg2> then itw orks fine
[20:06] <llogan> what is your ubuntu version?
[20:06] <tg2> 12.04 LTS
[20:06] <tg2> . ~/.profile
[20:06] <tg2> doesn't do it
[20:06] <tg2> plus after all is said and done it is installed to ~/bin
[20:07] <tg2> which isn't in the path
[20:07] <tg2> after installing yasm
[20:07] <tg2> when following the guide verbatim
[20:07] <tg2> "which yasm" returns nothing
[20:08] <tg2> which is what x264 configure/build does
[20:08] <llogan> i'll take a look
[20:08] <tg2> http://pastie.org/pastes/8320613/text?key=e6qnnxnoznixtekhta5z7q
[20:08] <tg2> thats what I have
[20:08] <tg2> substituded libfaac since it install nicely from apt
[20:08] <tg2> and added zlib
[20:08] <tg2> and pic
[20:08] <tg2> to ffmpeg
[20:08] <tg2> but nothing else majorly changed
[20:09] <tg2> you can copy/paste that into terminal and install ffmpeg
[20:09] <llogan> why did you --enable-pic?
[20:09] <tg2> was in a config I used previously
[20:09] <tg2> not necessary
[20:10] <llogan> --enable-zlib is superfluous
[20:10] <tg2> k, they can be omitted then
[20:10] <tg2> but the path issue remains
[20:11] <llogan> ok. thanks for mentioning it. i'll take a look at it later today.
[20:11] <tg2> otehr than that, all good
[20:21] <durandal_1707> michaelni: what scan order is on your mind?
[21:01] <durandal_1707> if i use recursive approach so that each matrix get into cache, i'm afraid will get out of stack
[21:06] <michaelni> recursive needs just O(log n) stack
[21:08] <durandal_1707> where n is?
[21:10] <durandal_1707> hmm
[21:12] <michaelni> n as in pixels or dimension or width if its a square
[21:13] <michaelni> also in theory you can implement it without stack or any equivalnent
[22:28] <cone-47> ffmpeg.git 03Carl Eugen Hoyos 07master:52cf08b4c885: Store the video bit_rate in the context when muxing mxf.
[22:28] <cone-47> ffmpeg.git 03Carl Eugen Hoyos 07master:d73565d5ddf4: Use rc_max_rate if no video bit_rate was specified when muxing mxf_d10.
[22:28] <cone-47> ffmpeg.git 03Michael Bradshaw 07master:7412a4a48f23: libopenjpegdec.c: Correctly scale gray16 output if precision < 16
[22:28] <cone-47> ffmpeg.git 03Carl Eugen Hoyos 07master:f99f7f40257c: Set bits_per_raw_sample when reading j2k frames.
[22:28] <cone-47> ffmpeg.git 03Michael Niedermayer 07master:e2774b3e3b51: Merge remote-tracking branch 'cehoyos/master'
[23:20] <cone-47> ffmpeg.git 03Carl Eugen Hoyos 07release/1.1:b5e2ef771d89: Store the video bit_rate in the context when muxing mxf.
[23:20] <cone-47> ffmpeg.git 03Carl Eugen Hoyos 07release/1.1:9fde820d8e3c: Use rc_max_rate if no video bit_rate was specified when muxing mxf_d10.
[23:20] <cone-47> ffmpeg.git 03Carl Eugen Hoyos 07release/1.2:3a6fbb0c8e06: Store the video bit_rate in the context when muxing mxf.
[23:20] <cone-47> ffmpeg.git 03Carl Eugen Hoyos 07release/1.2:b4f90013ab4f: Use rc_max_rate if no video bit_rate was specified when muxing mxf_d10.
[23:20] <cone-47> ffmpeg.git 03Carl Eugen Hoyos 07release/2.0:aaef59d53538: Store the video bit_rate in the context when muxing mxf.
[23:20] <cone-47> ffmpeg.git 03Carl Eugen Hoyos 07release/2.0:09bc4be3dbb3: Use rc_max_rate if no video bit_rate was specified when muxing mxf_d10.
[00:00] --- Fri Sep 13 2013
1
0
[01:22] <theo_> I followed UbuntuCompilationGuide and did all steps, but when I do "ffmpeg 2>&1 | head -n1" I get "The program 'ffmpeg' is currently not installed." Its a Ubuntu Server 12.4. Anybody have any idea?
[01:31] <cbsrobot_> theo_: where did you install it ?
[01:31] <theo_> cbsrobot:home directory
[01:31] <cbsrobot_> and is it in your PATH env variable ?
[01:32] <cbsrobot_> else you have to specify the path to ffmpeg: /hme/joe/bin/ffmeg
[01:33] <cbsrobot_> sorry for the typos
[01:33] <theo_> cbsrobot: ah ofcourse. All make clear now, thanks! =)
[01:37] <vl4kn0> I have a basic question that might not be fit for this IRC channel but I don't know where else to ask. How is represented audio track in the audio frame? I know that colorspace of video frame depends on codec used, but I can easily convert that to RGB24 but I know next to nothing about audio.
[02:21] <loller> hello! can anybody explain me how to show placement of x264 while compiling ffmpeg?
[02:27] <klaxa> ./configure --enable-libx264 [other options]
[03:09] <loller> klaxa: with this I get "ERROR: libx264 not found" ... that's why I'm asking how to show it to ffmpeg
[03:09] <klaxa> did you install libx264?
[03:10] <klaxa> including development files?
[03:10] <loller> I compiled it by myself
[03:10] <klaxa> did you run make install?
[03:10] <loller> but seems ffmpeg doesn't know where to find it
[03:10] <klaxa> if not you can add extra LDFLAGS
[03:11] <klaxa> ./configure --extra-ldflags="-L/path/to/libx264.so/" --enable-libx264
[03:11] <klaxa> where "/path/to/libx264.so/" is the directory containing libx264.so
[03:12] <klaxa> for example /home/user/build/x264/
[03:12] <loller> does it matter what param is first? I mean path and enablinf x264
[03:13] <klaxa> not really i think
[03:14] <loller> Thanks a lot ... trying right now
[03:24] <loller> hm ... the same eror
[03:24] <loller> maybe I compiled it incorrectly?
[03:24] <loller> I mean x264
[03:26] <loller> ./configure --prefix="/c/ffmpeg/x264" --enable-static
[08:02] <wolfgangw> hi, i'm looking into a framerate issue (apparently). i think i'm telling ffmpeg the correct input framerate (24 fps) but the input stream is identified as 25 fps. see q and full output at http://pastebin.com/fwJ8jE0x
[08:04] <relaxed> wolfgangw: you only need one -r before the inputs
[08:05] <wolfgangw> just saw that now. would it make a difference though?
[08:05] <relaxed> no, I mean you only need one -r and it should go before the inputs
[08:06] <wolfgangw> one -r in total? like i don't need it for the audio?
[08:06] <relaxed> correct
[08:07] <wolfgangw> i'll check that ...
[08:21] <wolfgangw> ok, now i used -r once (ffmpeg -y -r 24 -i ...). it still identifies the image input stream as 25 fps, 25 tbr, 25 tbn, 25 tbc. is there anything else i might be missing?
[08:23] <wolfgangw> relaxed, sorry, i need to run. should have started this with a bit more time on my hands :/ will be back. thanks so far
[09:00] <vlt> Hello. Does anyone know a better way to do things like cropping and stretching videos to specific corner coordinates than exporting all frames, applying some imageMagick magic and re-convert to video?
[09:04] <vlt> I do exactly this in realtime with OpenGL and the software "lpmt" but I need a way to script and render this to a new video file.
[09:07] <vlt> Does anyone know whether I can re-use what the video chip renders (from the OpenGL commands)?
[09:12] <cbsrobot_> wolfgangw: try -framerate 24
[09:48] <viric> hm I don't see how to use '-to'
[09:48] <viric> I want to clip from 00:05:00 to 00:10:00
[09:52] <viric> but I can't say -ss 00:05:00 -to 00:10:00
[09:53] <relaxed> -ss 00:05:00 -t 00:05:00
[09:53] <relaxed> seek five minutes and encode for 5 minutes
[09:58] <viric> relaxed: but -to exists. Why is it for?
[09:58] <viric> I'd like something that helped me avoid the subtraction of "duration=end-start" :)
[10:04] <relaxed> hmm, that's new to me.
[10:04] <relaxed> So -t works but -to dos not? Let me see your command.
[10:05] <relaxed> does*
[10:14] <theo_> After installed on Ubuntu Server resulted in an error http://pastebin.com/28ch0c4v . Trying to do mp3 to aac he. What have I done wrong?
[10:17] <relaxed> theo_: maybe add -f mp4
[10:21] <relaxed> "[libfdk_aac @ 0x20cf060] Unable to initialize the encoder: SBR library initialization error" can't be good
[10:24] <theo_> relaxed: It didn't work with -f mp4. So I must have done something wrong in my ffmpeg installation?
[10:25] <relaxed> Perhaps, compile again and install yasm beforehand.
[10:27] <theo_> relaxed: okey, I'll try that. =) thanks
[10:46] <viric> how can I know the meaning of '-q' parameter for mpeg2 encoding?
[10:47] <viric> -target film-dvd
[11:09] <viric> relaxed: my command: ffmpeg -ss 00:05:00 -to 00:10:00 -i blabla.avi ....
[11:09] <viric> relaxed: and it encodes beyond 5 minutes
[11:12] <relaxed> viric: -to needs to be after the input
[11:12] <viric> ah
[11:12] <viric> I will try
[14:34] <loller> hello! Could anyone explain me how to show ffmpeg wheer x264 is placed? on Windows
[14:35] <loller> I always get an erro "libx264 not found"
[14:36] <loller> --extra-cflags is not helpful
[14:47] <GoaLitiuM> do you have --enable-libx264 flag on the build itself?
[15:51] <loller> helooooooooo!
[15:51] <loller> so many people and silence?
[15:51] <spaam> everyone is dead
[15:53] <loller> :(
[15:59] <spaam> loller: anything you want help with ?
[16:02] <loller> spaam: actually I'mtrying to compile ffmpeg with x264 support, but any time with all the options I get "libx264 not found" ...
[16:03] <loller> so question is how to show ffmpeg the placement of x264?
[16:12] <spaam> loller: i guess you forgot to install the -dev or -devel packet of it
[16:13] <klaxa> oh... right um loller, i think i forgot something yesterday
[16:14] <klaxa> you need --extra-ldflags with the path to the shared library and --extra-cflags with the path to the headers to inlcude like this: --extra-cflags=-I/path/to/libx264-headers/
[16:15] <klaxa> you also never told me you were on windows, maybe you need to specify paths in windows style
[16:15] <klaxa> i.e. --extra-cflags=-IC:\path\to\somewhere\
[16:19] <loller> klaxa: well I tried all the flags, all the options, but x264 still not found, tht's why I'm thinking that I compiled x264 incorrectly and didn't get needed libs? what do you think?
[16:21] <klaxa> i don't know, maybe post directory listings of your x264 compilation folder on a pastebin-like site, also your config.log file in the ffmpeg folder
[16:23] <loller> klaxa: Thank you for your attention. I will
[16:26] <loller> klaxa: http://pastebin.com/9C9tzeE4 - listing
[16:27] <loller> klaxa: http://pastebin.com/SNj9233m - config.log
[16:29] <klaxa> you need too add --extra-ldflags=-L/c/x264/
[16:29] <klaxa> you need both, --extra-cflags and --extra-ldflags
[16:30] <loller> let my try
[16:30] <klaxa> but the config.log looks good, it seems to find x264.h
[16:44] <loller> klaxa: thank you so much! configure part was finished successfully)) make is running now
[16:50] <klaxa> nice
[17:03] <loller> klaxa: thank you again)) seems on this my level I did what I wanted just ddue to your help
[17:04] <loller> klaxa: but one more question please
[17:04] <loller> when I run my app I receive many messages that one dll is missing, then another and so on ... how to avoid this?
[17:05] <loller> first of all libz-1.dll which is part of MinGW ...
[17:29] <ideea> hi guys :D
[17:29] <ideea> The process has been signaled with signal "5".
[17:29] <ideea> 500 Internal Server Error - RuntimeException
[17:30] <ideea> help pls, what is it ? i cant google it ;[
[17:30] <spaam> what?
[17:30] <ideea> '/usr/local/bin/ffmpeg -i %s -vcodec copy -acodec copy -ss 00:00:00 -t 00:05:00 %s
[17:30] <ideea> when i run this command from command line - its fine
[17:30] <ideea> wen i trying to run it from code i receive "signal 5"
[17:31] <ideea> and, ofcourse, no movie converted.
[17:33] <ideea> guys?:{
[17:40] <ideea> spaam ?:D
[17:40] <spaam> ideea: woho?
[17:40] <spaam> ideea: i have no idea have you got that from that line.
[17:41] <ideea> its error code ;[
[17:41] <spaam> from what? :D
[17:41] <spaam> webpage?
[17:43] <spaam> maybe you can get the error in some logs from the webserver.
[17:43] <spaam> error.log or something
[17:43] <spaam> depending on your webserver.
[18:07] <wolfgangw> cbsrobot_: specifying -framerate 24 does indeed yield different (and correct) reporting on the image input stream (which is an image sequence). with it it reports 24 fps/tbr/tbn/tbc. thanks. you're not a bot, are ya?
[18:50] <Mista_D> 720x480 [SAR 40:33 DAR 20:11] --- what would be aspect ratio for this file if I switch to square pix?
[18:51] <JEEBsv> >>> (720 * 40) / 33
[18:51] <JEEBsv> 872.7272727272727
[18:52] <JEEBsv> so 873x480
[18:52] <Mista_D> what about DAR? Just ignore it?
[18:52] <JEEBsv> it's the same information
[18:52] <JEEBsv> it gives you it as SAR and DAR
[18:53] <JEEBsv> SAR being the aspect ratio of one sample, and DAR being the aspect ratio of the whole final picture
[18:53] <Mista_D> 720*20/11 = 1309
[18:53] <JEEBsv> that's because it's DAR
[18:54] <JEEBsv> >>> (480 * 20) / 11
[18:54] <JEEBsv> 872.7272727272727
[18:54] <Mista_D> Got it, thanks JEEB.
[18:59] <hans_henrik> >>> 8/0
[18:59] <hans_henrik> >>> 8/1
[18:59] <hans_henrik> hmm
[20:08] <oldfriend> hello! can anyone explain my how to avaid additional dependences after compiling ffmpeg?
[20:09] <oldfriend> I compiled shared dlls
[20:09] <hans_henrik> compile static?
[20:10] <oldfriend> hans_henrik: do you mean --enable-static?
[20:13] <hans_henrik> oldfriend, some people on the internet said --disable-shared --enable-static --extra-libs=-static --extra-cflags=--static
[20:16] <oldfriend> hans_henrik: Thank you! trying right now
[20:25] <causasui> how can i use ffmpeg to concatenate two video streams in mkv format?
[20:26] <Keshl> Give both as input files.
[20:26] <Keshl> Just specify -i twice.
[20:26] <klaxa> <hans_henrik> oldfriend, some people on the internet said --disable-shared --enable-static --extra-libs=-static --extra-cflags=--static
[20:26] <klaxa> it's --extra-ldflags=-static
[20:28] <klaxa> obviously you will need the static libraries for all external libraries you want to use
[20:29] <oldfriend> klaxa: I see, thank you ///trying
[20:34] <llogan> causasui: http://ffmpeg.org/faq.html#How-can-I-join-video-files_003f
[20:34] <llogan> http://trac.ffmpeg.org/wiki/How%20to%20concatenate%20%28join%2C%20merge%29%…
[20:36] <llogan> Keshl: that alone will not concatenate
[20:37] <Keshl> o Éo? Whus'ya do then? .É.
[20:47] <llogan> depending on the inputs it will create an ouput using the "best" streams as described in http://ffmpeg.org/ffmpeg.html#Stream-selection
[21:05] <oldfriend> hm ... using "--extra-ldflags='-static " I get again error about x264 not found ((
[21:07] <klaxa> do you have the static library of x264?
[21:07] <klaxa> it should be called libx264.a
[21:11] <oldfriend> yes, I have this one
[21:12] <oldfriend> ./configure --prefix=ffmpeg/ --enable-shared --enable-static --extra-cflags='-static -I/c/MinGW/include' --yasmexe='C:/yasm/yasm.exe' --enable-gpl --enable-small --disable-ffmpeg --disable-ffplay --disable-ffserver --disable-doc --enable-libx264 --extra-ldflags='-static -L/c/x264' --extra-cflags='-static -I/c/x264'
[21:12] <oldfriend> this is my configuration string
[21:13] <klaxa> you don't have to add -static to --extra-cflags
[21:14] <klaxa> is x264.h in /c/MinGW/include ?
[21:14] <klaxa> wait nvm
[21:14] <klaxa> also better put all the --extra-cflags into one argument
[21:16] <oldfriend> klaxa: do you mean like this
[21:16] <oldfriend> ./configure --prefix=ffmpeg/ --enable-shared --enable-static --extra-cflags='-static -I/c/MinGW/include -I/c/x264' --yasmexe='C:/yasm/yasm.exe' --enable-gpl --enable-small --disable-ffmpeg --disable-ffplay --disable-ffserver --disable-doc --enable-libx264 --extra-ldflags='-static -L/c/x264'
[21:16] <klaxa> yeah also remove the -static from --extra-cflags since it's a linker option
[21:17] <oldfriend> hm ... but I need to exclude additional dependences ...
[21:19] <klaxa> yes, -static is still not a compiler option
[21:19] <klaxa> you specified it in --extra-ldflags which is correct
[21:20] <oldfriend> I see ... trying
[21:32] <oldfriend> strange but --extra-ldflags='-static -L/c/x264' cause error ... but --extra-ldflags=-L/c/x264 proceeds normally ... why?
[21:34] <klaxa> do you have libx264.a?
[21:34] <klaxa> in the directory C:\x264
[21:35] <oldfriend> sure
[21:35] <oldfriend> that's why I'm confused
[21:35] <klaxa> pastebin config.log
[21:37] <oldfriend> klaxa: I really appreciate that you are wasting your time on me
[21:37] <oldfriend> pasting
[21:39] <oldfriend> http://pastebin.com/ST8xz5pA
[21:41] <fazias> I got a short question (depending on answer, will return with exact commands with what to reproduce my problem)
[21:42] <fazias> so, I splitted a video by keyframes. After that I encoded those videos and lastly put them together with ffmpeg
[21:42] <klaxa> looks like you need the pthreads static library
[21:42] <klaxa> oldfriend: see: http://www.hotscripts.com/forums/c-c/35895-help-pthreads-dev-c.html
[21:43] <fazias> Why does the concatenated video have duplicated frames at the times where 2 parts have been concatenated together.
[21:44] <fazias> this problem doesnt happen if I just split videos and then join them together (although for some reason, the fps changes a little)
[21:44] <oldfriend> klaxa: thank you again ... moving in this way
[21:47] <fazias> sure, ill ask again when I got the pastebin done.
[22:21] <SirGaga> lame: output buffer too small (buffer index: 11871, free bytes: 31541)
[22:21] <SirGaga> Audio encoding failed
[22:21] <nlewis>
[22:21] <SirGaga> What gives?
[22:23] <fazias> Here's the pastebin for my problem http://pastebin.com/qt0va4cB
[22:23] <nlewis> When I run ffprobe against a h264.mp4 that was created by ffmpeg I receive the following errors:
[22:23] <nlewis> Last message repeated 1 times [h264 @ 0x23c3460] no frame! [aac @ 0x23adb20] Reserved bit set. [h264 @ 0x23d0fe0] AVC: nal size 2063587373 Last message repeated 1 times [h264 @ 0x23d0fe0] no frame! [aac @ 0x23adb20] Prediction is not allowed in AAC-LC. [h264 @ 0x23d1600] AVC: nal size 75966110 Last message repeated 1 times [h264 @ 0x23d1600] no frame! [h264 @ 0x23d1c40] AVC: nal size 1836210915 Last message repeated 1 times
[22:24] <nlewis> Any idea what no frame! means and what might have caused the error?
[22:25] <SirGaga> Full command and output: http://pastebin.com/raw.php?i=0qQAtPXW
[22:26] <klaxa> maybe too little ram?
[22:26] <LithosLaptop> why no codecs specified?
[22:27] <SirGaga> Well, I just upgraded the ffmpeg.exe binary to the latest version. Mine was 11 months old. The new binary with the same command works.
[22:28] <SirGaga> klaxa: Well, I have 20 GB RAM, so...
[22:28] <klaxa> hmm that should be enough
[22:28] <SirGaga> LithosLaptop: Because "use the same quality".
[22:28] <SirGaga> Yes, I know, I look at ffmpeg as a magic black box.
[22:28] <LithosLaptop> :)
[22:28] <SirGaga> I don't fully understand it.
[22:29] <LithosLaptop> have you tried specifying a bitrate instead for the lame encoder?
[22:29] <LithosLaptop> it seems to be the one failing
[22:32] <LithosLaptop> ffmpeg.exe -i test.flv -c:a libmp3lame -b:a 128k -c:v mpeg4 -vtag xvid -b:v 600k test.flv.avi
[22:33] <LithosLaptop> try that
[22:34] <SirGaga> Huh?
[22:34] <SirGaga> Didn't you read when I said that it works?
[22:34] <SirGaga> "I just upgraded the ffmpeg.exe binary to the latest version. Mine was 11 months old. The new binary with the same command works."
[22:34] <SirGaga> Maybe it is some kind of bleeding-edge file.
[22:34] <SirGaga> After all, I just downloaded it as a .flv from YouTube.
[22:35] <SirGaga> The sewer of all sorts of videos.
[22:36] <LithosLaptop> oh ok, I was just afraid that it was also using the same settings as shown in the previous output. It was using less bitrate for the output even when going from x264 -> mpeg4
[22:38] <llogan> 11d695d is from Oct 2012
[22:46] <SirGaga> LithosLaptop: I'm afraid of asking this, but how can it possibly do that? It should use the *same* quality.
[22:46] Action: SirGaga fears this will spawn another multi-hour conversation of frustration.
[22:48] <klaxa> there isn't really *same* quality unless we are talking about lossless encoding
[22:48] <klaxa> just use -b:a 320k
[22:48] <klaxa> should be good enough, no?
[22:49] <SirGaga> 1:1 should be good enough.
[22:49] <klaxa> what is 1:1 ?
[22:49] <klaxa> you mean stream copy?
[22:49] <klaxa> then use -c:a copy
[22:49] <SirGaga> Identical.
[22:49] <LithosLaptop> cant be identical
[22:49] <LithosLaptop> it is lossy codecs
[22:50] <SirGaga> I feel the stress raising...
[22:50] <SirGaga> I must leave this channel.
[22:50] <klaxa> wtf
[22:50] <LithosLaptop> lol
[22:52] <LithosLaptop> yeah can be identical if he just copied instead of re-encoded
[22:52] <klaxa> what's the point of re-encoding youtube videos anyway?
[22:52] <LithosLaptop> yeah lol
[22:52] <klaxa> they already are poorly encoded, why degrade them even further?
[22:53] <klaxa> and if he says he doesn't understand ffmpeg at all, why does he not listen to people?
[22:55] <LithosLaptop> lol, don't know
[23:12] <fazias> so anybody looking into my problem? just wondering as it's quite quiet here.
[23:13] <klaxa> dunno, tbh i haven't run into this problem myself, but i don't use ffmpeg for splitting/joining
[23:14] <klaxa> since i primarily use the matroska container for everything, mkvtoolnix is doing a pretty good job
[23:14] <klaxa> even after re-encoding the separate parts with ffmpeg or x264 doesn't seem to show this behavior
[23:14] <klaxa> after joining them with mkvtoolnix again that is
[23:16] <fazias> I see.
[23:20] <fazias> Well I'll be idling here if there are any questions about my command lines.
[23:21] <fazias> Reason for using ffmpeg was simply because I felt it was easy enough to build having necessary libraries statically linked as I'm using the build on another server to which I have no administrator access.
[23:22] <viric> can anyone suggest a guide to produce a DVD video?
[23:22] <viric> I used -target pal-dvd... tried with dvdauthor -t with each mpg, then -T finally. And -T always produces an error.
[23:22] <viric> is -T supposed to work as an automatic creation of menu?
[23:23] <klaxa> you can create static build of the necessary mkvtoolnix binaries too
[23:23] <klaxa> *builds
[23:23] <viric> moreover, how to make a DVD without menu? I don't need a menu.
[23:24] <fazias> I was afraid you would answer that. That's not the point.
[23:24] <klaxa> ok
[23:25] <fazias> But thank you for informing me about it. I will keep it in mind, I would still prefer to have answer to this problem.
[23:48] <nlewis> Anyone here use ffmpeg/ffprobe to test for problematic h264 files? I've been using ffprobe -show_frames but I wondering if there is another way to approach identifying problematic files?
[23:54] <cbsrobot_> fazias: how do you split the file ?
[00:00] --- Fri Sep 13 2013
1
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[01:25] <cone-545> ffmpeg.git 03Vignesh Venkatasubramanian 07master:ae12d655384a: lavcodec: Adding support for End Trimming in Opus encoder
[05:04] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:7bc9fb8cde2b: avcodec/utils: add 16bit planar YUV formats to avcodec_align_dimensions2()
[05:04] <cone-859> ffmpeg.git 03Michael Niedermayer 07master:64b6279d1414: avcodec/mjpegdec: Support >8bpc color mjpeg
[11:37] <j-b> 'morning
[11:38] <Compn> so early
[11:39] <Compn> why'd you wake me up j-b ? :P
[11:41] <av500> j-b: im in paris tonight, can you deliver trocadero to my hotel?
[11:56] <j-b> av500: no. but beers at my place
[11:56] <av500> wo ist das?
[12:07] <j-b> myplace.com
[12:08] <av500> ill be there
[12:08] <Daemon404> no habbo hotel?
[12:09] <av500> i haz hotel by the lake
[12:09] <av500> I hope its not a toxic waste dump
[12:11] <Compn> j-b : does voo have enough space to host some dcp samples ?
[12:12] <Daemon404> av500, it's paris...
[12:12] <Daemon404> everywhere is some form of toxic
[12:12] <Daemon404> due ot cigarettes
[12:12] <Compn> i think buxiness mentioned having some he had to upload...
[12:18] <Compn> ehe
[12:18] <Compn> ubitux : people want to do chromakey/ greenscreen videos with ffmpeg (e.g. person in front of green screen replaces green with video)
[12:19] <ubitux> why are you telling me this?
[12:19] <ubitux> don't we have something for this?
[12:19] <ubitux> i think we discussed that a while ago
[12:20] <Compn> i couldnt remember
[12:21] <Compn> i guess i should be asking saste ?
[12:23] <saste> Compn, not sure it is possible, not without coding
[12:23] <saste> first we need some code to detect pixels in a given chroma interval, and create a mask from that
[12:23] <saste> then you can use overlay
[12:24] <saste> in theory geq could be used for the first stage, but i'm not sure it's so easy
[12:24] <durandal_1707> for first stage of what?
[12:25] <saste> durandal_1707, check which pixels are in a given chromatic interval or region
[12:26] <saste> HSV could be used (which reminds me of the fishdetect filter)
[12:27] <durandal_1707> and you need that for?
[12:28] <saste> durandal_1707, then you extract a mask, which corresponds to the region in the image corresponding to the given "background" color
[12:29] <saste> now you can use the mask to do an overlay, so you create a custom background which is overlaid on the background region of the original video
[12:30] <durandal_1707> i did it already in colorkeymask filter
[12:30] <ubitux> ah! see!
[12:30] <ubitux> i knew we discussed this and could do something
[12:31] <ubitux> durandal_1707: you didn't push it yet?
[12:31] <durandal_1707> its simple and crude, little better than internal avs filter
[12:31] <cone-744> ffmpeg.git 03Alexandra Khirnova 07master:f369b9356c46: avformat: Use av_reallocp_array() where suitable
[12:31] <cone-744> ffmpeg.git 03Michael Niedermayer 07master:cbe47b1e8452: Merge commit 'f369b9356c4606cd4d713d60f7db5de119d901fa'
[12:31] <cone-744> ffmpeg.git 03Michael Niedermayer 07master:32805f852116: avformat/matroskadec: ebml_parse_elem() should not deallocate priorly existing lists on failure
[12:31] <cone-744> ffmpeg.git 03Michael Niedermayer 07master:c9367c07091b: avformat/matroskaenc: functions that add entries should not destroy the whole list on failure
[12:31] <cone-744> ffmpeg.git 03Michael Niedermayer 07master:5ca39a0b899a: avformat/mxfdec: functions that add entries should not destroy the whole list on failure
[12:31] <cone-744> ffmpeg.git 03Michael Niedermayer 07master:8c9d6ee4f2ad: avformat/utils: functions that add entries should not destroy the whole list on failure
[12:32] <ubitux> durandal_1707: how dare you insult avs
[12:32] <durandal_1707> well there appears to be more poverful external avs filters
[12:32] <durandal_1707> but i doubt you will find what is used by pro
[12:32] <ubitux> ah i'm relieved then
[12:33] <durandal_1707> but we are really missing one such filter....
[12:34] <ubitux> yep
[12:35] <durandal_1707> i couldn't find how to correctly handle borders/transition
[12:35] <durandal_1707> you can read wikipedia entry for start..
[12:36] <durandal_1707> and find/make high-quality video with blue screen as background...
[12:40] <cone-744> ffmpeg.git 03Diego Biurrun 07master:c58f4069e1d5: fate.sh: Run git-clone quietly
[12:41] <cone-744> ffmpeg.git 03Michael Niedermayer 07master:24da9eb25d63: Merge commit 'c58f4069e1d5b5804c669b691510e1b8fabb67fc'
[12:43] <ubitux> any idea what motivated the quite clone?
[12:43] <ubitux> i liked having the status..
[12:49] <michaelni> ubitux, no idea, and no objections to revert as far as iam concerned
[12:49] <ubitux> well it doesn't really matter
[12:49] <cone-744> ffmpeg.git 03Diego Biurrun 07master:ec17d1aa8b83: doc: Drop VDPAU from list of supported codecs
[12:50] <cone-744> ffmpeg.git 03Michael Niedermayer 07master:c45e9097d839: Merge commit 'ec17d1aa8b833d49d8fc479e44e9e22858aa486c'
[12:50] <ubitux> i just like to see in my screen when there is an update of source files between instances run
[12:51] <Daemon404> why sit and watch automated fate runs?
[12:53] <ubitux> i happened to watch the status sometimes
[13:02] <cone-744> ffmpeg.git 03Diego Biurrun 07master:a0b901a348e4: Drop pointless directory name prefixes from #includes in the current dir
[13:02] <cone-744> ffmpeg.git 03Michael Niedermayer 07master:803445e02c77: Merge remote-tracking branch 'qatar/master'
[13:13] <cone-744> ffmpeg.git 03Paul B Mahol 07master:00e1bf8a587e: avformat/flic: check for memory allocation failures
[13:13] <cone-744> ffmpeg.git 03Paul B Mahol 07master:60caa3d8bc11: avformat/sierravmd: check for memory allocation failures
[13:13] <cone-744> ffmpeg.git 03Paul B Mahol 07master:f2e614e7e9cb: avformat/ffmetadec: return meaningful error codes
[13:13] <cone-744> ffmpeg.git 03Paul B Mahol 07master:848ad8db2208: avformat/libnut: check avformat_new_stream() return value
[13:13] <cone-744> ffmpeg.git 03Paul B Mahol 07master:9e505485f335: avformat/smacker: check avformat_new_stream() return value
[13:29] <cone-744> ffmpeg.git 03Michael Niedermayer 07master:a6ca08f1af31: avformat/gxfenc: Check and propagate return code from gxf_write_map_packet()
[13:29] <cone-744> ffmpeg.git 03Michael Niedermayer 07master:2832a82e17e9: avformat/gxfenc: Clear nb_fields on array deallocation
[13:29] <cone-744> ffmpeg.git 03Michael Niedermayer 07master:2175f7689409: avformat/asfenc: use av_freep() for extra saftey
[13:29] <cone-744> ffmpeg.git 03Michael Niedermayer 07master:b5b34c198373: avformat/matroskaenc: use av_freep() to avoid leaving stale pointers
[13:49] <cone-744> ffmpeg.git 03Michael Niedermayer 07master:e5290823697d: avformat/matroskadec: use av_realloc_array()
[13:49] <cone-744> ffmpeg.git 03Michael Niedermayer 07master:9bca8e534150: avformat/matroskaenc: use av_realloc_array()
[13:49] <cone-744> ffmpeg.git 03Michael Niedermayer 07master:c5714097eb4c: avformat/mxfdec: use av_realloc_array()
[13:49] <cone-744> ffmpeg.git 03Michael Niedermayer 07master:2a1c24a267e9: avformat/utils: use av_realloc_array()
[13:52] <michaelni> ubitux, mateo`, does one of you have time to look at todays mov.c changes ? i suspect some of it is not correct/ideal
[13:53] <ubitux> i'm not sure i'll have time today
[13:54] <michaelni> ubitux, ok, mateo` ?
[13:54] <michaelni> also doesnt need to be today ...
[13:58] <michaelni> if noone has time then ill look at it
[13:58] Action: michaelni hopes he doesnt forget
[14:01] <ubitux> michaelni: are you refering to the realloc thing?
[14:01] <michaelni> ubitux, yes
[14:02] <ubitux> you want to know if the empty list won't make everything explode?
[14:02] <michaelni> not only
[14:04] <michaelni> first question, is it a good idea? killing the lists is not very friendly if one wants to continue to use the demuxer
[14:04] <michaelni> and then of course yes, if its kept, all indexes that point into the arrays need to be reset too
[14:06] <ubitux> there are some cross references between the samples lists
[14:06] <ubitux> it should not be made inconsistent IMO
[14:07] <ubitux> if these realloc errors out, something nasty is going on, it might be relevant to error out
[14:07] <ubitux> but maybe you want to continue playback/decode anyway?
[14:10] <michaelni> yes
[14:12] <ubitux> i wonder why the checks above the alloc weren't removed
[14:12] <ubitux> since the reallocp_array should do them
[14:25] <ubitux> we should remove the ppa link to ffmpeg
[14:25] <ubitux> 0.10 is really too outdated
[14:32] <mateo`> where are the samples uploaded (k'!
[14:32] <mateo`> ?
[14:32] <mateo`> i'm looking for the sample related to https://trac.ffmpeg.org/ticket/2945
[14:36] <michaelni> ubitux, about 0.10, i suspect the old ubuntu versions need something soname compatible to that old version
[14:37] <michaelni> because there are heaps of binary packages linking to that soname
[14:37] <michaelni> newer ppas in addition are very needed though too
[14:38] <Compn> mateo` : on videolan server
[14:40] <durandal_1707> mateo`: reply to ticker and demand link to file (even better: also close it with needs_more_info)
[14:41] <Compn> durandal_1707 : its here > http://streams.videolan.org/incoming/Test_file_ticket_2945_support_remux_m2…
[14:41] <Compn> pm me if you need login info
[14:42] <durandal_1707> cehoyos should reupload it and provide link in ticket
[14:46] <Compn> why? the incoming ftp is accessible by all devs ?
[14:46] <Compn> i can put it on samples repo if you want ?
[14:47] <Compn> durandal_1707 : do you and cehoyos have some secret war going on ?
[14:47] <Compn> ehe
[14:48] <Daemon404> cehoyos drives me nuts too
[14:48] <Daemon404> i dont blame durandal_1707
[14:49] <Compn> he is strict on the bugreports and requiring information to reproduce an issue
[14:49] <Compn> do you want a more neutral place to report bugs ?
[14:50] <Daemon404> no he's hypocritical and uses blanket responses
[14:50] <Compn> maybe we could ask carl to go easy on ffmpeg devels like durandal_1707 and Daemon404 :)
[14:50] <Compn> ehehe
[14:50] <Daemon404> and cloes valid bugs with proper info with generic blanket statements
[14:50] <Daemon404> and doesnt follow the patch contribution rules
[14:50] <Daemon404> despite harping on others to do so
[14:50] <Daemon404> learn to use git-fucking-format-patch
[14:50] <Daemon404> :V
[14:51] <Compn> lol
[14:51] <Compn> hey i dont use that either :P
[14:51] <Daemon404> you dont submit patches
[14:52] <Daemon404> (except to vlc, where remi trolls you)
[14:53] <Compn> i dont get why vlc needs two entries for the same codec :P
[15:02] <durandal_1707> you are all overreacting
[15:42] <durandal_1707> mdsh: /win3
[15:43] <durandal_1707> sorry, now i'm really forced to ask you
[15:43] <durandal_1707> mdsh: do you plan to send any patches to ffmpeg ml?
[15:49] <mdsh> durandal_1707: I plan to send more patches to the ml, yes. Why - what have I missed...
[15:51] <durandal_1707> regarding w3fdif filter
[15:58] <mdsh> durandal_1707: when I get the time I plan to make w3fdif do >8 bit and try to increase the speed (for instance, michealni's suggestion of working on 32x32 rectangles - allowing multi-threading)
[15:58] <mdsh> Unless anyone else does it first...
[16:03] <durandal_1707> why 32x32 rectangles are needed for mt?
[16:07] <mdsh> michael said in an email about 2 weeks ago that doing smaller rectangles should be better for CPU cache hits - and implementing that will make MT easy
[16:08] <Daemon404> i havent looked at the code, but does w3fdif not have dependences on previous pixels?
[16:08] <Daemon404> or does it only need pre-intrpolation values?
[16:08] <mdsh> no, each output pixel is just a convolution of original surrounding pixels
[16:09] <Daemon404> ah ok.
[16:10] <durandal_1707> hmm, that would apply for single case too, and there are other filters that could make use of it...
[16:12] <durandal_1707> if gain is > 10% i would do it right now for transpose filter...
[16:12] <durandal_1707> i'm afraid if i do it, to find out it helps 0.1 %
[16:12] <Daemon404> our transpose filter can be made a *lot* faster as it is
[16:12] <Daemon404> without mt
[16:13] <Daemon404> it's very slow
[16:13] <Daemon404> i had a patch to add vapoursynth's transpose ASM at one point
[16:14] <mdsh> it would be fabulous if someone could create some ASM for w3fdif too - beyond my skill level
[16:14] <durandal_1707> well that would need smaller rectangles?
[16:15] <Daemon404> mdsh, btw is w3fdif implementable as a general convolution
[16:15] <Daemon404> or is it special?
[16:15] <durandal_1707> Daemon404: how much is vs transpose fast?
[16:15] <Daemon404> it was more than 2x
[16:15] <durandal_1707> compared to lavfi one (without mt)
[16:15] <Daemon404> when i tried
[16:15] <Daemon404> maybe it was 3x, cant remember
[16:15] <Daemon404> it was nontrivial
[16:16] <durandal_1707> it is in internal?
[16:16] <Daemon404> the patch?
[16:16] <Daemon404> i lost it
[16:16] <Daemon404> the asm is public
[16:21] <durandal_1707> i found asm and its lgpl
[16:22] <Daemon404> yes
[16:37] <durandal_1707> what you wanted to do with (now lost) patch?
[16:39] <Daemon404> i was going to submit it
[16:39] <Daemon404> but i forgot
[16:43] <durandal_1707> and it was fully working?
[16:55] <Daemon404> durandal_1707, mostly.
[16:57] <durandal_1707> what that means?
[16:58] <Daemon404> it had a few edge cases?
[16:58] <Daemon404> what does it matter?
[16:58] <Daemon404> the patch no longer exists
[16:58] <durandal_1707> :((
[16:58] <gnafu> "<@Daemon404> durandal_1707, mostly." <-- That's when they come out at night, right?
[17:00] <cone-744> ffmpeg.git 03Dylan Alex Simon 07master:38155b47fd56: avcodec: add const qualifier to avcodec_find_best_pix_fmt2 args
[17:03] <cone-744> ffmpeg.git 03Paul B Mahol 07master:99a283331c8e: avfilter/vf_traspose: move switch out of loop
[17:04] <cone-744> ffmpeg.git 03Paul B Mahol 07master:269845db86d9: avformat/nutdec: check for allocation failures
[17:05] <cone-744> ffmpeg.git 03Paul B Mahol 07master:a8e00cf926f3: avfilter: remove redundant use of AV_NE() macro
[17:05] <ubitux> durandal_1707: why dst is incremented but not src?
[17:05] <ubitux> not that it's a problem, but it's inconsistent :p
[17:15] <durandal_1707> libavformat/asfenc.c:872:18: warning: unsequenced modification and access to 'ret' [-Wunsequenced]
[17:20] <durandal_1707> ubitux: so what is 2.1 release's name?
[17:33] <ubitux> durandal_1707: quezac
[17:34] <ubitux> well, private french joke i believe
[17:34] <ubitux> forget this
[17:38] <mateo`> haha
[17:38] <durandal_1707> why they don't teach (y)asm in scool
[17:40] <mateo`> durandal_1707: they tends to prefer teaching web things instead ... the future you know :)
[19:01] <durandal_1707> ok i hacked something, but when i run it i only get bus error
[19:15] <durandal_1707> oh, got it, code is obviously ssse3 only
[19:20] <durandal_1707> or not :(
[19:29] <wm4> would it be possible to add the number of skipped samples to an AVFrame?
[19:33] <durandal_1707> hmm, it seems there is already some transpose code in lavc
[19:49] <durandal_1707> Daemon404: you sure that asm ever worked?
[19:58] <Daemon404> yes
[20:01] <durandal_1707> on 32bit?
[20:17] <durandal_1707> also code is little clusmsy, why it need 8 instruction to read 8 bytes....
[20:19] <durandal_1707> ah, it reads 8 bytes and not single one
[20:51] <wm4> does anyone know whether ingenientdec.c is actually supposed to handle this?: http://128.104.138.26/axis-cgi/mjpg/video.cgi
[20:52] <durandal_1707> maybe
[20:52] <durandal_1707> fetch it, do you see bunch MJPG in it?
[20:52] <wm4> no
[20:53] <wm4> it seems to be rather different
[20:53] <wm4> but who knows
[20:54] <wm4> hm I think it's just jpegs dumped as multi-part thing
[20:54] <wm4> libavcodec can actually decode the raw stream as jpg (I suppose it skips the header as garbage)
[20:55] <wm4> would probably be easy to write a libavformat demuxer for this
[20:57] <Daemon404> everyone and their grandma has a custom container with jpeg in it
[20:57] <Daemon404> :V
[21:08] <durandal_1707> s/grandma/dog
[21:10] <durandal_1707> how to move another half (like with movh)?
[21:37] <durandal_1707> simply using movhps with >= sse2 i get crash
[21:38] <durandal_1707> using just movh with mmx i do not get crash but result is half broken
[21:38] <durandal_1707> so it code does: movh [dstq], m0
[21:38] <durandal_1707> how to get other part of m0 ?
[22:26] <cone-744> ffmpeg.git 03James Almer 07master:7e4fe5162ab9: lavu/sha: Fully unroll the transform function loops
[22:26] <cone-744> ffmpeg.git 03James Almer 07master:bbcaf25d4d21: lavu/sha512: Fully unroll the transform function loops
[00:00] --- Thu Sep 12 2013
1
0
[00:35] <hans_henrik> can i use ffmpeg to check if a file is a valid jpeg/png/whatever file?
[05:33] <i336_> Hi everyone. I'm want to build FFmpeg as an output decoder for MPD. I only need to be able to play MP3/WAV/OGG/MOD(?) files so I probably don't need the other 4294967296 things FFmpeg has; what do you recommend I disable?
[05:34] <i336_> s/other \([^ ]*\)/\1 other/
[05:35] <klaxa> wow i don't speak sed that well
[05:35] <i336_> oh lol
[05:36] <klaxa> does ffmpeg even support MOD?
[05:36] Action: i336_ has no idea
[05:36] Action: i336_ will be using mikmod probbaly
[05:36] <i336_> probably*
[05:40] <klaxa> actually, if you just do ./configure && make it should be able to decode mp3 wav (pcm) and ogg (vorbis)
[05:40] <klaxa> most features are disabled by default
[05:44] <i336_> oh ok
[05:44] Action: i336_ wonders what DCT, DWT, LZP, LZO, MDCT, RDFT and FFT are
[05:44] Action: i336_ wonders what DXVA2, VAAPI, VDA and VDPAU are
[05:46] Action: i336_ will definitely be enabling --enable-libcaca :P
[05:47] <i336_> next question.
[05:47] <i336_> I downloaded an mplayer package for slackware and whenever I try to use it I get "Illegal instruction".
[05:47] <i336_> It works fine if I just want help, or provide no arguments; this happens when I try to actually *play* something (an FLV video).
[05:49] <i336_> I'm thinking this is because it uses optimizations that don't work on my CPU (an old AMD Duron).
[05:49] <klaxa> hmm yeah maybe
[05:49] <klaxa> compile from source maybe?
[05:49] <i336_> probably will be :P
[05:50] <i336_> my question is, do I have to disable optimizations that won't work on my CPU arch the same way I have to be careful not to use bad MARCH flags, or are the optimizations automatic?
[05:50] <i336_> in that, they disable if they won't work?
[05:51] <klaxa> autoconf should detect what CPU extentions are available and which are not
[05:51] <klaxa> that's pretty much the purpose of autoconf
[05:51] <i336_> ...ah :S
[05:51] <i336_> I'm compiling on one PC, but running on another
[05:52] <klaxa> oh
[05:52] <klaxa> in that case you have to pay attention
[05:52] <i336_> hmmmm
[05:52] <Plorkyeran> ffmpeg does not use autoconf
[05:52] <i336_> *this* is an AMD Duron; the laptop I'm compiling it for (a kind of "mini jukebox" type thing) is actually an AMD K6-2
[05:52] <i336_> ooh, ok
[05:53] <klaxa> wait... using ./configure is not part of autoconf?
[05:53] <Plorkyeran> no
[05:53] <Plorkyeran> it is a hand-written configure script
[05:53] <i336_> "configure" is a script name
[05:53] <klaxa> my world is breaking apart right now
[05:53] <i336_> awesome :>
[05:53] <i336_> elvis' "./configure" is auto handwritten
[05:53] <Plorkyeran> autoconf is a tool for generating configure scripts
[05:53] <klaxa> i see
[05:53] <Plorkyeran> if you're using a premade package it may have been built with runtime cpu detection disabled for some reason
[05:54] <i336_> actually it's enabled
[05:54] <i336_> I think
[05:54] <i336_> 3 lines:
[05:54] <i336_> CPU: AMD Duron(tm) (Family: 6, Model: 3, Stepping: 1)
[05:54] <i336_> CPUflags: MMX: 1 MMX2: 1 3DNow: 1 3DNow2: 1 SSE: 0 SSE2: 0
[05:54] <i336_> Compiled for x86 CPU with extensions: MMX MMX2 3DNow 3DNowEx SSE
[05:54] <i336_> ^ running mplayer w/ no args
[05:54] Action: i336_ haz idea
[05:56] <i336_> interesting
[05:56] <i336_> I ran it through strace and it crashed after loading fonts
[05:57] <i336_> renamed the font dir out and got 234589623498527642344509234058923452901209814352 "[flv @ 0x86807a0]Unsupported video codec (7)" messages
[05:57] <i336_> lol
[05:57] <klaxa> looks like it wasn't build with unfree decoders?
[05:57] <i336_> heh
[05:58] <i336_> MPlayer crashed by an 'Illegal Instruction'. It usually happens when you run it on a CPU different than the one it was compiled/optimized for. Verify this!
[05:58] <i336_> XD
[05:58] <i336_> probably an instruction issue. it's okay.
[06:36] Action: i336_ runs "bash-4.2$ ./configure --disable-dxva2 --disable-vaapi --disable-vdpau --disable-doc --disable-everything --enable-outdev=alsa --enable-outdev=oss --disable-iconv --enable-lto --enable-hardcoded-tables --enable-shared --enable-decoder={mp3,mp3adu,mp3adufloat,mp3float,aac,theora} --enable-demuxer={aac,mp3,wav,ogg} --enable-parser={aac,aac_latm,mpegaudio,vorbis} --enable-protocol={file,tcp,pipe}" and hopes it works
[06:37] <i336_> still mysteriously thinking :P
[06:37] <i336_> O.o
[06:37] <i336_> s/bash-4\.2\$ //
[06:38] Action: i336_ writes a progress bar for ./configure
[06:38] <i336_> apparently it needs one o.o
[06:39] <i336_> ooh
[06:39] <i336_> *output*
[06:40] <i336_> hm.
[06:41] <i336_> "*various CPU flags* enabled yes"
[06:41] Action: i336_ tries to figure out which ones will run and which won't
[06:41] <i336_> (on an AMD K6-2)
[08:08] <i336_> meep
[08:08] <i336_> I left ffmpeg building while I went out for a little while figuring it would take a while
[08:08] <i336_> I was wondering how broken it'd be when I got back :P
[08:09] <i336_> here's the whole scrollback of the compile (warning, 121KB): http://pastebin.com/wXzy7K0L
[08:09] <i336_> basically I'm getting a bunch of
[08:09] <i336_> `ff_bgr2UVOffset' referenced in section `.text' of /tmp/ccGFqtOq.ltrans9.ltrans.o: defined in discarded section `.text' of libswscale/x86/swscale.o (symbol from plugin)
[08:09] <i336_> hm
[08:10] Action: i336_ tries disabling the stripper
[08:14] <i336_> hrmmmmm
[08:14] <i336_> --enable-lto use link-time optimization
[08:14] Action: i336_ wonders what happens when you use that
[08:14] Action: i336_ tries removing it
[08:14] <i336_> :P
[09:12] <CoJaBo> While this bitrate and its obvious implication on my free disk space is appealing, will this cause a black hole if I leave it run?
[09:12] <CoJaBo> frame= 101 fps= 5 q=0.0 size= -0kB time=4.04 bitrate= -2.0kbits/s
[09:48] <i336_> agh
[09:48] <i336_> I keep getting compile errors
[09:48] <spaam> what kind of errors?
[09:48] <i336_> I've probably enabled some wrong combination of options
[09:48] <i336_> libavformat/libavformat.so: undefined reference to `ff_guess_image2_codec'
[09:49] <i336_> that's the entire error
[09:49] <i336_> *grabs configure string*
[09:49] <i336_> ../configure --disable-dxva2 --disable-vaapi --disable-vdpau --disable-doc --disable-everything --enable-outdev=alsa --enable-outdev=oss --disable-iconv --enable-hardcoded-tables --enable-shared --enable-decoder={mp3,mp3adu,mp3adufloat,mp3float,aac,theora} --enable-demuxer={aac,mp3,wav,ogg} --enable-parser={aac,aac_latm,mpegaudio,vorbis} --enable-protocol={file,tcp,pipe} --disable-amd3dnowext --disable-mmxext --disable-sse{,2,3,4}
[09:49] Action: i336_ probably did something wrong
[09:49] <i336_> I'm just building it (..!) as an output plugin for MPD.
[09:52] <t4nk931> hi all
[09:53] <t4nk931> does anyone have idea about the code :get_packet_send_clock() from ffserver.c
[09:53] <t4nk931> can anyone explain the code?
[09:57] <t4nk931> is ther any tolerance about rtp send packet when Audio + video in ffserver
[09:59] <t4nk931> because I see the code in ffserver.c function http_send_data() it says:
[09:59] <t4nk931> if((get_packet_send_clock(c) - get_server_clock(c)) > 0{ /* nothing to send yet: we can wait */ return 0; }
[10:00] <t4nk931> why there is no any tolerance if the frame duration is bigger than current server clock?
[10:23] <t4nk931> does anyone have idea about the time check in ffserver?
[10:24] <t4nk931> which criterion used by ffserver to decide send the data or not/
[12:33] <izemize> hy
[12:34] <izemize> i use vlc media player for stream webcam to rtmp server in h264, but im get "late buffer for mux caching" error. This vlc or ffmpeg problem? C:\Users\User\vlc>vlc -I dummy -vvv dshow:// :dshow-vdev="Logitech HD Pro Webcam C920" :dshow-size="320x240" --sout "#transcode{vcodec=h264}:standard{access=rtmp,mux=ffmpeg{mux=flv},dst=rtmp://192.168.56.10/deejay/1}" --sout-mux-caching="30"
[12:59] <Zaga> hi ppl
[12:59] <Zaga> has anyone been able to put ffserver working with ffmeg out-of-the-box?
[12:59] <Zaga> it always says to me connection reset by peer
[14:06] <IFN10EBrain> I unmerged ffmpeg to have xbmc pull it in. I hoped xbmc would set the right use flags for it to work with, but the flags I see in emerge -pv xbmc seem to be exactly the same as when ffmpeg was installed. xbmc works fine but the streaming of content is very glitchy and I think this is because of ffmpeg. Any pointers?
[14:47] <lentferj> when trying to compile current git master I get an error with xvid. This didn't happen back in July
[14:47] <lentferj> libavcodec/libxvid.c:362: error: .xvid_plugin_lumimasking_t. undeclared (first use in this function)
[14:47] <lentferj> libavcodec/libxvid.c:539: error: .masking_l. undeclared (first use in this function)
[14:47] <lentferj> libavcodec/libxvid.c:550: error: .masking_v. undeclared (first use in this function)
[15:00] <durandal_1707> lentferj: what xvid version?
[15:07] <durandal_1707> IFN10EBrain: what ffmpeg version? note that this is not xbmc channel
[15:46] <IFN10EBrain> durandal_1707: it's ffmpeg-7.0.1
[15:47] <durandal_1707> IFN10EBrain: there is no such ffmpeg (yet)
[15:47] <durandal_1707> latest is 2.0.1
[15:48] <IFN10EBrain> pardon me :) it's ffmpeg-1.0.7
[15:49] <durandal_1707> and it worked fine before?
[15:51] <IFN10EBrain> ffmpeg works fine. and xbmc too on this box with mint installed. now I have gentoo and the stream is very quirky.
[15:52] <IFN10EBrain> Are there any options that ffmpeg would need to meet with xbmc's requirements?
[15:52] <durandal_1707> dunno, better ask xbmc folks
[15:53] <IFN10EBrain> Guess so. Thanks anyhow.
[15:53] <lentferj> durandal_1707: libxvidcore4 2:1.2.2-0.1
[15:54] <lentferj> durandal_1707: this is debian squeeze with deb-multimedia.org packages
[15:58] <durandal_1707> i dunno why that happens headers should be there...
[16:41] <roger21> hey
[16:42] <roger21> if i want to swith the order of to audio stream, i just add -map 0:a:1 -map 0:a:0 ?
[16:42] <roger21> two*
[16:50] <vl4kn0> Hi, is there a way to get average luminance of a frame in libav?
[16:57] <JoeyJoeJo> I've got a quick question about compressed audio files. Say I have a 1 minute long MP3 that's just silence. Is that file the same size as one that's not silent at the same bit rate?
[16:57] <JoeyJoeJo> Sorry for being a little off topic. I just don't know of a better place to ask this sort of question
[16:58] <Mavrik> now that's a very good question O.o
[16:59] <JoeyJoeJo> My thought is that filesize = bitrate * length of file, so it shouldn't matter if the file is just silence or full of sound
[17:00] <Mavrik> JoeyJoeJo, the catch is that encoders for some formats will not use full bitrate if they have nothing to store
[17:00] <JoeyJoeJo> Ah, ok
[17:00] <Mavrik> but honestly, for mp3 I really don't know
[17:01] <JoeyJoeJo> I was just using mp3 as an example.
[17:01] <Mavrik> for most encoders the file will be noticably smaller
[18:33] <vl4kn0> Hi, where can I find api-example of libavfilter? I tried github repository but there's nothing like that.
[18:49] <Plorkyeran> ffmpeg.c is probably it
[18:49] <Plorkyeran> I don't know of any external projects which use libavfilter
[19:01] <JEEB> LAV Video from LAV Filters does actually use libavfilter, and I think mpv has a libavfilter interface?
[19:10] <theo_> I want to compile FFmpeg with libfdk_aac, but where to put the - "--enable-libfdk_aac --enable-nonfree"? I'm using FFmpeg compile guide https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide
[19:33] <spaam> theo_: configure line
[19:46] <theo_> spaam: Thanks! Just for clarification, is it the configure line when I install fdk-aac? or ffmpeg?
[19:47] <spaam> theo_: ffmpeg
[19:48] <spaam> theo_: since you want to add support for libfdk_aac in ffmpeg
[19:49] <theo_> So maybe I don't need fdk-aac? I want to convert mp3 to m4a =)
[19:54] <klaxa> m4a is not a codec though
[20:10] <theo_> klaxa: I'm always forgetting that. Mp3 to aac, isn't it?
[20:10] <klaxa> i'd think so
[20:11] <klaxa> and fdk_aac is the recommended aac encoder
[20:15] <theo_> klaxa: Do I also need libmp3lame? I never going to convert to mp3, just from it.
[20:15] <klaxa> i don't think you need libmp3lame to decode mp3
[20:16] <theo_> klaxa: That was really helpfull. thanks you!
[21:09] <christian-c> allright.. so I have followed every step in this guide, and I still get "-bash: ffmpeg: command not found" when typeing "ffmpeg" in the cmd. My server is running Debian 7.0 64-bit. Am I missing something?
[21:09] <christian-c> the guide: https://trac.ffmpeg.org/wiki/UbuntuCompilationGuide#CompilationInstallation
[21:30] <relaxed> christian-c: ~/ffmpeg_build/bin/ffmpeg
[21:31] <relaxed> wait, it should be in ~/bin- try ~/bin/ffmpeg
[22:18] <Patrickk> hi all , please help me... i try to install ffmpeg on centos 6....
[22:18] <Patrickk> and if i make ./configure --enable-version3 .......
[22:18] <Patrickk> i have this error
[22:18] <Patrickk> If you think configure made a mistake, make sure you are using the latest version from Git.
[22:19] <Patrickk> admin on?
[22:21] <spaam> what error?
[22:21] <spaam> with configure line ?
[22:21] <spaam> can you show us the whole output?
[22:23] <Patrickk> [root@youtubecrestin ffmpeg-0.9.1]# ./configure --enable-version3 --enable-shared --enable-gpl --enable-nonfree --enable-postproc --enable-libmp3lame --enable-libvorbis --enable-libxvid --enable-libfaac --enable-libtheora --enable-libx264 ERROR: libtheora not found If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user@ffmpeg.
[22:23] <spaam> Patrickk: ERROR: libtheora not found
[22:23] <spaam> Patrickk: you need to install the -devel packet for libtheora
[22:23] <Patrickk> it is installed
[22:24] <spaam> then you might have a old version
[22:24] <Patrickk> [root@youtubecrestin ffmpeg-0.9.1]# ls -l /usr/local/lib/libtheora* -rw-r--r-- 1 root root 447526 Sep 11 20:13 /usr/local/lib/libtheora.a -rwxr-xr-x 1 root root 941 Sep 11 20:13 /usr/local/lib/libtheora.la lrwxrwxrwx 1 root root 19 Sep 11 20:13 /usr/local/lib/libtheora.so -> libtheora.so.0.3.10 lrwxrwxrwx 1 root root 19 Sep 11 20:13 /usr/local/lib/libtheora.so.0 -> libtheora.so.0.3.10 -rwxr-xr-x 1 root root 353884 Sep 1
[22:25] <spaam> i have no idea have to read what version that is
[22:25] <Patrickk> libtheora?
[22:25] <spaam> yes.
[22:26] <Patrickk> libtheora.so.0.3.10
[22:26] <spaam> thats not the version number.
[22:27] <spaam> looks like libtheora 1.1.1
[22:27] <Patrickk> http://downloads.xiph.org/releases/theora/libtheora-1.1.1.tar.gz
[22:27] <spaam> maybe configure didnt find it? check config.log
[22:28] <Patrickk> bow?
[22:30] <spaam> open it ?
[22:30] <spaam> cat config.log
[22:30] <spaam> and look and the bottom and see if you find somthing about theora
[22:32] <Patrickk> checking for Ogg... no *** Could not run Ogg test program, checking why... *** The test program failed to compile or link. See the file config.log for the *** exact error that occured. This usually means Ogg was incorrectly installed *** or that you have moved Ogg since it was installed. configure: error:
[22:35] <mittens> do you have libogg?
[22:35] <Patrickk> i reinstalled now... libogg.. i think , now work
[22:43] <Patrickk> Found yasm 1.1.99.HEAD Minimum version is yasm-1.2.0
[22:43] <Patrickk> a tutorial to install yasm 1.20 on centos ?
[22:46] <JEEB> grab the source code, run the configure with --prefix set to where you want to install it, make, make install
[22:46] <JEEB> yasm is very straightforward
[22:55] <Patrickk> 2013-09-11 20:51:04 (500 KB/s) - `yasm-1.2.0-1.el6.x86_64.rpm' saved [753752/753752] [root@youtubecrestin tmp]# rpm -ivh yasm-1.2.0-1.el6.x86_64.rpm Preparing... ########################################### [100%] 1:yasm ########################################### [100%]
[22:56] <Patrickk> and then my ERROR [root@youtubecrestin x264-snapshot-20130910-2245-stable]# ./configure --enable-shared Found yasm 1.1.99.HEAD Minimum version is yasm-1.2.0
[22:57] <Patrickk> AND UP I INSTALLED YASM 1.2.0-1
[22:57] <spaam> good
[22:58] <spaam> maybe you have an older version of yasm installed somewhere else?
[22:58] <spaam> yasm --version say ?
[23:00] <Patrickk> http://dl.fedoraproject.org/pub/epel/6/x86_64/yasm-1.2.0-1.el6.x86_64.rpm
[23:00] <Patrickk> ir is not old
[23:02] <spaam> as i said
[23:02] <spaam> you might have an old version installed already
[23:02] <Patrickk> it is not old..... i installed now from this link
[23:03] <Patrickk> http://dl.fedoraproject.org/pub/epel/6/x86_64/yasm-1.2.0-1.el6.x86_64.rpm
[23:03] <Patrickk> it is 1.2.0
[23:03] <Patrickk> not 1.9
[23:04] <Patrickk> [root@youtubecrestin tmp]# rpm -ivh yasm-1.2.0-1.el6.x86_64.rpm Preparing... ########################################### [100%] 1:yasm ########################################### [100%]
[23:05] <spaam> what does yasm --version say?
[23:05] <Patrickk> [root@youtubecrestin x264-snapshot-20130910-2245-stable]# ./configure --enable-shared Found yasm 1.1.99.HEAD Minimum version is yasm-1.2.0 If you really want to compile without asm, configure with --disable-asm.
[23:06] <spaam> what does yasm --version say?
[23:06] <Patrickk> i installed 1.20-1
[23:06] <spaam> but what does yasm --version say=
[23:08] <Patrickk> yasm 1.1.99.HEAD Compiled on Sep 11 2013.
[23:09] <spaam> there you have it
[23:10] <spaam> somewhere in your $PATH you have an old version
[23:12] <Patrickk> do you know how i can make this wihout yasm ?
[23:12] <Patrickk> cd /tmp/x264-snapshot-20120214-2245-stable/ ./configure --enable-shared make make install
[23:13] <spaam> you want yasm
[23:13] <Patrickk> i dont whant
[23:14] <spaam> if you going to encode stuff using x264 you _really_ want it. if you dont have it it will be soooo sloow
[23:15] <Patrickk> i whant to install libx264
[00:00] --- Thu Sep 12 2013
1
0
[01:53] <cone-630> ffmpeg.git 03Vignesh Venkatasubramanian 07master:bb47aa5850c8: opus/matroska: Adding support for DiscardPadding in muxer
[01:53] <cone-630> ffmpeg.git 03Vignesh Venkatasubramanian 07master:d92b1b1babe6: lavf/matroska: Adding support for Opus CodecDelay
[02:30] <cone-630> ffmpeg.git 03Carl Eugen Hoyos 07master:fe8e01361f30: Use 0x88 as stream id when muxing DTS in program streams.
[02:30] <cone-630> ffmpeg.git 03Michael Niedermayer 07master:f7003c7a3bc6: Merge remote-tracking branch 'cehoyos/master'
[10:07] <cone-545> ffmpeg.git 03Martin Storsjö 07master:a6a4596e1219: tcp: Explicitly convert a pointer to a boolean integer
[10:07] <cone-545> ffmpeg.git 03Michael Niedermayer 07master:16ae64ce244a: Merge commit 'a6a4596e12192375e1d9dffeae24d6d4c41b25c1'
[10:16] <cone-545> ffmpeg.git 03Martin Storsjö 07master:8b524ab0c43c: movenc: Add a warning message if conflicting options have been specified
[10:16] <cone-545> ffmpeg.git 03Michael Niedermayer 07master:ada9b17e7a3d: Merge commit '8b524ab0c43c1767919530abb79d3656113cdc47'
[10:46] <cone-545> ffmpeg.git 03Martin Storsjö 07master:bdf990425e2b: movenc: Simplify setting the fragmentation flag
[10:46] <cone-545> ffmpeg.git 03Michael Niedermayer 07master:46405813c3c1: Merge remote-tracking branch 'qatar/master'
[11:12] <TimNich> Some split!
[12:45] <cone-545> ffmpeg.git 03Piotr Bandurski 07master:5cfa51929fbe: avformat/riff: add 0x594a TwoCC
[12:56] <cone-545> ffmpeg.git 03Michael Bradshaw 07master:9b07d34efb12: Make avcodec_find_best_pix_fmt_of_list const-correct
[14:21] <cone-545> ffmpeg.git 03Michael Niedermayer 07master:bb23bf8fd7fb: avfilter/avfilter: fix null pointer dereference with queued ping filters
[14:21] <cone-545> ffmpeg.git 03Michael Niedermayer 07master:f8f4e48b6da0: ffmpeg: print command replies starting on column 0
[14:21] <cone-545> ffmpeg.git 03Michael Niedermayer 07master:15251073138a: ffmpeg: improve built in docs for passing commands
[14:21] <cone-545> ffmpeg.git 03Michael Niedermayer 07master:7883827cb7f3: ffmpeg: check for unsupported "queing of command on first filter supporting the specific commend"
[14:23] <ubitux> michaelni: "queue"! no one knows what the hell is "que" :p
[15:37] <fsbfsb> I'm using ffmpeg to capture video MJPEG from an ip camera, but sometimes the camera lag and timeout. I'm trying to reset/reopen/reconnect to camera without restart the application. I'm using ffmpeg api version 1.2.3. I tried to close the AVFormatContext to reopen. I tried open a new AVFormatContext. I tried reopen the AVCodec. Nothing work, how do I do this?
[15:38] <av500> can you close the AVFormatContext?
[15:38] <av500> successfully?
[15:39] <fsbfsb> no
[15:39] <fsbfsb> an error occurs
[15:40] <fsbfsb> i will change the code to regenerate the error
[15:41] <fsbfsb> to tell you the details
[17:44] <cone-545> ffmpeg.git 03Paul B Mahol 07master:942babd87f18: avcodec/vp56: use av_reallocp_array() and check if allocation failed
[18:35] <cone-545> ffmpeg.git 03Timothy Gu 07master:83165e39fee9: doc: Rename avtools-common-opts to fftools-common opts
[18:47] <cone-545> ffmpeg.git 03Paul B Mahol 07master:bf5ceeffc3c4: avfilter/vf_psnr: >8 bit planar support
[21:16] <Daemon404> michaelni, my fate stuff is back up as of tonight
[21:40] <michaelni> Daemon404, \o/
[21:41] <michaelni> BBB, ubitux, that vp9 code fails fate (missing ref files it seems) should i add any files to fatesamples ?
[21:41] <ubitux> the ref can be generated with libvpx
[21:42] <ubitux> 00:26 <@BBB> ubitux: make testdata after cloning or downloading and configuring libvpx
[21:42] <nevcairiel> vp9 is ready now? :)
[21:43] <ubitux> it is since a few weeks i think
[21:43] <ubitux> just no optim yet
[21:43] <michaelni> dunno i just was curious about it so thought id try to compile & fate it
[21:45] <michaelni> if i should review or merge something, someone has to tell me ...
[21:45] <michaelni> also back to the fate samples, should they be put on rsync fatesamples or should i wait ?
[21:46] <ubitux> i'm not sure about merging right now, i'll work on it this week end (couldn't the last one)
[21:46] <michaelni> ok then ill wait with merging until you or BBB say i should merge it
[21:50] <cone-545> ffmpeg.git 03Clément BSsch 07release/1.1:af225c41d42e: avformat/vobsub: fix seeking.
[21:50] <cone-545> ffmpeg.git 03Clément BSsch 07release/1.1:63e370b2cf86: avformat/subtitles: check lower bound for duration overlap seeking.
[21:50] <cone-545> ffmpeg.git 03Clément BSsch 07release/1.2:8cea63c48ae0: avformat/vobsub: fix seeking.
[21:50] <cone-545> ffmpeg.git 03Clément BSsch 07release/1.2:6a782e20d761: avformat/subtitles: check lower bound for duration overlap seeking.
[21:50] <cone-545> ffmpeg.git 03Clément BSsch 07release/2.0:4e0c29451b6f: avformat/vobsub: fix seeking.
[21:50] <cone-545> ffmpeg.git 03Clément BSsch 07release/2.0:283e07087772: avformat/subtitles: check lower bound for duration overlap seeking.
[21:50] <ubitux> wm4 ^
[21:50] <ubitux> hop that's good enough
[21:51] <wm4> awesome
[21:51] <ubitux> saste: so, are you going to fix the examples? :)
[23:07] <cone-545> ffmpeg.git 03Vignesh Venkatasubramanian 07master:889bc79b5f5c: lavc/utils: Add support for discarding samples from the end
[23:26] <cone-545> ffmpeg.git 03Vignesh Venkatasubramanian 07master:7b0a839b0bbd: opus/matroska: Adding support for End Trimming in demuxer/decoder
[00:00] --- Wed Sep 11 2013
1
0
[01:10] <plitter> is it possible to extract pgs subtitles from file?
[01:11] <plitter> with ffmpeg
[02:31] <causasui> hi, i have a couple flv videos i am trying to extract sections out of and then join the sections, but i am an ffmpeg noob, as you can see:
[02:31] <causasui> $ ffmpeg -vcodec avi -acodec mp3 -i LiveLeak-dot-com-87363-rosskempstlouis4.flv -ss 00:07:25 newfile1.avi
[02:31] <causasui> Unknown decoder 'avi'
[02:34] <klaxa> syntax is ffmpeg [input_options] -i <input> [output_options] <output>
[02:34] <klaxa> so put the output options you chose behind the input
[02:46] <causasui> klaxa: ah i'll try that
[02:46] <klaxa> ah also, avi is not a video codec
[02:46] <klaxa> you might just want to try -c copy instead of -vcodec avi -acodec mp3
[02:48] <causasui> $ ffmpeg -c copy -i LiveLeak-dot-com-87363-rosskempstlouis4.flv -ss 00:07:25 -vcodec 3gp -acodec mp3 newfile1.avi
[02:48] <causasui> Unknown decoder 'copy'
[02:48] <klaxa> no no no
[02:48] <causasui> :<
[02:48] <klaxa> don't put -c copy before the input
[02:48] <causasui> haha
[02:48] <klaxa> if you want to use 3gp and mp3 don't use -c copy at all
[02:49] <causasui> klaxa: okay i did this but the output file seems to have only audio and no video
[02:49] <causasui> ffmpeg -i LiveLeak-dot-com-87363-rosskempstlouis4.flv -ss 00:07:25 -c copy newfile1.flv
[02:49] <klaxa> that is indeed kinda weird
[02:49] <klaxa> can you pastebin your input + the complete output of ffmpeg?
[02:50] <causasui> klaxa: ok
[02:51] <causasui> klaxa: https://privatepaste.com/2f414ea77f
[02:51] <klaxa> it should have video
[02:51] <klaxa> what player are you using?
[02:53] <causasui> klaxa: smplayer
[02:53] <klaxa> funny, can you try ffplay?
[02:54] <klaxa> although it is probably almost the same codebase
[02:54] <causasui> klaxa: i am getting a spectrogram with audio
[02:54] <klaxa> hmm...
[02:54] <klaxa> that usually means no video haha
[02:55] <klaxa> >[flv @ 0x18df880] Stream discovered after head already parsed [flv @ 0x18df880] New data stream 0:2 at pos:21830905 and DTS:538.873s
[02:55] <klaxa> this looks suspicious
[02:56] <klaxa> you could try a different container but i doubt it'll help
[02:56] <klaxa> if all fails, re-encode :x
[02:59] <causasui> klaxa: source is already pretty much shit quality
[02:59] <causasui> filmed with a toaster etc
[02:59] <klaxa> lool
[02:59] <klaxa> well then it shouldn't make much difference if you encode to h264 with libx264
[02:59] <causasui> klaxa: would it do anything to give it an ending timestamp? i just want everything from that stamp to the end of the clip
[03:00] <klaxa> you gave it a starting timestamp
[03:00] <klaxa> it will encode to the end
[03:00] <klaxa> like i said you can try a different container
[03:00] <klaxa> just use newfile1.mkv instead of newfile.avi
[03:23] <causasui> klaxa: ok
[03:24] <causasui> klaxa: i think that worked
[03:30] <causasui> klaxa: yep, victory. thanks :)
[03:30] <klaxa> nice :)
[04:53] <Mista_D> Can I add yadif first and then add logo? -vf "movie=147x124.mov [watermark]; [in] [watermark] overlay=550:10 [out]"
[06:31] <Mista_D> -vf "movie=147x124.mov [watermark]; [in] [watermark] overlay=550:10,yadif [out]" worked \o/
[09:11] <niksmac> i am getting this error
[09:11] <niksmac> Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
[09:11] <niksmac> this is what i tried /usr/bin/ffmpeg -i '/home/onevid/public_html/sites/default/files/videos/original/D India on four Wheels.avi.avi' -vf 'scale=320:240' -strict experimental -vcodec 'h264' -qscale '12' -acodec 'vorbis' -ac '2' -pass '1' -passlogfile '/tmp/1378796929-522ec58158cf6-multipass' -y /tmp/1378796929-522ec58158b8e.mp4
[09:12] <niksmac> this is my configuration
[09:12] <niksmac> configuration: --prefix=/root/ffmpeg_build --extra-cflags=-I/root/ffmpeg_build/include --extra-ldflags=-L/root/ffmpeg_build/lib --bindir=/root/bin --extra-libs=-ldl --enable-gpl --enable-nonfree --enable-libfdk_aac --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libtheora
[09:13] <niksmac> see more here http://pastebin.com/KJsmwGE7
[09:14] <niksmac> sacarasc you there ?
[09:17] <niksmac> anybody ?
[09:19] <niksmac> any idea on this http://pastebin.com/KJsmwGE7 ?
[09:25] <niksmac> nobody here ?
[10:08] <niksmac> any idea on this http://pastebin.com/KJsmwGE7 ?
[10:17] <niksmac> saste any idea on this http://pastebin.com/KJsmwGE7 ?
[10:23] <niksmac> ods15 ubitux uau any idea on this http://pastebin.com/KJsmwGE7 ?
[10:24] <ubitux> no
[10:24] <relaxed> don't highlight random people
[10:26] <relaxed> -vcodec libx264 -acodec libvorbis
[10:27] <Mavrik-> yeah, stuffing vorbis into mp4 probably won't work.
[10:30] <niksmac> Mavrik what to do now ?
[10:34] <Mavrik> well don't use vorbis for audio if you're muxing into mp4 right?
[10:36] <niksmac> yes
[10:36] <niksmac> which is supported ?
[10:36] <Mavrik> AAC is your safest bet
[10:36] <niksmac> have you seen my configuration ? http://pastebin.com/KJsmwGE7
[10:38] <niksmac> Mavrik i tried this /usr/bin/ffmpeg -i '/home/onevid/public_html/sites/default/files/videos/original/D India on four Wheels.avi.avi' -vf 'scale=320:240' -strict experimental -vcodec 'h264' -qscale '12' -acodec 'aac' -ac '2' -pass '1' -passlogfile '/tmp/1378802292-522eda74833ed-multipass' -y /tmp/1378802292-522eda74832b5.mp4
[10:39] <niksmac> FFmpeg reports the error to be "Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height".
[10:39] <Mavrik> niksmac, the error is few lines up
[10:39] <niksmac> Mavrik means ?
[10:40] <Mavrik> it means that your actual error is few lines above what you pasted.
[10:40] <Mavrik> read that.
[10:41] <niksmac> ic here is it
[10:41] <niksmac> PHPVideoToolkit error: Execute error. It was not possible to encode "/home/onevid/public_html/sites/default/files/videos/original/D India on four Wheels.avi.avi" as FFmpeg returned an error. Note, however the error was encountered on the second pass of the encoding process and the first pass appear to go fine. The error is with the video codec of the input file. FFmpeg reports the error to be "Error while opening encoder for output stream #0:0 - maybe
[10:41] <niksmac> incorrect parameters such as bit_rate, rate, width or height".
[10:43] <niksmac> Mavrik you understands right ?
[10:43] <niksmac> if yes please let me know what to do next
[10:46] <niksmac> Mavrik any idea ?
[14:31] <t4nk413> hi
[14:32] <t4nk413> what is av_rescale_q?
[14:32] <t4nk413> ffmpeg and ffserver use a lot of av_rescale_q for pts and dts
[14:32] <t4nk413> does anyone know when I can find the document that describe av_rescale_q?
[14:33] <t4nk413> does anyone know where I can find the document that describe av_rescale_q?
[14:34] <ubitux> simple cross product
[14:36] <ubitux> rescale X (within [A;B]) to [C;D] range
[14:37] <ubitux> it's generally used to switch time base
[15:00] <fsb> hi
[15:04] <t4nk413> hi
[15:04] <t4nk413> there are several transfer between input stream and output stream
[15:37] <fsb> I'm using ffmpeg to capture video MJPEG from an ip camera, but sometimes the camera lag and timeout. I'm trying to reset/reopen/reconnect to camera without restart the application. I'm using ffmpeg api version 1.2.3. I tried to close the AVFormatContext to reopen. I tried open a new AVFormatContext. I tried reopen the AVCodec. Nothing work, how do I do this?
[15:58] <xzise> Hi, is there a shortcut for the mapping when merging to files? I got something like ffmpeg -i a.mkv -i b.mkv -map 0:0 -map 1:1 -map ALL_OTHER_STREAMS_OF_a.mkv -c copy out.mkv
[15:58] <xzise> -map 0:0 -map 1:1 -map 0 -map -0:0 will also override the first map
[16:06] <klaxa> maybe ffmpeg -i a.mkv -i b.mkv -map 0 -map 1:1 -c copy out.mkv
[16:06] <klaxa> ?
[16:07] <klaxa> otherwise, have a look at mkvmerge
[16:07] <klaxa> the mkvmerge gui (executable is named "mmg") is rather straight forward
[17:03] <dorp> Is it possible to split video streams based on I frames, accurately? If ffprobe indicates that the 3rd I frame position is at 29.958333, what value would -ss translate as the 3rd I frame? 30? 29.959? 29.958? Is there any certainty for having consistency behind this conversion?
[17:06] <Mavrik> dorp, by default ffmpeg seeks back to the first I-frame it can find for most formats
[17:06] <Mavrik> so 29.959 will cause ffmpeg to seek back to the first I-frame before that timestamp
[17:07] <Mavrik> so rounding up is your best bet
[17:07] <Mavrik> also, if you're transcoding (not just using copy as codec) you should probably use -noaccurate-seek
[17:11] <dorp> Mavrik: I see. Thanks
[17:14] <xlinkz0> use -avoid_negative_ts 1 if you ever want to concat those cuts
[17:19] <dorp> xlinkz0: Thanks for the input, I'm just reading about it
[17:27] <dorp> xlinkz0: This is a little bit weird, I've found quite a few hits for -avoid_negative_ts. It's even mentioned in the documentation- but I can't find a detailed 'description' of it?
[17:29] <xlinkz0> ffmpeg will, when cutting on a non i-frame timestamp, create videos that have the starting timestamp ( in the metadata ) different from 0
[17:29] <xlinkz0> like if i cut from second 3 to 5 and the i-frame is on second 2, the resulting cut will have a starting timestamp of 1
[17:29] <xlinkz0> this screws up the concat demuxer
[17:30] <xlinkz0> i'd go as far as say it's a bug in it because i see no reason to go by that metadata timestamp but whatever
[17:31] <dorp> xlinkz0: How about the case of: splitting a video based on I frames, and then trying to concating it- is it likely to 'rebuild' properly?
[17:32] <xlinkz0> yes but often times you have no way of telling if you are cutting on an i-frame or not
[17:32] <xlinkz0> certain video sources don't have constant i-frames like say every second
[17:33] <xlinkz0> just to be safe if you are going to concat them using ffmpeg just use the flag there's no performance penalty or anything
[17:33] <dorp> xlinkz0: What if I use ffprobe for telling the I frames timestamps, and then use it with ffmpeg -ss?
[17:33] <xlinkz0> how is that easier and more practical than adding a flag?
[17:33] <xlinkz0> you'll have to parse ffprobe every time you want to cut, it complicates your code for no reason
[17:34] <dorp> xlinkz0: I'm not referring only to the case this specific flag is supposed to fix, my question was about the big picture?
[17:34] <xlinkz0> big picture?
[17:35] <dorp> I would like to be able to split video files, based on their I frames, and then to be able to 'rebuild' them. Is it something feasible, or unlikely to go smoothly?
[17:35] <xlinkz0> it is feasible, as long as you either cut directly on i-frames or use that flag
[17:36] <dorp> I see, okay- I'll run a few tests and see what it looks like
[19:00] <_goTAN> hey
[19:01] <_goTAN> I want to convert avi to mp4 but without using the libfaac.
[19:05] <sacarasc> Try ffmpeg -i blah.avi -c:v libx264 -preset:v veryslow -crf 25 -c:a aac -strict experimental -b:a 128k output.mp4
[19:08] <_goTAN> ah
[19:08] <dlb76> hey guys
[19:08] <dlb76> [NULL @ 0000000002534f60] Unable to find a suitable output format for '123.amv'
[19:09] <dlb76> is there a way to make it work ?
[19:09] <dlb76> AMV encoder...
[19:09] <dlb76> where to leech ffmpeg.exe for win with it ?
[19:10] <_goTAN> thanks sacarasc
[19:11] <_goTAN> is there a reason why I shouldn't use 256k for the audio?
[19:11] <sacarasc> _goTAN: If you can compile yourself, you might want to look at libfdk_aac. I've heard it's pretty good, but never used it myself.
[19:11] <sacarasc> Not really, but you're probably encoding from a lossy codec with a lower bitrate...
[19:12] <sacarasc> And lossy ¶ lossy always loses something.
[19:12] <_goTAN> well yea I could but tbh I am encoding alot of different stuff and everything is working atm so I dont wanna change a running system :)
[19:12] <_goTAN> yea k
[19:12] <brontosaurusrex> what would this subtitles format be called? http://paste.debian.net/plain/38010
[19:13] <sacarasc> That looks like srt, brontosaurusrex.
[19:13] <brontosaurusrex> nope, srt doesnt use timecodes and the times are not space separated
[19:14] <sacarasc> I thought srt did use timecodes?
[19:14] <sacarasc> 1
[19:14] <sacarasc> 00:03:10,500 --> 00:00:13,000
[19:14] <sacarasc> Elephant's Dream
[19:14] <sacarasc> That's a bit of srt... It is a little different, though.
[19:15] <brontosaurusrex> yeah
[19:15] <Plorkyeran> srt uses times
[19:15] <brontosaurusrex> see the last number is in miliseconds
[19:15] <brontosaurusrex> what i have here is 25 fps timecode
[19:15] <Plorkyeran> timecodes in this context means smpte timecodes, aka frame numbers written funnily
[19:16] <brontosaurusrex> basically i'am looking for a way to get this format into premiere cs6 (can be ugly looking, just for preview purposes)
[19:20] <_goTAN> sacarasc: hm is there a way to speed up ur example you gave me? Actually speed doesnt matter to me in this case but I have to say its really really slow
[19:22] <_goTAN> well I could use a different preset
[19:22] <_goTAN> :)
[19:22] <sacarasc> _goTAN: You can choose from ultrafast/superfast/veryfast/faster/fast/medium/slow/slower/veryslow/placebo. The further one the left the faster, the further on the right the slower.
[19:23] <_goTAN> yea I remember that
[19:24] <_goTAN> well I think I leave veryslow but I just wanted to test it and now its already running 30 min for 2 min video
[19:25] <brontosaurusrex> probably why its called "veryslow"
[19:27] <_goTAN> yea lol
[19:28] <_goTAN> now that you say that... it actually makes sense
[19:41] <_goTAN> damn I cant play the converted mp4 on my Ipad
[19:41] <_goTAN> dont know why
[19:46] <CentRookie> hi all
[19:47] <CentRookie> is there a way to encode mkv with soft subs embedded directly to mp4?
[19:47] <CentRookie> without extract the subs first?
[19:47] <CentRookie> (talking about hardcoding subs into mp4)
[20:53] <brontosaurusrex> CentRookie, as far as i know mp4 is supporting something called timed text format
[20:53] <brontosaurusrex> l8r ~
[20:56] <klaxa> CentRookie: there is: ffmpeg -i input.mkv -vf ass=input.mkv [encoding settings] output.mp4
[20:57] <klaxa> softsubs are nicer than hardsubs though
[21:11] <CentRookie> oh
[21:11] <CentRookie> i didnt know that klaxa, nice XD
[21:11] <CentRookie> does this work for ssa?
[21:11] <klaxa> it should
[21:12] <CentRookie> that s amazing
[21:12] <CentRookie> no more extracting
[21:12] <CentRookie> but i need to enable ssa lib right and ffmpeg need sto be compiled to include it?
[21:13] <CentRookie> havent been able to find which is the lib for ssa
[21:13] <klaxa> you need libass yes
[21:13] <CentRookie> i got libass
[21:13] <CentRookie> will it support ssa?
[21:13] <klaxa> >libass is a portable subtitle renderer for the ASS/SSA (Advanced Substation Alpha/Substation Alpha) subtitle format. It is mostly compatible with VSFilter.
[21:13] <CentRookie> hm
[21:14] <CentRookie> if this also work swith srt then it would be a perfect day for me
[21:14] <CentRookie> -vf srt=video.mkv ?
[21:14] <klaxa> libass might be able to render srt too
[21:14] <klaxa> i'm not too sure though
[21:16] <CentRookie> i think for srt it is "subtitles"
[21:16] <CentRookie> subtitles=subtitle.srt
[21:16] <CentRookie> wonder if it accepts mkv
[21:21] <CentRookie> eating a wagner stone oven pizza
[21:22] <CentRookie> damn those germans, their frozen pizzas are so good
[21:23] <CentRookie> http://techcrunch.com/2013/09/09/cota-by-ossia-wireless-power/
[21:23] <CentRookie> awesome
[21:30] <CentRookie> oh, 10 m and not 10 feet
[21:31] <CentRookie> even in the internet age ppl still use emperical units
[21:32] <CentRookie> ups, wrong chat sorry ^^
[22:52] <causasui> CentRookie: imperial*
[22:52] <causasui> :)
[00:00] --- Wed Sep 11 2013
1
0
[00:00] <cone-377> ffmpeg.git 03Carl Eugen Hoyos 07master:43353bf32c92: libxvid: guess a good aspect when we cant store the exact one.
[00:00] <cone-377> ffmpeg.git 03Michael Niedermayer 07master:7caaa72e6248: Merge remote-tracking branch 'cehoyos/master'
[01:53] <cone-377> ffmpeg.git 03Michael Niedermayer 07master:ce228206274b: avcodec/mjpegdec: fix shift_output() with lowres
[03:20] <cone-377> ffmpeg.git 03wm4 07master:b4e1630d4d25: lavc: don't show "Invalid and inefficient vfw-avi..." warning in mpeg4 parser
[09:31] <cone-337> ffmpeg.git 03Luca Barbato 07release/1.1:0eb465f981de: nuv: check ff_rtjpeg_decode_frame_yuv420 return value
[09:31] <cone-337> ffmpeg.git 03Luca Barbato 07release/1.1:1e9e311e2107: dv: Add a guard to not overread the ppcm array
[09:31] <cone-337> ffmpeg.git 03Anton Khirnov 07release/1.1:777bc81a91a4: lavf: fix the comparison in an overflow check
[09:31] <cone-337> ffmpeg.git 03Sean McGovern 07release/1.1:007f3f416573: Prepare for 9.9 RELEASE
[09:31] <cone-337> ffmpeg.git 03Sean McGovern 07release/1.1:4d073ddac95d: Update Changelog
[09:31] <cone-337> ffmpeg.git 03Michael Niedermayer 07release/1.1:bf312714786a: Merge remote-tracking branch 'qatar/release/9' into release/1.1
[11:40] <cone-337> ffmpeg.git 03James Almer 07master:452ac2aaecf7: lavu/ripemd: Fully unroll the transform function loops
[11:40] <cone-337> ffmpeg.git 03James Almer 07master:8702a94e4992: lavu/ripemd: Add a size optimized version of the transform functions
[12:01] <cone-337> ffmpeg.git 03Rainer Hochecker 07master:7d75fb381ba7: h264: do not discard NAL_SEI when skipping frames
[12:33] <durandal_1707> its nice when you add new feature to encoder (but do not to decoder) just to find out decoder crash
[12:33] <durandal_1707> NOT
[12:35] <durandal_1707> michaelni: ff_alloc_packet2 in ffv1enc does not take 16 bit into account correctly
[12:36] <mateo`> isn't cehoyos on irc ?
[12:38] <durandal_1707> only on special occasion
[12:38] <ubitux> mateo`: he's reading the backlog offline
[12:38] <ubitux> mateo`: so you can talk to him like if he was here right now
[12:38] <ubitux> and you'll get a mail answer later ;)
[12:39] <mateo`> i just want to let him know that i'll review the mxfenc patches this evening
[12:39] <durandal_1707> like he is in 12 light hours away
[12:40] <mateo`> at first look, i suspect something fishy about the second patch
[12:40] <ubitux> you can also just assign the ticket to yourself
[12:45] <wm4> does the rtsp:// implementation do anything special that it can't use a user-provided AVIOContext? (like mplayer does)
[12:46] <wm4> I'm getting crappy playback with mplayer ffmpeg://rtsp://... but not with ffplay
[12:46] <mateo`> ubitux: i'll just review the patch when i got some time, if some extra work is needed i'll assign myself to the ticket
[13:05] <wm4> durandal_1707: how does 10 bit yuva work at all? is the alpha component shifted like the other components?
[13:06] <durandal_1707> it spans all values
[13:07] <durandal_1707> just like 8bit
[13:07] <wm4> spans all values as in?
[13:07] <durandal_1707> 0 - 2^10-1
[13:08] <wm4> ok
[13:08] <nevcairiel> he just means that alpha is the same as the other planes
[13:11] <Compn> wm4 : are you using live555 rtsp:// in ffplay ?
[13:11] <Compn> or the native one
[13:11] <Compn> same question goes to mplayer ffmpeg://rtsp://
[13:12] <wm4> I don't think ffmpeg supports live555
[13:12] <wm4> mplayer seems to pick live555 for rtsp://
[13:12] <wm4> and that works fine
[13:12] <wm4> but with ffmpeg://rtsp://... you get video corruption (most likely due to aggressive packet dropping?)
[13:13] <Compn> what url ?
[13:13] <wm4> rtsp://live10.tiesraides.lv/live/tiesraides.lv.10_2
[13:15] <Compn> i think my ip is blocked
[13:19] <durandal_1707> why you listen to that guy...
[13:46] <durandal_1707> huh, segv happens when i use av_reallocp_array
[13:48] <durandal_1707> wtf why memset is there?
[13:57] <cone-337> ffmpeg.git 03Paul B Mahol 07master:6e07bb36393f: avcodec/truemotion2: use av_reallocp_array() and check return value
[13:58] <cone-337> ffmpeg.git 03Paul B Mahol 07master:9bc59c108baf: avcodec/bmpenc: return meaningful error code
[13:58] <cone-337> ffmpeg.git 03Paul B Mahol 07master:5e66d8ac63f9: avcodec/xwdenc: use AV_LOG_ERROR in error message
[14:10] <cone-337> ffmpeg.git 03Paul B Mahol 07master:81f231b5c7c9: avcodec/asfdec: check return value of av_mallocz()
[14:28] <cone-337> ffmpeg.git 03Paul B Mahol 07master:a5615b82eb11: avcodec/eatgv: use av_reallocp_array() and check return value
[14:44] <xlinkz0_> Hi, i'm trying to set the time_base of a file i'm transcoding but i get the following error : Codec AVOption time_base () specified for output file #0 (newintro.mp4) is not an encoding option.
[15:01] <durandal_1707> xlinkz0_: again, wrong channel
[15:02] <xlinkz0_> sorry
[16:03] Action: ubitux finally understands why ffplay fails with vid.stab
[16:04] <ubitux> it's simply that seeking breaks the frame data id
[16:04] <ubitux> and so frames get a transform that don't belong to them...
[16:05] <ubitux> anyway, the deshake filter really looks like vid.stab code (and actually copyright of vid.stab dev is in the header)
[16:05] <ubitux> but the licenses mismatch
[16:05] <ubitux> can anyone comment on this?
[16:06] <ubitux> i don't like this at all, so i'll likely contact the vid.stab guy soon about this to get his opinion
[16:06] <durandal_1707> so it does nothing new functionality?
[16:08] <ubitux> well, segmentation and motion detect look different
[16:09] <ubitux> but even that, not that much
[16:09] <ubitux> OTOH it's single pass
[16:09] <durandal_1707> how different?
[16:10] <ubitux> so there is no advanced preprocessing like smoothing (which requires N pre and post differences)
[16:10] <durandal_1707> and it is GPL?
[16:10] <ubitux> vid.stab is GPL, vf deshake is LGPL
[16:10] <ubitux> the whole transformation code is almost identical
[16:10] <ubitux> the general logic is the same
[16:11] <ubitux> anyway, if the author didn't agree, the license needs to be fixed IMO
[16:11] <durandal_1707> well you can fix single typo and relicense to GPL...
[16:11] <ubitux> fix typo? huh?
[16:11] <Compn> hes saying gpl is a typo :P
[16:11] <Compn> but yes, you have to contact author :)
[16:12] <durandal_1707> iirc you are ok to relicense LGPL to GPL and fork it...
[16:12] <ubitux> i wanted to work on deshake because it had the advantage of not being gpl but well..
[16:13] <durandal_1707> but adding wrapper for our forked filters is bummer
[16:13] <ubitux> mmh?
[16:13] <ubitux> vid.stab didn't fork deshake, it was the other way around
[16:13] <durandal_1707> well you sain there is nothing serious improved over lgpl one
[16:14] <ubitux> yes, afaict there is hardly anything better in deshake, but we need to fix the license because it's too much based on vid.stab
[16:15] <ubitux> but i wanted to work on it because it was lgpl :(
[16:15] <durandal_1707> i't tottaly confused
[16:15] <durandal_1707> are you saying we should change our deshake to GPL?
[16:15] <ubitux> s/change/fix/
[16:15] <ubitux> durandal_1707: the main benefit of deshake over vid.stab is the license
[16:16] <durandal_1707> but what was first? chicken or egg?
[16:16] <ubitux> but since the code is heavily based on vid.stab, the LGPL is very dubious
[16:16] <ubitux> vid.stab was first
[16:16] <wm4> why would anyone care about gpl vs. lgpl
[16:16] <wm4> other than actual libav* users, but you don't care about them anyway
[16:16] <durandal_1707> lool
[16:17] <wm4> I mean what I'm trying to say is, why would ffmpeg.c care?
[16:18] <ubitux> i'm raising the topic because there is currently likely a license violation
[16:18] <ubitux> now about prefering lgpl about gpl, i actually needs to use the libraries
[16:18] <durandal_1707> i wonder why this was not raised by original author
[16:19] <ubitux> yes, especially when he submitted his wrapper
[16:19] <ubitux> now i need to dig the mailing list about the deshake submission
[16:28] <Compn> i think he wanted different license on it
[16:28] <Compn> iirc
[16:28] <Compn> i dont know why.
[16:34] <ubitux> https://lists.ffmpeg.org/pipermail/ffmpeg-devel/2010-April/080827.html mmh
[16:37] <durandal_1707> line 307 of lavc/avpacket.c : i think this leaks
[17:11] <durandal_1707> kierank: why you care about license violators?
[17:11] <kierank> open source trolls on linkedin
[17:11] <kierank> that troll VLC and others for not supporting pro formats yet their support is based on gpl violation
[17:12] <j-b> kierank: take a list
[17:13] <j-b> I'll take a lawyery look at IBC
[17:18] <TimNich> ffmpeg meet at IBC then ?
[17:36] <kierank> TimNich: open source meet at ibc?
[17:36] <kierank> TimNich: are you going to ibc?
[17:37] Action: kierank thought TimNich was on the invite list
[17:50] <TimNich> kierank: yes, what invite list?
[17:53] <kierank> TimNich: basically I organise an OSS meetup every year at IBC now
[17:53] <kierank> mark came last year
[17:54] <kierank> but it's on Saturday at 1800 at 4.A61a
[17:55] <kierank> there will be beer and snacks
[17:55] <TimNich> I'll put it in my calender
[17:58] <durandal_1707> where is that?
[17:58] <kierank> durandal_1707: ibc show, amsterdam
[17:58] <kierank> TimNich: is there anything you want on the agenda?
[17:58] <TimNich> give me time to think...
[17:59] <kierank> or do you want to say a few words about your organisations' use/interest in OSS
[17:59] <kierank> atm the speakers are (in no particular order) - Jonas/Rob from SVT talking about Casparcg, j-b from VideoLAN, Mike from BBC News, Lisa from Sourcefabric
[18:04] <TimNich> I can give a brief overview odf where it sits...
[18:11] <kierank> hmmm...could end up being bbc heavy
[18:11] <kierank> especially if r&d end up speaking
[18:35] <TimNich> I'm easy...
[18:37] <kierank> there will be an audience discussion session
[18:37] <kierank> which i will try to avoid from sinking into a GPL/patents discussion
[18:38] <kierank> whilst GPL/patents are important they end up clouding the reality of using OSS when the legal issues are resolved and/or ignored
[18:38] <kierank> it's like discussing DRM, the debate goes nowhere
[18:43] <mdsh> But you could discuss how to obtain a license to use ProRes ... ;-)
[18:52] <kierank> mdsh: you ask freddie mercury
[18:52] <mdsh> kierank: lol
[18:53] <kierank> but anyway there are a lot of large companies using the lavc prores encoder and decoder
[18:56] <cptspiff> anybody here happen to know a public hls+https stream?
[18:56] <mdsh> kierank: I'm possessive no companies would use codecs for which they have not purchased licensed (or found that no license fee is applicable) - but that's off topic and I'll shut up
[20:27] <cone-630> ffmpeg.git 03Michael Niedermayer 07master:9a0e20817aee: avcodec/util: Make size argument of ff_alloc_packet2() int64_t
[20:27] <cone-630> ffmpeg.git 03Michael Niedermayer 07master:904a2864bdaf: avcodec/ffv1enc: fix size used for ff_alloc_packet2()
[20:27] <cone-630> ffmpeg.git 03Michael Niedermayer 07master:3728603f1854: avcodec/ffv1enc: update buffer check for 16bps
[20:27] <cone-630> ffmpeg.git 03Michael Niedermayer 07master:77f521d9e512: avcodec/ffv1enc: check encode_line()s return code
[20:27] <cone-630> ffmpeg.git 03Michael Niedermayer 07master:8393b80b7d94: avcodec/ffv1dec: Support decoding planes as raw PCM in 1.4
[20:27] <cone-630> ffmpeg.git 03Michael Niedermayer 07master:3576b564ec4f: avcodec/ffv1enc: encode slice as raw PCM in 1.4 when the buffer is too small.
[20:51] <cone-630> ffmpeg.git 03Reimar Döffinger 07master:4ebf09c3461c: Make packed B-frame warning message more useful.
[20:51] <cone-630> ffmpeg.git 03Reimar Döffinger 07master:723cf4b29e8b: Move packed B-frames message level to info.
[22:33] <durandal_1707> michaelni: what PCM means in ffv1?
[22:34] <nevcairiel> i would assume the same as PCM in h264?
[22:34] <michaelni> yes
[22:35] <nevcairiel> it does mean pulse-code modulation, the same as the audio acronym. :p
[22:42] <durandal_1707> huh, since when removing av_log optimize anything except size of binary...
[22:49] <nevcairiel> there is a small call overhead
[22:49] <beastd> durandal_1707: depends on what optimize you mean. anyway reimar summerized the challenge quite precisely. i am not a fan of don's proposals in that case. it misses the point that some av_log calls should always be there in any case. if it would gain any significant speed advantage i would guess someone placed an av_log call at the wrong point. but that is not claimed in the mails AFAICT.
[22:59] <cone-630> ffmpeg.git 03Paul B Mahol 07master:a27227d401ad: avcodec/ffv1dec: fix format detection
[22:59] <cone-630> ffmpeg.git 03Paul B Mahol 07master:d1a16564a237: avcodec/ffv1: YUVA(444,422,420) 9, 10 and 16 bit support
[23:01] <durandal_1707> i like that there is way to disable logs, but there is some bugs hiding there
[23:04] <beastd> well is spares some space. so it might be ok. i can't see any huge advantage though. saving space could probably be done more effectively by different measures and i do not know how much space you do actually save with disabling logs after those
[23:07] <beastd> also if it is to obvious how to out compile every av_log to a certain level and someone uses ffmpeg tools build that way could get exhausting to get proper verbose debug logs.
[23:08] <ubitux> wm4: i'm gonna backport the vobsub fixes and a few other things in a moment
[23:09] <ubitux> i wonder how many branches will need that fix..
[23:09] <beastd> ubitux: did backport fix the skip line overread patch already? because i am doing that ATM
[23:09] <ubitux> oh please do
[23:09] <ubitux> i didn't start :)
[23:15] <beastd> k, continuing :)
[23:15] Action: beastd could not have accepted another duplicate work today
[23:15] <ubitux> :D
[23:16] <ubitux> what did i miss?
[23:17] <durandal_1707> beastd: another?
[23:18] <beastd> not in ffmpeg. i wanted to commit a patch in mplayer, but when i had everything ready i compile-tested and went away for making sth to eat. when i returned the compile was finished and reimar committed the patch already.
[23:18] <ubitux> :)
[23:18] <ubitux> morality: eating is bad
[23:19] <beastd> patch was trivial, but patching, testing and writing the commit already took me precious time
[23:19] <ubitux> morality2: making food is not for developers
[23:19] <beastd> yeah, i am never gonna eat again.
[23:19] <ubitux> if eating was good for health we already knew it
[23:20] <ubitux> and so far, everyone before us died, and they all ate
[23:20] <beastd> or maybe i take moral #2 and just don't cook anymore. then i could be eating japanese junk food in front of the computer all the time ;D
[23:21] <durandal_1707> better leave long hanging fruit to someone else
[23:24] <mateo`> cooking is ok and don't take that much time, cleaning all the stuff required for cooking however does
[23:24] <mateo`> that sucks :(
[23:25] <ubitux> soylent to the rescue.
[23:27] <ubitux> ( https://en.wikipedia.org/wiki/Soylent_%28food_substitute%29 )
[23:28] <durandal_1707> you do need cooking
[23:28] <mateo`> "raw chemical powders" sounds fantastic :D
[23:30] Action: michaelni feels like vomiting after reading the ingredients list
[23:30] <ubitux> :D
[23:30] <ubitux> here is an interesting feedback: http://www.fourhourworkweek.com/blog/2013/08/20/soylent/
[23:30] <ubitux> i actually really wants to try this
[23:31] <ubitux> it should be commercialized soon ;)
[23:32] <durandal_1707> got out of chair!
[23:35] <durandal_1707> *get
[23:36] Action: beastd actually enjoys cooking
[23:36] <durandal_1707> from where you get all ingredients?
[23:37] Action: beastd is also happy to own a dishwasher
[23:42] <durandal_1707> 120g of protein? but what protein, there are bunch of them....
[23:47] <ubitux> "raw"
[23:47] <ubitux> it's atomic cooking
[23:48] <ubitux> mmh actually i'm too tired to do the vobsub backports, maybe tomorrow
[23:48] <ubitux> 'night all
[23:48] <beastd> n8 ubitux
[00:00] --- Tue Sep 10 2013
1
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[00:02] <gregf> never mind, figured it out
[00:41] <Raccoon> Is there a flag or means to auto identify the audio stream type in an mp4 and specify the correct extension when using "-acodec copy"
[00:41] <Raccoon> if it's aac then tack on .aac, if mp3 then .mp3, etc
[00:44] <Max-P> Raccoon, you probably can wrap that in a shell script and use ffprobe to get it. I don't think ffmpeg plays with the file names at all unless assembling jpegs into a movie
[00:44] <Max-P> (with the %04d thing or something similar, can't remember)
[00:45] <Raccoon> Hmm. Well I intend to write it as a windows batch script tbh
[00:46] <Raccoon> is there any concise output for -i that will just give the audio stream info, so it can be easier parsed?
[00:46] <Raccoon> or perhaps it'd be better to just patch ffmpeg to do this itself
[00:47] <Raccoon> eg: ffmpeg -demux audio1 [output_name_without_extension_optional_else_use_input_name]
[00:52] <Max-P> Raccoon, Your best bet would be to read the ffprobe manual, it can print the information in multiple formats
[00:53] <Max-P> ffprobe -loglevel panic -show_streams is getting close to returning the format
[00:56] <Max-P> Writing some sort of vbscript would probably still be easier than patching ffmpeg to support this, all you need is to find the line that contains the codec name, split the string to get the codec name and if elseif else on that to generate the extension
[01:49] <Raccoon> consider i'm not familiar with VB script, that sounds tedious :)
[02:00] <vlt> Hello. If I put a DV stream into ffmpeg and let it encode to h264 in realtime, how long does it take for a player to begin playback?
[02:01] <vlt> I mean how many frames must be present?
[02:05] <vlt> Or which ffmpeg options should I use when I need the smallest possible latency?
[02:33] <klaxa> vlt: probably intra-only encoding
[02:33] <klaxa> won't be small though
[03:06] <wiak> am trying to add a sequence of 00000001 png files, but it does not work :O
[03:14] <wiak> Could find no file with path '%04d.png' and index in the range 0-4
[03:15] <wiak> that message is funny
[03:40] <wiak> wohoo got it working :P
[08:33] <vlt> klaxa: Thank you. Size is no problem. I stream via 1GB/s LAN to a Raspberry Pi. Can you explain _why_ intra-only might be better? I mean, the first frame will be sent to the player anyway, wont it?
[11:54] <JoeGibken> Hi! I keep getting this error "error: undefined symbol `x264_cabac_encode_decision_bmi2.update_queue_low'" whenever I try to compile via "make" command the x264 from a git pull
[11:54] <JoeGibken> Any ideas how to resolve this?
[12:06] <xlinkz0_> if I compile and install a new version of x264 do i have to recompile ffmpeg to use it?
[12:23] <xlinkz0_> JEEB: sorry to trouble you but do you know if i need to recompile ffmpeg every time i change something in the x264 library?
[12:24] <smj> How do I re-encode losslessly a filtered H264 video, while preserving the profile and settings of the original one?
[12:25] <smj> I need to append them together and one of them is upside down
[12:29] <JEEB> xlinkz0_, if you didn't change the ABI and you use shared libraries you should be able to only recompile libx264
[12:29] <JEEB> otherwise you will have to recompile ffmpeg as well
[12:36] <xlinkz0_> thanks
[12:45] <smj> nevermind I threw it away
[13:38] <JoeGibken> Okay, I'm getting this error now when compiling ffmpeg - http://pastebin.com/v1i0L8sH
[13:38] <JoeGibken> This is the config.log - http://pastebin.com/EX4Vmebh
[13:39] <JoeGibken> I'm just following these instructions - http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide#ffmpeg
[13:39] <JoeGibken> Any help would be much appreciated
[13:39] <JoeGibken> Thanks!
[13:40] <cbsrobot_> JoeGibken: update libvpx or remove the --enable-libvpx configure option
[13:40] <JoeGibken> cbsrobot_: I did tried that, but now there is a new error that it cannot find x264
[13:41] <JoeGibken> But I did everything from line to line as said here - http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide
[13:41] <JoeGibken> So I'm not sure if the instructions posted here - http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide#ffmpeg - is correct
[13:41] <JoeGibken> Or at the least path
[13:41] <JoeGibken> Lemme pastebin if I disable the libvpx
[13:43] <JoeGibken> *or at least the PATH indicated
[13:43] <JoeGibken> I'm not really sure I mean
[14:03] <JoeGibken> cbsrobot_: Here is the new error I get - http://pastebin.com/c0RWa1Ha
[14:04] <JoeGibken> cbsrobot_: This is the config.log of ffmpeg directory - http://pastebin.com/V1YH3csZ
[14:09] <bencc2> what codec can I user to create a loseless movie from png files?
[14:19] <viric> bencc2: h264 has a lossless part
[14:28] <JoeGibken> Any help regarding my current problem in compiling ffmpeg please?
[14:29] <mittens> JoeGibken: it says libx264 not found
[14:30] <mittens> either install it or disable it
[14:30] <JoeGibken> mittens: Yeah, but I compiled everything line by line from this tutorial - http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide
[14:30] <JoeGibken> I need x264 that's why I'm compiling ffmpeg
[14:30] <JoeGibken> It's quite odd honestly
[14:31] <xlinkz0_> anyone have any idea how to modify the time_base of a stream?
[14:31] <JEEB> JoeGibken, is /home/me/ffmpeg_build/ the prefix you installed all of the prerequisites into?
[14:31] <mittens> JoeGibken: redo the steps for libx264, you may have missed something
[14:32] <JEEB> does /home/me/ffmpeg_prefix/include include x264.h and does the lib/ contain libx264.a or libx264.so?
[14:32] <JoeGibken> JEEB: Yeah, that's what I did
[14:32] <JEEB> if no, then you forgot something and you probably want a static library so re-configure, build and install x264 with --enable-static
[14:32] <JoeGibken> mittens: I did around 4 times already, as I discussed in the #x264 channel with the help of JEEB (thanks again!)
[14:33] <JEEB> also make sure your --prefix was set and you did make install at the end
[14:33] <JEEB> that's pretty much it
[14:34] <JoeGibken> <JEEB> does /home/me/ffmpeg_prefix/include include x264.h and does the lib/ contain libx264.a or libx264.so? --> 1st question: No (crap - what the hell?!) ; 2nd question: No o_o (what the?!)
[14:35] <JoeGibken> Lemme try to redo x264 rebuilding and libvpx rebuilding - gaaah - 3 hours in and I'm still compiling T_T
[14:35] <JoeGibken> Appreciate your help cbsrobot_ , mittens and JEEB! Lemme try that again
[14:36] <sacarasc> Didn't relaxed make a script to compile ffmpeg and dependencies?
[14:39] <JoeGibken> sacarasc: I tried that but it gives me other errors I don't understand
[14:40] <JoeGibken> I think it was libvorbis error or something
[15:02] <xlinkz0_> i'm trying to set the time_base of a file i'm transcoding but i get the following error : Codec AVOption time_base () specified for output file #0 (newintro.mp4) is not an encoding option.
[15:04] <durandal_1707> explanation is clear, isn't it?
[15:04] <xlinkz0_> i'm sorry but i don't get it
[15:04] <xlinkz0_> i mean from what i understand ffmpeg doesn't allow me to reencode with a different timebase
[15:04] <xlinkz0_> is that what it means?
[15:04] <durandal_1707> not at all
[15:04] <durandal_1707> you do not set time_base with AVOption
[15:05] <durandal_1707> at least not for private encoder option
[15:05] <xlinkz0_> how do i set codec options?
[15:05] <xlinkz0_> docs say : Options may be set by specifying -option value in the FFmpeg tools, or by setting the value explicitly in the AVCodecContext options or using the libavutil/opt.h API for programmatic use.
[15:05] <durandal_1707> you can change time_base directly
[15:06] <xlinkz0_> then goes on to say time_base is an option
[15:06] <durandal_1707> avcodeccontext->...
[15:06] <xlinkz0_> i don't use the libs
[15:07] <xlinkz0_> how do i do it with ffmpeg tool?
[15:08] <durandal_1707> settb filter?
[15:09] <xlinkz0_> will try it, thanks
[15:15] <xlinkz0_> durandal_1707: i've set it to 1/90000 and the resulting file has 1/15360
[15:23] <durandal_1707> vide/audio ?
[15:28] <xlinkz0_> durandal_1707: video only
[15:28] <xlinkz0_> ffmpeg -y -i intro.mp4 -an -s 640x360 -r 30 -filter settb=1/90000 -c:v libx264 -pix_fmt yuvj420p -profile:v baseline -x264opts level=41 newintro.mp4
[15:30] <durandal_1707> and what you use to see actual time base of video in output?
[15:32] <niksmac> hi
[15:32] <niksmac> i have installed ffmpeg why it says ffmpeg version git-2013-09-09-7d75fb3 Copyright (c) 2000-2013 the FFmpeg developers
[15:33] <niksmac> ask for any more info please
[15:35] <sacarasc> Because you installed from git, so that is the git version.
[15:35] <niksmac> so how can i identify the original ffmpeg version ?
[15:35] <niksmac> sacarasc can you help me to convert webm/mp4/ogg ?
[15:36] <sacarasc> Original ffmpeg version?
[15:37] <sacarasc> And if you ask specific questions, you'll get better answers. :D
[15:37] <spaam> niksmac: the git is the latest and the greatest. only the cool kids using it
[15:37] <niksmac> how can i identify it ?
[15:37] <sacarasc> By the version number you just posted.
[15:37] <niksmac> this one ? git-2013-09-09-7d75fb3
[15:38] <sacarasc> Yes.
[15:38] <niksmac> okay
[15:38] <niksmac> can i convert files to mp4 with this ?
[15:38] <niksmac> configuration: --prefix=/root/ffmpeg_build --extra-cflags=-I/root/ffmpeg_build/include --extra-ldflags=-L/root/ffmpeg_build/lib --bindir=/root/bin --extra-libs=-ldl --enable-gpl --enable-nonfree --enable-libfdk_aac --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264
[15:39] <sacarasc> Yes, you can.
[15:40] <niksmac> but when i try keep getting errors
[15:40] <niksmac> but i am not using it from command line
[15:40] <niksmac> when i try this
[15:40] <niksmac> /usr/bin/ffmpeg -i '/home/onevid/public_html/sites/default/files/videos/original/D India on four Wheels.avi.avi' -vf 'scale=320:240' -strict experimental -vcodec 'msmpeg4v3' -qscale '19' -acodec 'aac' -b '200k' -ac '2' -ar '11025' -pass '1' -passlogfile '/tmp/1378733175-522dcc77b3456-multipass' -y /tmp/1378733175-522dcc77b3311.ogv
[15:40] <niksmac> FFmpeg returned an error. Note, however the error was encountered on the second pass of the encoding process and the first pass appear to go fine. The error is with the video codec of the input file. FFmpeg reports the error to be "Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height".
[15:40] <sacarasc> Where did you get that command line from? :\
[15:41] <niksmac> its from Drupal module
[15:41] <niksmac> is that wrong ?
[15:41] <niksmac> i can configure the preset form back end
[15:41] <niksmac> i will share a screenshot now
[15:42] <sacarasc> If you want an mp4, try something more like: ffmpeg -i input.avi -vf scale=320:240 -c:v libx264 -crf 22 -c:v libfdk_aac -ac 2 -a:b 200k -ar 11025 output.mp4
[15:42] <niksmac> http://pastebin.com/uA3GSX4q
[15:42] <niksmac> this is for ogg
[15:42] <niksmac> can you edit the file to match my ffmpeg config
[15:43] <niksmac> if yes i can import it to Drupal
[15:43] <sacarasc> Why do you want to use msmpeg4v3?
[15:44] <sacarasc> If you want an MPEG4, try just mpeg4.
[15:44] <niksmac> i don't know i have a configure screen like this
[15:44] <sacarasc> Also, you installed libfdk_aac, which is a good aac encoder, but you're not using it? :|
[15:46] <niksmac> http://s11.postimg.org/4upr25uo3/Ogg_123_Video.png
[15:46] <niksmac> i use this screen to configure codecs
[15:47] <niksmac> you want access to that site ?
[15:47] <niksmac> sacarasc you want access to that site ?
[15:50] <niksmac> sacarasc you busy or left ?
[15:50] <sacarasc> niksmac: Sorry, I am battling a rather severe headache at the moment and as such can't give my full attention.
[15:51] <niksmac> but can you help me later sometime ?
[15:52] <sacarasc> If no one else does in the meantime.
[15:52] <niksmac> okay
[15:54] <niksmac> i need help
[16:22] <xlinkz0_> durandal_1707: ffprobe -show_streams
[16:24] <durandal_1707> is there tool that can actually change that one?
[16:24] <durandal_1707> libx264 in ffmpeg uses codec time_base for frame rate thing...
[16:29] <kauwlna> i am on archlinux, how can i find out what settings ffmpeg uses when i specify "-preset slow" for libx264?
[16:30] <kauwlna> nvm, it's a x264 preset then, x264 --longhelp
[16:32] <xlinkz0_> durandal_1707: libx264 has a timebase parameter
[16:33] <durandal_1707> and that one changes timebase reported by ffprobe when you encode using x264?
[16:36] <xlinkz0_> no idea
[16:36] <xlinkz0_> wll try now
[17:35] <plitter> I'm trying to rip the subtitles from a bluray with ffmpeg -v 9 -loglevel 99 -analyzeduration 999999999 -probesize 999999999 -i File.m2ts -vn -an -scodec pgssub sub.srt this and get http://sprunge.us/hDBg I also tried with -scodec copy http://sprunge.us/XHSZ could anyone point me in the right direction?
[17:35] <plitter> wait a sec
[17:56] <xlinkz0_> durandal_1707: it does work with x264
[17:57] <durandal_1707> xlinkz0_: same container?
[17:57] <xlinkz0_> yes, mp4
[17:58] <durandal_1707> xlinkz0_: well create feature request...
[17:59] <xlinkz0_> where?
[18:00] <plitter> I'm trying to rip the subtitles from a bluray with ffmpeg -v 9 -loglevel 99 -analyzeduration 999999999 -probesize 999999999 -i File.m2ts -vn -an -scodec copy -map 0:4 sub.sub with -scodec as copy, pgssub and hdmv_pgs_subtitle. but I get http://sprunge.us/jHjY, http://sprunge.us/DCbK and http://sprunge.us/jbBO could someone point me in the right direction?
[18:03] <plitter> the first adress should be http://sprunge.us/hBdL instead... any help
[18:05] <xlinkz0_> durandal_1707: but why do i get an error when i try to just supply it to libx264 using x264opts?
[18:09] <durandal_1707> even if you provide it via x264opts why you think it will apply if that is stored outside of video bitstream
[18:09] <xlinkz0_> hmm
[18:10] <durandal_1707> try x264-params instead
[18:10] <xlinkz0_> is it though? because i get that info with ffprobe -show_streams
[18:10] <xlinkz0_> i don't get it from the container, i get it from the stream
[18:11] <durandal_1707> then giving it via x264-params should work
[18:11] <xlinkz0_> anyway i'm cleaning up the current ffmpeg and going to compile latest brb
[18:34] <xlinkz0_> now i get [libx264 @ 0x18b4d60] Error parsing option 'timebase = 1/90000'.
[18:34] <xlinkz0_> :\
[18:37] <xlinkz0_> damn durandal quit
[18:37] <xlinkz0_> anyone got any ideas?
[18:37] <xlinkz0_> the ffmpeg tool call is
[18:37] <xlinkz0_> ffmpeg -y -i intro.mp4 -an -s 640x360 -r 30 -c:v libx264 -pix_fmt yuvj420p -profile:v baseline -x264-params "timebase=1/90000:level=41" newintro.mp4
[00:00] --- Tue Sep 10 2013
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