[Ffmpeg-devel-irc] ffmpeg.log.20160115

burek burek021 at gmail.com
Sat Jan 16 02:05:02 CET 2016

[00:03:11 CET] <furq> ok apparently bilinear filtering on a prescaled screen looks much better
[00:03:19 CET] <furq> http://i.imgur.com/vRCrnUM.png
[00:03:24 CET] <furq> so thanks for that inadvertent tip if nothing else
[00:06:50 CET] <furq> i'll be sure to keep you all up to date with all the latest developments on what filters i use to play neo-geo games
[01:43:57 CET] <genii> Hi. I can't figure out why one video plays properly (exiftool results for that one  http://paste.ubuntu.com/14500184/ ) and the other doesn't ( exiftool results for that one: http://paste.ubuntu.com/14500181/ )  Could it be Major Brand                     : MP4  Base Media v1 [IS0 14496-12:2003]          versus  Major Brand                     : MP4 v2 [ISO 14496-14]   ? If so can I get it to encode to that version, I already tried -level 4.2 option
[01:48:29 CET] <genii> The format is side-by-side left view first ( 3D )
[01:49:01 CET] <Guest83269> why don't you add a progress bar in your ffmpeg ? its 2016 and no native progress bar
[01:50:09 CET] <Guest83269> also why dont you add a basic basic encoding queue system inside ffmpeg
[02:49:01 CET] <llogan> patch welcome, anonymous complainer who isn't here anymore.
[02:49:58 CET] <llogan> genii: play in what? are you using ffmpeg?
[02:51:30 CET] <ac_slater> hey all. I'm writing an encoder and I cant figure out how to use the "b" (bitrate) option. When I do `-b:v myCodec` via the ffmpeg binary, how does this make it to my codec and how to I get the value?
[02:52:19 CET] <ac_slater> s/myCodec/500 or some number
[02:56:20 CET] <waressearcher2> ac_slater: hallo und herzlich willkommen
[02:57:07 CET] <ac_slater> waressearcher2: thanks mate
[02:57:31 CET] <ac_slater> I figured my issue out. '-b:v' makes its way into AVCodecContext.bit_Rate
[02:57:35 CET] <ac_slater> bit_rate *
[03:07:11 CET] <k_sze[work]> Hello.
[03:07:31 CET] <k_sze[work]> Just me or by video frames are refcounted when decoding FFV1 level 3?
[03:07:45 CET] <k_sze[work]> *by default
[04:29:19 CET] <hayatae> Hey guys, I'm looking at the exact same problem as this:  https://lists.libav.org/pipermail/ffmpeg-user/2010-September/027126.html
[04:29:32 CET] <hayatae> Doesn't look like an answer was ever posted. Does anyone here have any ideas?
[04:44:25 CET] <waressearcher2> hayatae: hallo und herzlich willkommen
[04:48:00 CET] <hayatae> hej, tak
[06:40:34 CET] <ac_slater> hey guys. I have a frame of video in a AVFrame. To pass it to my encoder, I need to copy the data to an AVPacket. My pixel data is YUV420P. My AVPackets are just green. This probably has to do with copying the planes. Clues?
[06:53:17 CET] <waressearcher2> ac_slater: hallo und herzlich willkommen
[07:01:33 CET] <ac_slater>  /quit
[07:03:50 CET] <Jetfantastic> Hi, I was very recently (like 5 minutes ago) recommended ffmpeg because I needed an easy way to convert multiple files from h.265 to h.264 at the same time, specifically because my bluray player doesn't support h.265. How would I convert a folder of h.265 mkv files into h.264 mp4 files?
[07:09:18 CET] <douglasbay> How can I tell if the I frame in my video is an IDR frame, through ffprobe?
[07:10:23 CET] <xintox> what's a good HLS player for web
[07:10:43 CET] <xintox> also looking for a desktop player that can open multiple instances simultaneously.
[07:11:05 CET] <xintox> i've been usign clappr but its really sluggish
[07:11:12 CET] <xintox> and vlc only lets me open one instance
[07:46:51 CET] <Jetfantastic> Hi, I was very recently (like 5 minutes ago) recommended ffmpeg because I needed an easy way to convert multiple files from h.265 to h.264 at the same time, specifically because my bluray player doesn't support h.265. How would I convert a folder of h.265 mkv files into h.264 mp4 files?
[07:54:13 CET] <Betablocker> hello world !
[07:55:15 CET] <Jetfantastic> Does anyone use this program and know how to convert many files at once?
[07:59:23 CET] <relaxed> Jetfantastic: you can write a script to do it
[07:59:39 CET] <Jetfantastic> How would I do that
[08:00:10 CET] <Jetfantastic> I just got recommended the program and there seems to be no guide for the inexperiences
[08:00:16 CET] <Jetfantastic> *Inexperiences
[08:00:18 CET] <relaxed> Which OS?
[08:00:23 CET] <Jetfantastic> Win10
[08:00:42 CET] <Jetfantastic> 64bit
[08:01:37 CET] <relaxed> I don't know how to script for windows, but google should give you plenty of examples. Google:  windows ffmpeg for loop
[08:02:09 CET] <Jetfantastic> I just wanted to watch Rick and Morty on my Bluray Player :(
[08:05:21 CET] <relaxed> if it's a smart bluray player then it might read whatever have off a disc
[08:06:11 CET] <Jetfantastic> Nah, It doesn't read h.265 or mkv
[08:06:17 CET] <Jetfantastic> for whatever reason
[08:06:49 CET] <Jetfantastic> I might just do it on my TV through PLEX
[08:09:46 CET] <chungy> in bash you'd do: for file in *.mkv; do ffmpeg -i "$file" -c:a copy -c:v h264 "$(basename "$file" .mkv).mp4"; done
[08:09:55 CET] <chungy> I have no clue about cmd/powershell myself either
[08:11:41 CET] <chungy> you can actually install bash on Windows via Cygwin, that's what I'd do, but it's probably a heavyweight solution
[11:03:21 CET] <EvilDin> hi, I have source video which ffprobe says it has length 11.99s, then I transcode that high quality video to lower quality, when I checked transcoded file with ffprobe I got length 12,03s, 40ms more then original. We have web player which detects video length and it is weird that when you switch quality video length is changed. Is there anyway how could I transocde that this would not happen?
[11:21:57 CET] <waressearcher2> EvilDin: hallo und herzlich willkommen
[12:24:42 CET] <Fyr> guys, how to scale down volume since 34:02 to 38:42 of a WAV?
[12:30:46 CET] <waressearcher2> -af volume=volume=-20dB:precision=fixed ?
[12:31:19 CET] <Fyr> for the whole file?
[12:39:37 CET] <Fyr> is it possible to split frequencies and scale down only high ones?
[13:08:35 CET] <durandal_1707> Fyr: that's called equalizer
[13:09:35 CET] <durandal_1707> and high one can be removed with lowpass filter too
[13:18:38 CET] <Fyr> durandal_1707, thanks for the idea. is there a way to draw a curve to decrease some frequencies while keeping the others?
[13:20:23 CET] <durandal_1707> Fyr: drawing by hand - no, but anequalizer can draw curves from params you set
[13:24:20 CET] <Nanabot> ffmpeg -loop 1 -i img.png -i audio.mp3 -c:a copy -shortest out.mp4
[13:24:51 CET] <Nanabot> This works for making a music video with a static image, but I noticed it added 4 seconds of silence to the end of my video...
[13:25:38 CET] <waressearcher2> Nanabot: -shortest
[13:26:17 CET] <Nanabot> err yeah I had that -r 1 -shortest output.mp4
[13:26:31 CET] <waressearcher2> -t length-4
[13:29:03 CET] <Nanabot> I noticed that removing -r 1 (thus default framerate 25) makes the duration normal
[13:29:49 CET] <Nanabot> ah thanks, -t (actual song length) works
[14:01:42 CET] <Nanabot> Is there a flag I can add so the colors aren't shifted? I noticed the colors are off.
[14:51:29 CET] <Nanabot> Oh, it was just my video player being idiotic. -pix_fmt yuv420p (encoding png to h264)
[14:52:27 CET] <Nanabot> It didn't show any difference until I switched player.
[14:56:24 CET] <Nanabot> I spent too long to figure that out.
[16:40:47 CET] <Prelude2004c> hey guys.. good day.. anyone know how with ffmpeg i can set the version to 2 for HLS .. i think right now default is hls version 3
[16:41:50 CET] <JEEB> I'm really sorry if you're having to support something that only supports that :P
[16:41:58 CET] <Betablocker> lol
[16:44:15 CET] <Neon> I'm trying to mix 10 audio files using ffmpeg and after a while it fails with the message Error while filtering: Cannot allocate memory. I'm using a command like: ffmpeg -i file1 -i file2 ... -filter_complex amix=inputs=10,volume=10 -c aac mixdown.aac. Any idea how I can fix this problem?
[16:46:14 CET] <durandal_1707> try amerge filter
[16:46:19 CET] <Neon> Damn, it fails after mixing 1:29:54.21 h and the longest input file has 1:30:01 h... Coincidence?
[16:49:03 CET] <Neon> [aac @ 0000000002d696a0] Unsupported number of channels: 10
[16:49:50 CET] <Neon> When I try to use amerge. I've already mixed another set of files and it went well. I used amix for them, because amerge gave me that error.
[16:59:23 CET] <Neon> Trying amerge with libvorbis now.
[17:04:24 CET] <durandal_1707> Neon: what version you use?
[17:04:44 CET] <durandal_1707> the limit is 64 here
[17:06:33 CET] <Neon> durandal_1707:
[17:06:34 CET] <Neon> ffmpeg version N-77504-gbaf4c48
[17:07:24 CET] <Neon> Not sure if that's the version string. It looks unusual to me. But that's what I get from ffmpeg -v.
[17:07:40 CET] <Neon> Maybe it's because it's a Windows version.
[17:10:26 CET] <Neon> Well that didn't work as expected. I've got a 20 minute ogg now that seems silent.
[17:22:32 CET] <alcros> Hey guys, i have a question. Is it possible to cut the ads (mostly 2-3) out of a TV capture without reencoding the video in one ffmpeg command?
[17:30:10 CET] <Neon> alcros: If it helps, this explains how to cut a piece out of a video (with copy encoding): http://superuser.com/questions/377343/cut-part-from-video-file-from-start-position-to-end-position-with-ffmpeg/377407#377407 But you need to know the time markers and I don't know if you can make multiple cuts in one command.
[17:38:28 CET] <alcros> Neon: i know the time markers, this is not the problem...when i use something like this: ffmpeg -i test.ts -ss 10 -t 10 -ss 60 -t 10 -c:a copy -c:v copy test_cutted.ts i get only a 10sec video...but thanks for your help
[18:51:42 CET] <Zeranoe> Does 0.5.15 support any way to find the build time configuration? I don't believe it has avcodec_configuration(void).
[19:06:58 CET] <yongyung> Not really an ffmpeg question, but since people here are using it: I want to youtube-dl bestaudio+bestvideo, but I don't like the default behavior. For example, youtube-dl prefers 128k vorbis over 160k opus, and h264 over vp9. I tried -f 141/251/bestaudio, but somehow it still downloads vorbis instead of 251 when 141 isn't available
[19:14:38 CET] <yongyung> Actually, running from console works but not from a batch file, even though it should be exactly the same looking at the batch output
[19:14:40 CET] <livingBEEF> 248+251/248+250/137+140/best
[19:14:49 CET] <livingBEEF> how about something like that?
[19:19:14 CET] <yongyung> livingBEEF: LMAO I got it. If you quote the link it doesn't work, if you don't it does. Oo
[19:21:35 CET] <livingBEEF> Is that bash?
[19:23:07 CET] <PsychiC> Hello  guys and gals , i´m having issues with transcoding an input stream from OBS , reencode it and send to another rtmp destination. I can´t get ffmpeg to work or it gives audio only or out of sync etc. I´ve been trying for days with various OS and ffmpeg versions , but I´m stuck :(  I could really use some help which would be greatly appreciated.
[19:23:26 CET] <furq> yongyung: -f bestvideo[ext=webm]+bestaudio[ext=webm]
[19:23:35 CET] <furq> please don't blame me for how ridiculous that is
[19:24:31 CET] <livingBEEF> furq: doesn't seem all that bad to me
[19:25:07 CET] <livingBEEF> -o seems much worse
[19:25:26 CET] <furq> well you'd expect -f best[ext=webm] to work
[19:25:41 CET] <furq> but that does the same thing as -f webm
[19:26:34 CET] <yongyung> furq: The problem with that is that it would prefer opus 160k over aac 256k, and not all videos seem to have vp9 yet
[19:26:48 CET] <yongyung> (141 is aac 256k)
[19:26:49 CET] <livingBEEF> separage video/audi DASH probably has something
[19:26:54 CET] <livingBEEF> to do with that
[19:28:32 CET] <livingBEEF> well... if opus is better even at those bitrates, it's the correct thing
[19:28:57 CET] <yongyung> livingBEEF: Opus 160k is guaranteed not better than aac 256k
[19:28:58 CET] <livingBEEF> it can always mux it together as mkv or something
[19:29:14 CET] <furq> opus 160k should still be pretty much transparent though
[19:29:20 CET] <livingBEEF> how sure are you about that?
[19:29:20 CET] <yongyung> The only reason it prefers opus there is the extension
[19:29:33 CET] <furq> does bestaudio prefer vorbis over opus
[19:29:37 CET] <livingBEEF> Isn't aac quite a bit older?
[19:29:48 CET] <furq> good aac encoders are competitive with opus
[19:30:03 CET] <furq> good aac and opus are both basically transparent at 128k though afaik
[19:30:14 CET] <furq> 256k seems like overkill for stereo
[19:30:24 CET] <yongyung> livingBEEF: Audio codecs really don't have the same development as video codecs, mp3, vorbis, aac, opus are all pretty close, a significantly higher bit rate will always sound better
[19:30:26 CET] <LeonG> what aac encoder does youtube use?
[19:30:32 CET] <furq> no idea
[19:30:40 CET] <furq> probably not appleaac which i believe is still the best
[19:30:54 CET] <LeonG> because some were really badf at 256kbps
[19:30:56 CET] <yongyung> The description says "google", but I'd assume they use fdk_aac
[19:30:59 CET] <furq> if it's using FAAC or some old library then 160k opus probably is better
[19:31:05 CET] <furq> i doubt they would be though
[19:31:06 CET] <LeonG> yeah
[19:31:16 CET] <livingBEEF> afaik mp3/opus is a BIG difference at very low bitrates
[19:31:37 CET] <yongyung> There is, but 192k mp3 still sounds better than 128k opus
[19:31:42 CET] <livingBEEF> you can have like 10-20k mono w/ opus
[19:32:07 CET] <furq> apparently youtube did use faac at one point
[19:32:50 CET] <furq> according to an unsourced hydrogenaudio post they use fdk now
[19:35:11 CET] <furq> does bestaudio not always prefer aac then
[19:35:17 CET] <furq> it does on everything i've tried
[19:36:06 CET] <yongyung> furq: It does, but it's extension isn't webm
[19:36:19 CET] <furq> i mean just bestaudio with no extension specified
[19:36:27 CET] <LeonG> webm only can contain vorbis or opus audio
[19:36:54 CET] <yongyung> furq: When you don't specify and extension aac is preferred, which is good, but it prefers 128k vorbis over 160k opus
[19:37:12 CET] <yongyung> And a lot of videos don't have 256k aac available
[19:37:22 CET] <yongyung> In which case I want to choose opus, not vorbis
[19:37:29 CET] <LeonG> good point
[19:38:20 CET] <furq> are there videos where 160k opus is available but 256k aac isn't
[19:38:28 CET] <furq> i assumed those both went with the 720p video
[19:38:46 CET] <furq> the next step down is 128k vorbis and then 70k opus
[19:40:36 CET] <yongyung> furq: Yes there are, you want one?^^
[19:40:54 CET] <furq> sure
[19:41:09 CET] <furq> youtube seems like a mess so it wouldn't surprise me, but i couldn't find any
[19:41:24 CET] <yongyung> https://www.youtube.com/watch?v=qKHavcFDaSc
[19:42:18 CET] <furq> that's really dumb
[19:43:23 CET] <furq> i can sort of see why youtube-dl's -f specification is so nuts now
[19:43:35 CET] <furq> not that it's as good as it could be
[19:44:33 CET] <yongyung> Well if youtube-dl --output "%(title)s.%(ext)s" -f "303/299/302/298/248/bestvideo+141/251/bestaudio" would work it'd be fine with me
[19:44:41 CET] <yongyung> But he's just ignoring the audio
[19:45:28 CET] <yongyung> And yeah, not defining --output has no effect
[19:49:40 CET] <yongyung> I guess I'm gonna load them separately and write a script to merge everything
[19:54:44 CET] <livingBEEF> yongyung: thoe problem is that you have to have EVERYTHING you want between '/' signs.
[19:55:21 CET] <livingBEEF> You can't say "I want any of these video formats in this order and than any of these audio formats"
[19:55:48 CET] <livingBEEF> you have to specify them as pairs like 1+2/1+3/9+2
[19:56:13 CET] <yongyung> livingBEEF: That'd be so many pairs that writing a script is probably easier
[19:58:32 CET] <livingBEEF> yongyung: you can only specify the semsible ones - only webm+webm and mp4+mp4
[19:59:05 CET] <yongyung> livingBEEF: They're all sensible to me, I merge to mkv
[20:00:29 CET] <livingBEEF> 303+141/303+251/303+bestaudio/299+141/299+251/299+bestaudio/302+141/302+251/302+bestaudio/298+141/298+251/298+bestaudio/248+141/248+251/248+bestaudio/bestvideo+141/bestvideo+251/bestvideo+bestaudio/
[20:05:12 CET] <livingBEEF> you could have someting like `{format1,format2,...}+{aformat1,aformat2,...} | tr -d ' '` if it's bas
[20:05:16 CET] <livingBEEF> *bash
[20:08:49 CET] <livingBEEF> concerning http://news.softpedia.com/news/zero-day-ffmpeg-vulnerability-lets-anyone-steal-files-from-remote-machines-498880.shtml
[20:09:02 CET] <livingBEEF> an idea which versions are affected
[20:09:09 CET] <livingBEEF> *any?
[21:06:19 CET] <micechal> how can I use ffmpeg to fix a file broken this way: https://bpaste.net/show/ec9119777994 ?
[21:06:48 CET] <micechal> I'd like to enable seeking there and get rid of the errors
[22:15:00 CET] <Prelude2004c> hey guys.. quick question.. i have a source that sometimes the frame rates per second drop.. any way for me to deletect frames and if they drop below say 20fps to automatically exit so ic an restart the request
[22:15:26 CET] <klue> What versions does the recent ffmpeg vulnerabiliy affect? (The one that provides SSRF, local file reads, etc)
[22:29:31 CET] <JEEB> klue: everything but current master and releases made a few hours ago http://git.videolan.org/?p=ffmpeg.git;a=summary
[22:31:44 CET] <JEEB> looking at the patches to HLS recently I'm not really sure what else could have been achieved other than reading local files that the running ffmpeg process had access to?
[22:35:04 CET] <JEEB> ok, so it was about meta playlists
[22:35:07 CET] <JEEB> that contain urls
[00:00:00 CET] --- Sat Jan 16 2016

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