[Ffmpeg-devel-irc] ffmpeg.log.20131014

[burek]

[01:47] <Fandekasp> hi there
[01:49] <Fandekasp> I'm trying the gource software, and cannot get a mp4 file out of it. Trying the 2-steps methods, I get a ppm file, and when I try to run the ffmpeg command, it fails with this error: http://sprunge.us/bTFC?console
[01:50] <Fandekasp> I tried slighly different commands, but all fail, and I'm don't know ffmpeg enough. Anyone has some time to help me debug this through ?
[01:54] <DrSlony> Fandekasp
[01:55] <Fandekasp> hi DrSlony
[01:57] <DrSlony> for s in "0.8" "1.0" "1.2" "1.4"; do fps="30"; vid="gource_${s}spd_${fps}fps"; bitrate="2M"; preset="slower"; gource --multi-sampling -r ${fps} -1280x720 --seconds-per-day ${s} --title "Your Program" -o - | ffmpeg -y -b:v 10000K -r ${fps} -f image2pipe -vcodec ppm -i - -vcodec libx264 -preset ultrafast -crf 12 -bf 0 -threads 0 /tmp/${vid}.mp4 && ffmpeg -y -i /tmp/${vid}.mp4 -an -vcodec libx264 -pass 1 -preset ${preset} -b:v ${bitrate} -bf 0 -threads
[01:57] <DrSlony> 0 -f rawvideo /dev/null && ffmpeg -y -i /tmp/${vid}.mp4 -an -vcodec libx264 -pass 2 -preset ${preset} -b:v ${bitrate} -bf 0 -threads 0 /tmp/${vid}_${bitrate}_${preset}_`date +%F_%H%M`.mp4 && ls -lh /tmp/${vid}*.mp4; done
[01:57] <DrSlony> that's one command, so copy and join the two lines
[01:57] <DrSlony> and tweak as you see fit
[01:59] <Fandekasp> DrSlony: why are we creating 4 different videos ?
[01:59] <DrSlony> you dont want to bore your viewers, yet you dont want to zoom through your program too fast, so tweak the seconds-per-day values in my example, which are 0.8, 1.0, 1.2
[01:59] <Fandekasp> Already set it to 0.2, project is quite huge
[01:59] <DrSlony> alrighty then :]\
[02:00] <Fandekasp> ok let me try
[02:03] <DrSlony> by the way, dont change the crf 12 or preset ultrafast - at that stage you want maximum quality and speed
[02:07] <Fandekasp> DrSlony: when running the command, I see those errors http://sprunge.us/ANYI . Nothing important ?
[02:08] <Fandekasp> Something that annoys me is that even with seconds-per-day 0.15, after 3 months, it starts becoming super slow like 30sec per day :( Nothing to do with ffmpeg though hehe
[02:13] <DrSlony> is that in the second part? then you could try moving it before the "-i /tmp/${vid}.mp4"
[02:14] <DrSlony> i dont remember which version of gource i ran it on
[02:15] <DrSlony> must have been 0.37, worked well, didnt notice any seconds-per-day issues
[02:18] <Fandekasp> yeah no /tmp/*avi file is been created
[02:19] <Fandekasp> I ran gource v0.3
[02:19] <Fandekasp> gource v0.4*
[02:21] <Fandekasp> DrSlony: error is as follow http://sprunge.us/aUHR
[02:38] <DrSlony> Fandekasp: i'd replace the && with newlines to be sure of the place where it screws up, and then run just the problematic part. Then i'd move the "-i -" to the end right before the output file (first check the documentation of the ffmpeg version youre using)
[02:39] <DrSlony> worked fine 1.5 years ago :]
[02:39] <DrSlony> good night
[10:22] <jankarlitos> When using ffmpeg for streaming and the Internet connection is lost, ffmpeg hangs forever. Is there any timeout i can set for this?
[10:25] <Apic> A splendid wonderful fine Morning (UGT) on this gorgeous Boomtime!
[11:03] <sim590> hi! I'm trying to screencast using this command ffmpeg -f x11grab -y -r $FRAMERATE -s $SIZE -i :0.0 -f alsa -i hw:0,0 -acodec flac -vcodec ffvhuff -qscale 0 ./out.mkv and it works but the audio is getting spike lags
[11:03] <sim590> the audio just cuts and uncuts
[12:08] <dol> hi all. when I call avcodec_decode_video2(c1, av_pic, &got_picture, &pkt); it fills the av_pic information and I use this information when converting from YUV to RGB. I see that 50 is added into the value of av_pic.linesize[0].
[12:08] <dol> this is the reason I see a crop in my output image
[12:09] <dol> is there any way to omit this or to know that 50 is already added?
[12:09] <dol> this happens only in some specific resolutions
[12:34] <cheri> hi how do I create the index on a mpeg2ts file
[12:48] <starfox21> hey guys I am trying to convert an mp4 to an ogv on a mac but I am getting this error -> Encoder (codec none) not found for output stream #0:0
[12:48] <starfox21> how do I install the missing codecs?
[14:21] <allengreen> can Faac encode a sample rate to another?
[14:22] <JEEB> no, you do it with a resampling filter before pushing the samples into an encoder
[14:23] <allengreen> thanks, Could you tell me more about filter?
[14:24] <JEEB> I think it's just -ar after the input? In older versions of ffmpeg it used something else, in newer ones it uses libswresample
[14:25] <JEEB> generally you don't need to change the rate, though
[14:25] <JEEB> also, if you are going to encode AAC I recommend you compile ffmpeg with fdk-aac
[14:25] <allengreen> I mean do you know any independent project of filter? ffmpeg is too big.
[14:25] <JEEB> that encoder is much better
[14:26] <JEEB> allengreen, libswresample or libavresample (both included in ffmpeg, the former enabled by default) contain the actual features
[14:26] <JEEB> and are the libraries called to do this
[14:26] <JEEB> also there are plenty of non-ffmpeg related libraries that do this stuff :P
[14:26] <JEEB> sox and friends come into mind as well
[14:28] <allengreen> understand.
[14:29] <JEEB> btw, if you are making a project that is supposed to be distributing binaries, faac will not fly
[14:30] <JEEB> it's not really GPL (since it uses code from one of the reference implementations, which is not under a license that is compatible)
[14:30] <allengreen> yes, GPL sucks,  I like BSD.
[14:31] <JEEB> well, the reference code doesn't fit either :P
[14:31] <JEEB> if you are making a project that just needs an AAC encoder, I recommend using libavcodec's, as it's better than the vo-aacenc one and LGPL. If you are making something payware then I recommend just taking a license from fraunhofer or whatever for their open source fdk-aac encoder
[14:32] <JEEB> this is for binary distribution, btw
[14:32] <JEEB> if you can just ship source code to users or in case of open source just ship source code, then you can just expect the user to build his/her own fdk-aac binary
[14:34] <allengreen> I' gonna first solve the usability problem, then solve the license problem.
[14:35] <JEEB> well, just saying
[14:35] <JEEB> since it's not fun to find out that the thing you were going to use isn't distributable as binaries :P
[14:36] <allengreen> BTW, do you known independent project about PCM resampling? I mean ffmpeg is too big for me currently.
[14:37] <JEEB> well, I said that there are those two libraries within ffmpeg that handle that, one of which you can start using. Or then there's (lib)sox and so forth :P
[14:37] <allengreen> I can use ibswresample or libavresample without ffmpeg?
[14:38] <JEEB> all the libraries are separately usable, although the libavutil helper library might be most useful to be used as well :P
[14:38] <allengreen> sounds like resampling is not a difficult algorithm, if I am familiar with audio codec, I can write my own.
[14:40] <JEEB> anyways, I don't know what you're doing but I'd generally not try to reinvent wheels :D I already noted two alternatives from within the ffmpeg's source tree (since we're at #ffmpeg after all), and even noted one separate alternative ((lib)sox)
[14:40] <JEEB> so yeah
[14:41] <allengreen> get it.
[16:38] <dharriso> Hi, having problems setting libfdk_aac codec private options on the CLI, see http://pastebin.com/Xbh9wge2
[16:39] <dharriso> Im not sure how to set the VBR quality everything else works well i.e. profile and CBR options but not VBR
[16:39] <dharriso> I followed the recommendations here; http://www.hydrogenaudio.org/forums/index.php?showtopic=95989
[16:39] <dharriso> any thoughts on what im doing wrong?
[17:01] <Mavrik> dharriso, you're probably looking for this: https://trac.ffmpeg.org/wiki/AACEncodingGuide
[17:01] <Mavrik> the parameter is "-vbr" apparently.
[17:01] <dharriso> yea
[17:03] <dharriso> ah -vbr without the flags options then
[17:07] <dharriso> thanks for the link
[17:20] <xlinkz0> is flv better for live streaming than mp4?
[17:22] <Mavrik> those are just containers
[17:23] <Mavrik> there isn't much difference except that mp4 might be better supported
[17:24] <xlinkz0> thanks
[17:24] <ubitux> mp4 was never designed for streaming
[17:25] <Mavrik> yet it's still used for it -_-
[17:25] <ubitux> it's one of the worse for streaming, even though it's used a lot for this
[17:25] <Mavrik> yep
[17:25] <Mavrik> sadly MPEG2-TS isn't as widely supported as I would liek
[17:25] <ubitux> flv is good enough
[17:25] <ubitux> for most cases
[17:26] <JEEB> <ubitux> it's one of the worse for streaming, even though it's used a lot for this <- I don't actually agree, at least with movie fragments
[17:26] <Mavrik> unless your device won't read it.
[17:26] <ubitux> JEEB: yeah, with a giant muxing factory :)
[17:26] <JEEB> too bad fragments aren't actually that supported >_>
[17:26] <JEEB> ?
[17:27] <ubitux> mp4 is complex, adding a segmenting/playlist layer on top really looks like a hack to me, but well
[17:27] <JEEB> I thought fragments were just mini-indexes, just like the main index in the moov atom and related
[17:27] <ubitux> if you want to live stream, you're forced to chunk your output
[17:27] <JEEB> huh
[17:28] <JEEB> you just pipe a movie fragment using mp4 into some libavformat-using parser/player just fine, and get a picture
[17:28] <JEEB> and then how quickly you get that depends on  how often you put the fragment-related indexes
[17:28] <JEEB> (methinks)
[17:29] <ubitux> yeah sure ok you can do that
[17:29] <JEEB> but in any case, the main problem is the support for that feature :P
[17:29] <JEEB> because while it is useful, it is still quite rarely used
[17:30] <ubitux> muxing flv is way simpler :)
[17:30] <ubitux> 13B header, then interleaved a/v packets with ts
[17:30] <ubitux> izi
[17:30] <JEEB> sure, but you have libavformat/l-smash/whatever to do most of the hard work for you in any case if you're doing the muxing from your code
[17:30] <JEEB> most people don't NIH muxing
[17:31] <ubitux> depends
[17:31] <ubitux> flv can be hand muxed without much pain
[17:31] <JEEB> but as I said, the problem here is not is it good or not, but the fact if stuff is supported or not :P
[19:01] <chrisstreeter> hi all, i'm trying to debug why an audio file (m4a) is having trouble getting converted using ffmpeg. I run ffmpeg and got the following output: http://dpaste.com/1416644/
[19:02] <chrisstreeter> is there any way to better understand what would cause an issue like this?
[19:03] <chrisstreeter> the source audio file is from a recording on an iPad. I'm also thinking that the source audio file is generated by combining two files together. my initial hunch is that the first part of the audio is recorded with single channel aac audio, and then the second half is multiple channel
[19:46] <vl4kn0> Hi, how do I create thumbnail every 1/4 of second from 00:00:00 to 00:30:00 ?
[20:38] <ka1ser> *we* are getting several messages of this kind while using ffmpeg h264 codec: [h264 @ 0x50c8ba0] no frame!..... but video looks fine, any idea of what could make these appear as errors while video seems to be fine? this is on a h264 stream being decoded through network (softphone)
[22:30] <vl4kn0> Hi, is it required to call av_frame_unref() after calling the avcodec_decode_video2 or avcodec_decode_audio4?
[22:33] <vl4kn0> The problem is that I have a program using ffmpeg decoding and it usually after 20 minutes starts to consume huge amounts of memory until it gets killed by the kernel. I suspect that either AVFrame is not deallocated or I do not read all the data from packet buffer.
[23:03] <jkli> Does anybody know how to apply denoise / deblock filter on images when creating thumbnails from a video file?
[23:05] <llogan> ffmpeg -i input -ss 12 -vframes 1 -vf <one of several avilable denoise filters> output.png
[23:08] <jkli> thanks
[23:08] <jkli> How do I auto pace thumbnails, like every 5% of the movie length?
[23:08] <jkli> is there a way?
[23:10] <llogan> you can possibly use the select filter, or maybe -r
[23:11] <llogan> http://ffmpeg.org/ffmpeg-filters.html#select_002c-aselect
[23:14] <Chat5388> Hi
[23:14] <jkli> thanks a lot llogan :)
[23:15] <MorehouseJ09> hey anyone ever gotten the error "bitrate tolerance too small for bitrate"
[23:15] <MorehouseJ09> can't find too much information on actually working with this using the libav* libs
[23:28] <Mavrik> MorehouseJ09, well I think the message it pretty clear
[23:35] <AndrzejL> hi folks
[23:35] <AndrzejL> I need something that grabs the stream inside the lan then changes the res and bitrate and  then streams it to the wan
[23:36] <AndrzejL> I hear that ffmpeg can do that - what am I looking for exactly? how can I achieve such thing?
[23:36] <Mavrik> with that little info.
[23:36] <Mavrik> magic.
[23:38] <AndrzejL> ok lets say I have a local machine that streams video and music. I would like to be able to access it from my tablem when I am not at home but I dont want to use to much bandwidth so I would like to use my server that faces wan to grab the local stream, change the bitrate and the resolution of the videos AND then stream it to wan.
[23:39] <AndrzejL> s/tablem/tablet
[23:40] <AndrzejL> is this possible to do with ffmpeg?
[23:41] <Mavrik> yes, but you will require alot of knowledge
[23:41] <Mavrik> just buy Plex
[23:41] <Mavrik> it does exactly what you want.
[23:41] <AndrzejL> is plex available for linux? :)
[23:42] <MorehouseJ09> Mavrik: yeah, I've been playing around with the tolerance but can't get anything to work :(
[23:42] <AndrzejL> I would preffer to actually gain the knowledge and do it using ffmpeg ;)
[23:43] <Mavrik> MorehouseJ09, well, with that info we can't really help you
[23:43] <Mavrik> I suggest you look under doc/examples
[23:43] <Mavrik> and step through ffmpeg code with gdb to see what triggers the error
[23:46] <MorehouseJ09> definitely. So its not looking like a trivial fix unfortunately :(
[23:47] <Mavrik> it probably is, but I forgot my crystal ball at home
[23:47] <Mavrik> it'll take you whole two minutes to step through code with a debugger ;)
[23:48] <MorehouseJ09> Mavrik: thanks. Working on it now
[00:00] --- Tue Oct 15 2013