[Ffmpeg-devel-irc] ffmpeg.log.20140813

[burek]

[00:52] <c_14> fajung: try removing `-flags +aic+mv4' from the commandline
[00:57] <fajung> is there any way to make the output the same  the same bitrate as the input? so I don't have to keep changing it -b:v xxxxk
[00:59] <c_14> no ffmpeg-internal way that I know of and if the input uses variable bitrate encoding, no good way unless the encoder writes pertinent settings into the metadata
[02:50] <kline> is there some sort of political issue with ffmpeg and debian? im seeing in various debian places notes saying that ffmpeg is deprecated, but the site shows recent blog entries
[02:52] <kline> oh gg
[02:52] <kline> cant help but think you guys got a little shafted because the package maintainer fell on the other side
[02:56] <sacarasc> IIRC, Debian is getting ffmpeg back soon.
[02:56] <c_14> If the flamewar ever ends, that is.
[05:20] <jrgill> Is there any common issue where the native AAC encoder renders excessive artifacts for stereo tracks?  The output is unusable at 256 Kbps stereo while 192 Kbps mono is fine.
[05:21] <c_14> The encoder is experimental, there might be issues. What version of FFmpeg are you running?
[05:22] <jrgill> ffmpeg version N-65366-g1c59419 Copyright (c) 2000-2014 the FFmpeg developers
[05:25] <jrgill> I'll give libvo_aacenc a try.  Working on Windows so that's my only other option right now.
[05:26] <jrgill> Will update back here soon.
[05:28] <jrgill> I checked this a while ago but not in full.  Maybe it's related.  https://trac.ffmpeg.org/ticket/2686
[05:34] <jrgill> Wow, apparently this is a big issue.  Just noticed the scrollbar, heh.
[06:09] <jrgill> Well, at least libvo_aacenc consistently sounds like crap.  Will try 192 Kbps stereo with native encoder...
[07:11] <jrgill> Does libmp3lame support -r:a 48k?
[07:14] <relaxed> jrgill: yes, see "ffmpeg -h encoder=libmp3lame"
[07:15] <jrgill> Thanks, checking now.
[07:20] <jrgill> Is any of this invalid?  -c:a libmp3lame -r:a 48k -b:a 320k -af pan="stereo|c0=c0|c1=-1*c1"
[07:20] <jrgill> AAC-LC 320 Kbps input
[07:21] <jrgill> 48k
[07:21] <relaxed> why are you encoding it?
[07:22] <jrgill> I need to invert the right channel and the included AAC encoders suck.
[07:23] <jrgill> Other times it will be PCM, so just this one time re-encoding.
[07:24] <relaxed> you probably want single quotes around the whole filter
[07:34] <jrgill> I'm not seeing options for CBR and VBR.  http://privatepaste.com/download/59de9f1057
[07:35] <relaxed> -q:a 0 would be libmp3lame's highest quality vbr
[07:35] <relaxed> cbr would be -b:a $value
[07:37] <jrgill> Well, let's see what happens with -q:a 0.
[07:45] <towski> I think the recent debian ffmpeg changes broke a script I run
[07:46] <towski> I maybe figured out some of the changes, but I still get
[07:46] <towski> Cannot find a matching stream for unlabeled input pad 1 on filter Parsed_overlay_0
[07:46] <towski> this is the command I'm running
[07:46] <towski> http://pastebin.com/xPE4ggru
[07:47] <towski> also please let me know if I need to be asking in avconv, I'm not really clear on the different
[07:54] <relaxed> towski: what is that filter supposed to be doing?
[07:55] <relaxed> -q:v 3 does nothing there
[07:59] <towski> relaxed: it used to be -qscale 3
[07:59] <towski> but that threw an error that it was ambiguous
[08:00] <towski> here is the command that used to work: http://pastebin.com/Yn0x7tqz
[08:00] <towski> I changed -f pulse since that didn't seem to work anymore to -f alsa -i pule
[08:00] <towski> pulse*
[08:00] <relaxed> towski: ask in #libav
[08:03] <towski> relaxed: thanks. do you have any info on what changed recently in debian testing? I saw an email about a big change and assume that's what broke this
[08:03] <towski> it could just be a newer version
[09:36] <jrgill> I guess M4V can't contain MP3 audio.  Figures...
[10:02] <polysics> hello! cna I use the concat filter so that I can concatenate portions of different files?
[10:03] <polysics> example: I have two 10 second videos. I want the full first video and 5 seconds of the second, concatenated
[10:03] <polysics> I was hoping that could be done in one command
[10:08] <termos> How can I set the volume per channel using ffmpeg audio filters? I looked at the volume filter and it doesn't seem to be doing exactly this
[10:25] <polysics> the concat demuxer is also behaving weirdly: it stops the stream at the longest clip, ie. if I use a 5sec clip and a 10sec clip, it plays the full first clip and 5 seconds of the second
[10:27] <polysics> I also think I am using the duration directive in there wrong - it does not truncate files, does it?
[10:56] <polysics> hmm
[10:57] <polysics> https://gist.github.com/polysics/31ad6235f1219cdfb7a1
[10:57] <polysics> gets me
[10:57] <polysics> [Parsed_trim_1 @ 0x7fb959500560] Media type mismatch between the 'Parsed_trim_1' filter output pad 0 (video) and the 'Parsed_concat_3' filter input pad 1 (audio)
[10:57] <polysics> [AVFilterGraph @ 0x7fb95942c1e0] Cannot create the link trim:0 -> concat:1
[10:58] <polysics> I am not sure what I am doing wrong, trying to add audio to those files and join them
[11:03] <polysics> or maybe concat does not do what I think it does? :)
[11:23] <polysics> shouldn't this work now? https://gist.github.com/polysics/37582ec1c7707fc3b448
[11:30] <polysics> oh, Atrim
[11:30] <polysics> great, now one last thing: I would like to encode this in very high quality h264
[11:30] <polysics> any recommended settings?
[11:32] <DopeLabs> -c:v libx264
[11:32] <relaxed> polysics: https://trac.ffmpeg.org/wiki/Encode/H.264
[11:33] <polysics> resulting files look like they are a bit too big- but maybe I could encode to 720p or even 480p
[11:33] <polysics> 1080p is large no matter what, I assume
[11:34] <DopeLabs> depends on the bitrate, fps, etc
[11:34] <sfan5> I'd suggest keeping the input res.
[11:35] <sfan5> Higher res. is usually better
[11:37] <slowguy> http://pastebin.com/FVYjBtrd  i am giving this command but ffmpeg says No such filter: '"crop'
[11:38] <polysics> 4s file at 1080p is 20Mb
[11:39] <Diogo> hi i need to convert a video file .avi into hls file(h264 aac) in shorter time possible? like run multiple instance of ffmpeg ?
[11:47] <fling> How to create a hyperlapse?
[11:48] <DopeLabs> i think thats some new hotness developed by microsoft
[11:49] <ubitux> patch welcome, but please something testable (aka a filter that doesn't take hours to process 10 seconds of video)
[11:49] <fling> DopeLabs: I see old hyperlapse videos on youtube
[11:50] <DopeLabs> ah mkay
[12:00] <AurelPop> Hi, I'm writing a small library in C, which is supposed to take frames from rtmp stream. Sometimes I get "Stream discovered after head already parsed" and the process just hangs... Can somebody explain to me exactly why? and how can I solve this? .... ome timeout and try again?...
[12:13] <t4nk460> Hi
[12:27] <AurelPop> Hi, I'm writing a small library in C, which is supposed to take frames from rtmp stream. Sometimes I get "Stream discovered after head already parsed" and the process just hangs... Can somebody explain to me exactly why? and how can I solve this? .... ome timeout and try again?...
[13:10] <fl0bbe> why hallo there :)
[13:19] <Fyr> how to convert an audio file into raw format?
[13:20] <Fyr> I have an m4a and a flac, I want to compare their raw audio.
[13:20] <spaam> ffmpeg -i superDuperFile.m4a SuperDUper_sacarasc.wav ?
[13:22] <Fyr> their md5sums have a difference. =(
[13:23] <AurelPop> How can I get the behavoir of "stimeout" from code?
[13:23] <sacarasc> Fyr: They will, because AAC is lossy, so some of the information is lost when encoded.
[13:23] <Fyr> ALAC
[13:23] <Fyr> m4a was encoded with ALAC.
[13:24] <Fyr> how can I compare the quality of two files?
[13:25] <Fyr> I've already converted FLAC-ALAC-FLAC and found that their md5-sums coincide.
[13:25] <spaam> but are they converted from the same source?
[13:26] <Fyr> it's what I need to find out.
[13:32] <Fyr> they maybe converted from the same source, but one file can be louder.
[14:13] <fajung> is it possible to force as default subtitle.srt ? like mkvmerge does
[14:14] <fajung> I've this code:"-i '~/input.avi' -sub_charenc UTF-8 -i '~/input.srt' -c:s:0 mov_text -metadata:s:s:0 language=spa -metadata:s:s:0 default-flag=0 -acodec libfdk_aac -metadata title="title...." -vcodec libx264 -b:v 950k '~/out.mp4'"
[16:38] <kode54> is anyone interested in looking at a WMA Lossless file that breaks in the decoder?
[16:38] <kode54> it's four samples long, which should at least produce one packet of audio
[16:39] <kode54> but the decoder hits the first real packet of data (13KB) and simply returns without producing any samples
[16:39] <kode54> and since the duration is not actually 1ms or more, the file's headers report a duration of 0
[16:58] <nicoit> Can anybody give me any example(c++ code) how to use ffmpeg to decode video using dxva2?
[17:15] <SkoobyNoob> does anyone know how to turn a group of mp4 files into mp3s with ffmpeg or do i have to use a script
[17:16] <sacarasc> A 1 line script.
[17:16] <Chaz6> for i in *.mp4; do ffmpeg -i $i $i.mp3; done
[17:16] <BammBamm> wow, freaky, this was my question almost exactly
[17:16] <SkoobyNoob> ok
[17:17] <BammBamm> how can i use this one (  ffmpeg -i video.mp4 -vn -acodec copy audio.m4a)   in WinFF?
[17:17] <Mavrik> of course that command line will murder your audio quality :P
[17:17] <Chaz6> yep choose your options wisely
[17:18] <SkoobyNoob> i tried ffmpeh -i *.mp4 -f mp3 -ab 192000 -vn '%(mp3)s'
[17:18] <BammBamm> how can I set up WinFF so that it rips LOSSLESSLY the m4a outta the mp4s?
[17:18] <SkoobyNoob> but that made my laptop freeze
[17:18] <relaxed> BammBamm: we don't support WinFF here
[17:18] <BammBamm> crap
[17:19] <sacarasc> BammBamm: for i in *.mp4; do ffmpeg -i "$i" -c:a copy -vn "$i".audio.m4a; done
[17:19] <relaxed> SkoobyNoob: on windows?
[17:19] <sacarasc> Play around with more bash to get it done properly.
[17:19] <SkoobyNoob> no on ubuntu
[17:19] <SkoobyNoob> i got ubuntu 14.4
[17:20] <relaxed> for i in *mp4; do ffmpeg -i "$i" -q:a 0 "${i%.*}".mp3; done
[17:20] <SkoobyNoob> hmm
[17:20] <relaxed> er, wait!
[17:20] <SkoobyNoob> yes that makes more sense
[17:20] <relaxed> for i in *mp4; do ffmpeg -i "$i" -map 0:a -q:a 0 "${i%.*}".mp3; done
[17:20] <SkoobyNoob> could you explain what each option does please?
[17:20] <SkoobyNoob> i know -i
[17:21] <relaxed> bourne shell for loop, -q:a 0 == high quality vbr
[17:22] <SkoobyNoob> cool
[17:22] <SkoobyNoob> what about -map?
[17:22] <relaxed> map only the audio (which wasn't really needed)
[17:22] <SkoobyNoob> since its already turning it into a mp3?
[17:23] <relaxed> correct
[17:23] <SkoobyNoob> so i dont have to suppress with -vn?
[17:24] <SkoobyNoob> ok is working
[17:24] <SkoobyNoob> thanks i have been looking for this answer for days
[17:24] <SkoobyNoob> ok another dumb question
[17:24] <SkoobyNoob> how can i keep the same thumbnail
[17:25] <SkoobyNoob> from the video on the mp3?
[17:25] <BammBamm> sacarasc: thanks. But when I tried that, it says "i unexpected at this moment"
[17:25] <SkoobyNoob> you are using WINFF BammBamm ?
[17:25] <BammBamm> yes
[17:26] <SkoobyNoob> man you should just come to the dark side
[17:26] <BammBamm> SkoobyNoob: no
[17:26] <SkoobyNoob> we have cookies
[17:26] <sacarasc> Then you'd also get more support here. :p
[17:26] <BammBamm> that command sacarasc gave i tried in a dosbox via ffmpeg own batchfile
[17:27] <SkoobyNoob> hm
[17:27] <SkoobyNoob> hmmmm
[17:27] <SkoobyNoob> hmmmmmmmmm
[17:27] <BammBamm> ff-prompt bat
[17:27] <sacarasc> BammBamm: It was a bash line, not a CMD line.
[17:27] <BammBamm> aaah
[17:27] <BammBamm> bash = linux, yes?
[17:27] <SkoobyNoob> you gotta use the bourne again friend
[17:27] <sacarasc> You can get bash on Windows, BSDs, others, too.
[17:28] <BammBamm> i used this succesfully in dosbox :  ffmpeg -i video.mp4 -vn -acodec copy audio.m4a
[17:28] <BammBamm> but how do i make that for a bunch of files
[17:28] <relaxed> google windows batch scripting
[17:29] <BammBamm> relaxed: i did, and this irc-channel popped up .... ;)
[17:30] <sacarasc> BammBamm: http://stackoverflow.com/questions/2591758/batch-script-loop
[17:31] <sacarasc> Or you could try powershell!
[17:32] <fajung> if I add a subtitle.srt into an output.mp4, is there any way to force as default, like in the mkvmerge does ?
[17:34] <BammBamm> sacarasc: that's a bit above my pay grade. my point is, now, WinFF accepts FFmpeg commands, so, how would I go about lossless audio extraction?
[17:34] <sacarasc> WinFF is not supported here, and you already had it.
[17:35] <sacarasc> [16:17:21] <BammBamm> how can i use this one (  ffmpeg -i video.mp4 -vn -acodec copy audio.m4a)   in WinFF?
[17:35] <sacarasc> That would be lossless.
[17:35] <BammBamm> sacarasc: that bash thing you gave me at first, can you convert it to something that would work in a dosbox
[17:35] <BammBamm> sacarasc: OK sorry bout that
[17:36] <sacarasc> I have never done any batch scripting, and if it's above your paygrade, how much do I get paid to learn? :D
[17:36] <BammBamm> kisses
[17:36] <BammBamm> ;)
[17:39] <BammBamm> so, this should work?
[17:39] <BammBamm> for /l %x in (1, 1, 100) do (
[17:39] <BammBamm>    echo %x
[17:39] <BammBamm>    ffmpeg -i *.mp4 -vn -acodec copy audio.m4a
[17:39] <BammBamm> )
[17:40] <BammBamm> ???
[17:40] <BammBamm> sacarasc: ?
[17:40] <sacarasc> [16:36:19] <sacarasc> I have never done any batch scripting,
[17:42] Action: BammBamm thinks she was counting on sacarscs  natural intelligence and chivalry, although not necessarily in that order .....
[17:42] <BammBamm> ;(
[17:42] <BammBamm> ;)
[17:42] <sacarasc> You missed my natural laziness and aversion to learning.
[17:43] <BammBamm> sacarasc:  I soooooooooo picked up on that,
[17:44] <BammBamm> i just am unrealistically optimistic
[17:52] <SkoobyNoob> well BammBamm i thought u were a dude
[17:52] <SkoobyNoob> so that is were the Chivalry cuts off for me
[17:52] <BammBamm> 95% chance you were right
[17:52] <BammBamm> so i don't bl
[17:52] <SkoobyNoob> 95%? interesting
[17:53] <BammBamm> this stupid batch keeps giving me "x is unexpected at this time"  etc
[17:53] <SkoobyNoob> i wonder if women or users with female nicks have better luck
[18:12] <nicoit> did anybody see my question?
[18:12] <nicoit> "(17:58:15) nicoit: Can anybody give me any example(c++ code) how to use ffmpeg to decode video using dxva2?"
[18:21] <saste> nicoit, http://trac.ffmpeg.org/wiki/HWAccelIntro
[18:21] <saste> also, did you see ffmpeg_dxva2.c in the source code?
[18:24] <BammBamm> so, can FFmpeg do batch lossless audio ripping/extraction?
[18:24] <BammBamm> if yes, how?
[18:25] <Fjorgynn> I use EAC for that on Windows
[18:26] <sacarasc> BammBamm: FFmpeg itself can't, but you can write a script to do it.
[18:28] <slowguy> i have a video which has dvb_teletext based subtitles..ffmpeg would say decoder dvb_teletext not found
[18:28] <slowguy> but i want just audio and video in output streams
[18:28] <slowguy> can't i tell ffmpeg to ignore the teletext stream?
[18:29] <sacarasc> -sn
[18:29] <slowguy> in all output streams?
[18:29] <slowguy> okay got it thank you
[18:48] <RenatoCRON> to com dor
[18:48] Action: RenatoCRON sorry, wrong window!
[18:54] <fajung>  is there any way to force a subtitle.srt as default track, like in the mkvmerge does ?
[19:00] <BammBamm> sacarasc: HOW  would i go about writing scripts for that? is there a ..... website? ??
[19:01] <sacarasc> There are probably many that can explain a for loop for batch or maybe a foreach loop for powershell.
[19:06] <circ-user-w3x6R> I wish to post & subscribe to ffmpeg-user at ffmpeg.org see https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-user/, but apparently something goes wrong what. "To post a message to all the list members, send email to ffmpeg-user at ffmpeg.org" I did, but my post doesn't appears in https://ffmpeg.org/pipermail/ffmpeg-user/2014-August/thread.html
[19:07] <circ-user-w3x6R> Same issue with subscription, send email, but nothing
[19:36] <pmac_> Is there any way to have FFMPEG get itself all setup, spesifically for connecting to an RTMP output, and hold in that configruation until some signal is sent to start it?
[20:46] <Phlarp> http://pastebin.com/MMpPwtL1 Say I had this command, and wanted to incorporate an adelay option so the mixed audio stream doesn't start for 3 seconds, how would I add that?
[22:25] <c_14> Either just add a ,adelay=[foobar] to the end of the filter_complex, or create a silent stream with aevalsrc of the length you want and then concat them
[00:00] --- Thu Aug 14 2014