[Ffmpeg-devel-irc] ffmpeg.log.20170527

Sun May 28 03:05:02 EEST 2017

[00:07:38 CEST] <c_14> use 2 outputs
[00:59:03 CEST] <guther> Does -preset in libx264 influence quality when used with -cfr ?
[00:59:41 CEST] <furq> yes
[00:59:56 CEST] <furq> assuming you mean -crf, but it'll do that with any rc
[01:00:53 CEST] <iive> it either influences quality or bitrate
[01:01:01 CEST] <iive> or both
[01:03:10 CEST] <furq> quality much more than bitrate
[01:03:19 CEST] <furq> it's totally normal for bitrate to increase when you use a slower preset
[01:03:34 CEST] <marquisor> true :)
[01:05:25 CEST] <iive> furq: shouldn't it be the opposite :P
[01:05:30 CEST] <furq> no
[01:05:44 CEST] <furq> i mean intuitively yes but it's never worked like that
[01:06:07 CEST] <furq> the meaning of a crf value depends on the encoder settings
[01:06:27 CEST] <guther> sorry, lost connection :(
[01:06:51 CEST] <furq> guther: http://ffmpeg.gusari.org/irclogs/ffmpeg.log.20170527
[01:06:53 CEST] <guther> furq, which codec do you preferre?
[01:07:01 CEST] <furq> x264
[01:07:17 CEST] <guther> Uh, theres a log - good to know, thanks
[01:07:52 CEST] <guther> x264 is the same as that libx264 i use?
[01:07:55 CEST] <furq> yes
[01:08:04 CEST] <guther> Fine
[01:08:24 CEST] <guther> Does n-pass enc any sense nowadays?
[01:08:33 CEST] <furq> not unless you're targeting a particular filesize
[01:09:02 CEST] <guther> how much is it in percent, roughly
[01:09:19 CEST] <furq> i have no idea
[01:09:27 CEST] <guther> What preset do you use?
[01:09:30 CEST] <furq> veryslow
[01:10:17 CEST] <furq> the only use for 2-pass with x264 is in abr mode, which you almost never use
[01:10:33 CEST] <guther> uhm, i thought, even slower seems a bit too time consuming to me
[01:10:44 CEST] <furq> the first pass makes it more likely to hit the target bitrate
[01:10:59 CEST] <furq> in crf mode it makes literally no difference
[01:11:15 CEST] <furq> other than your cpu fan will be running for longer
[01:11:32 CEST] <guther> Yeah, thats the impression i got from reading the inet
[01:11:56 CEST] <furq> any preset below faster will give passable results
[01:12:05 CEST] <furq> you can always just increase the crf to compensate
[01:12:57 CEST] <furq> i used to use slow at crf 20 for >=720p and that stuff all still looks fine to me
[01:14:18 CEST] <furq> if you keep your sources backed up somewhere then it doesn't really matter what you use
[01:14:55 CEST] <guther> I'm experimenting with crf 21 and slow/slower - that seems reasonable to me
[01:15:34 CEST] <guther> With 1280
[01:18:11 CEST] <guther> Btw, is there a way to pass cutlists to ffmpeg?
[01:19:42 CEST] <furq> !muxer segment @guther
[01:19:42 CEST] <nfobot> guther: http://ffmpeg.org/ffmpeg-formats.html#segment_002c-stream_005fsegment_002c-ssegment
[01:19:45 CEST] <furq> see -segment_times
[01:19:48 CEST] <furq> that's the closest thing i know of
[01:20:22 CEST] <BtbN> Wasn't cue sheet support recently merged? Or is it still on the ml?
[01:20:30 CEST] <furq> oh really
[01:20:36 CEST] <furq> that would make something i was planning on writing much easier
[01:21:50 CEST] <BtbN> nah, still sitting on the ml
[01:22:41 CEST] <tute23> Hello, a question, I use ffmpeg to make a live broadcast by icecast from "alsa", I have a problem launching a new instance of ffmpeg for alsa record in a file and being able to restart the streaming ffmpeg for any problems without cutting the recording Of file , error : "device is bussy"
[01:23:16 CEST] <BtbN> seems like your device does not support multiple processes accessing it at the same time
[01:24:51 CEST] <tute23> Is there a way to launch ffmpeg without locking the device ???
[01:25:47 CEST] <tute23> or any possible solution on the alsa side with alsa loop?
[01:25:50 CEST] <BtbN> it's up to the device if it supports multiple client
[01:25:56 CEST] <BtbN> this one does not
[01:26:03 CEST] <BtbN> could use pulseaudio to workaround that limitation
[01:26:08 CEST] <tute23> is a USB audio device
[01:27:56 CEST] <tute23> Ok, thank you, I'll try with press, should not have problems launching for example arecord and ffmpeg at the same time?
[01:29:00 CEST] <BtbN> whoever accesses the device first wins
[01:29:05 CEST] <BtbN> no matter if it's ffmpeg or arecord.
[01:29:36 CEST] <DHE> av_interleaved_write_frame() is supposed to effectively write all streams in a globally sorted DTS order, right? because I'm looking at the output (mpegts using wireshark) and that's not true.
[08:01:44 CEST] <crziter> I'm going to use FFMPEG in my app to display live video from cameras, do I need to synchronize video when live streaming like this?
[08:02:11 CEST] <bigbrous40> i have $250,000 to invest  , what should i buy
[08:06:04 CEST] <tdr> bigbrous40, how about a lot of candy corn and swedish fish?
[08:23:43 CEST] <thebombzen> Sweedish Fish are much tastier than Candy Corn
[08:23:48 CEST] <thebombzen> Kit Kats are still bet candy though
[08:28:08 CEST] <tdr> i prefer candy corn, they dont stick to your teeth
[08:28:22 CEST] <tdr> still rather have decent chocolate tho
[09:05:26 CEST] <Fyr> guys, I can't google out examples of usage of loudnorm.
[09:05:31 CEST] <Fyr> could you help me out?
[09:06:59 CEST] <Fyr> or with dynaudnorm filter
[09:34:21 CEST] <c3r1c3-Win> Fyr: Maybe try "Broadcast Loudness Standards"
[09:34:51 CEST] <Fyr> c3r1c3-Win, what?
[09:35:13 CEST] <Fyr> I need an example of -filter:a loudnorm
[09:35:27 CEST] <Fyr> I has a huge number of parameters which I don't understand.
[09:35:46 CEST] <c3r1c3-Win> Fyr: Yesa nd do you know this filter exists?
[09:36:00 CEST] <c3r1c3-Win> Do you know what basis this filter has and what the parameters mean?
[09:36:05 CEST] <Fyr> no
[09:36:08 CEST] <Fyr> not at all
[09:36:21 CEST] <Fyr> found it on stackexchange without examples.
[09:36:34 CEST] <c3r1c3-Win> It comes from the CALM act passed in the USA (and before that another one passed in Europe) to controll the peak and overall loudness of audio.
[09:36:45 CEST] <c3r1c3-Win> *of audio for broadcast.
[09:37:01 CEST] <Fyr> c3r1c3-Win, how do I use those filters?
[09:37:07 CEST] <JEEB> wasn't the ebu r-128 filter for that?
[09:37:19 CEST] <JEEB> it did some loudness stuff
[09:37:43 CEST] <Fyr> I have a crappy videos with too quiet sound and some times, normal sound.
[09:38:15 CEST] <c3r1c3-Win> So since you have no idea what the various variables mean (or even why this filter exists) I told you to google about the "Broadcast Loudness Standards" and that would (hopefully) educate you about it.
[09:38:16 CEST] <Fyr> I need to make it sound more pleasurable.
[09:38:33 CEST] <c3r1c3-Win> Fyr: If you just want to lower the dynamic range, use a compressor.
[10:05:33 CEST] <crziter> Does avcodec_send_packet/avcodec_receive_frame reorder video frames by PTS value internally?
[10:06:04 CEST] <crziter> Or we have to reorder it by ourself?
[12:03:37 CEST] <debianuser> tute23, BtbN: it probably doesn't matter any more, but in case you're curious - don't capture from "hw:CardName" pcm if you want multiple apps capturing at the same time, in plain alsa use "dsnoop:CardName" instead (or capture from "default" pcm, it should include "dsnoop" in capture chain by default)
[12:13:04 CEST] <momomo> if i setdar=16/9 ... what is going to happen .. it will adjust the ratio right ... but what is setsar for ??
[12:21:30 CEST] <Threads> https://ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar
[12:21:43 CEST] <Threads> momomo read ^
[12:21:47 CEST] <Threads> !filter
[12:51:39 CEST] <vlitomsk> Hi guys! Is there media container which supports video,audio and some 2d vector overlay for video? I need to append some geometry primitives to the video without re-coding it. Heard about mpeg4 BIFS, but looks like it died in 2000s :(
[12:52:31 CEST] <vlitomsk> Sometimes i need different overlays for same "clean" video, and re-coding is cpu-wasting. Is there any widely (or not so widely ;) ) supported format including this feature?
[13:51:15 CEST] <ChocolateArmpits> vlitomsk, do you need support for additional data tracks ? rendering stuff on top is not a common function with media players outside of subtitles
[13:53:33 CEST] <vlitomsk> ChocolateArmpits: yes, i need additional data track. Can it be done through subtitles?
[13:54:58 CEST] <ChocolateArmpits> Substation alpha format can draw vectors, but I'm not sure how far you can push it
[13:58:08 CEST] <ChocolateArmpits> MXF has support for custom data tracks, but you'd have to do the rendering part yourself. MXF decoders presume that any unidentified parts should be skipped
[14:11:07 CEST] <vlitomsk> ChocolateArmpits: i understood why you doesn't use abbreviation for Substation alpha format :-) Yes, it supports vector graphics (http://docs.aegisub.org/3.2/ASS_Tags/) and  I hope that the drawing scripts are encoded to smth binary
[15:26:15 CEST] <tgunb> Hey guys, why doesn't ffmpeg/ffprobe detect the picture metadata in jpegs from my camera? (like camera model, settings and stuff)
[15:26:33 CEST] <tgunb> I use ffmpeg version 2.8.11-0ubuntu0.16.04.1
[15:29:10 CEST] <tgunb> the configuation contains --disable-decoder=libopenjpeg --enable-libopenjpeg. is that the reason?
[15:29:20 CEST] <JEEB> no
[15:29:26 CEST] <JEEB> openjpeg is used only for j2k
[15:29:56 CEST] <JEEB> also no idea how many tags libavformat reads from JPEG files :P
[15:31:39 CEST] <tgunb> even -map_metadata doesn't copy them. are you saying, that that is not implemented?
[15:32:05 CEST] <tgunb> (reading those jpeg tags)
[15:32:14 CEST] <JEEB> seems like 2.1+ has some EXIF support
[15:32:17 CEST] <JEEB> https://github.com/FFmpeg/FFmpeg/commit/bb4e1b4cf910af0de2bc884c75544603c40010cc
[15:32:50 CEST] <JEEB> of course no idea if ffmpeg.c or ffprobe.c utilize that
[15:35:17 CEST] <momomo> i am getting this message
[15:35:18 CEST] <momomo> Past duration 0.934120 too large
[15:35:24 CEST] <momomo> and then it disconnects
[15:35:28 CEST] <momomo> when i am generating hls
[15:35:54 CEST] <momomo> i've tried with reconnect_at_eof reconnect_streamed and more ... but nothing seems to bite
[15:35:59 CEST] <momomo> the connection dies
[15:37:39 CEST] <momomo> if I set reconnect_at_eof it works and does not drop the connection
[15:37:45 CEST] <momomo> however, it sends up in a loop
[15:38:13 CEST] <momomo> jumping back in the stream and then encountering the same problem again and do that over and over again
[15:38:34 CEST] <tezogmix> hey all, how would I create a custom batch file that has unique ffmpeg command line entries and file inputs for fffmpeg-windows? I posted 4 examples to pastbin right now. In the 1st 2 examples, I just run each of those batch files separately and once the full task is done. Example 3 & 4 are just manual commands.
[15:38:49 CEST] <tezogmix> https://pastebin.com/REFycT3Q
[15:39:49 CEST] <tezogmix> I found some of the 1st 2 example batch files from a regular web search and those have been working alright but sometimes I have a bunch of different files and would be nice to have run overnight
[15:40:12 CEST] <furq> are you asking how you run all of those commands at once in order
[15:40:24 CEST] <tezogmix> hey furq , yes something like that...
[15:40:32 CEST] <furq> you do the thing you just did on pastebin
[15:40:36 CEST] <tezogmix> so each example is a single operation -
[15:40:49 CEST] <tezogmix> i just copied it all and put on paste bin
[15:41:03 CEST] <tezogmix> oh, so i just put all those lines as is in one batch file then?
[15:41:06 CEST] <furq> yes
[15:41:23 CEST] <furq> also i don't think the third command does what you think it does
[15:41:37 CEST] <furq> -r doesn't do anything as an input option if the input is mp4
[15:42:02 CEST] <tezogmix> oh with example 3, sometimes i had a 59.94fps file and wanted to convert it to 29.97
[15:42:21 CEST] <tezogmix> and sometimes it's 50fps and want it to be 25
[15:42:57 CEST] <tezogmix> unless it's something else you were talking about.
[15:44:54 CEST] <momomo> furq: can you help with this ? i've been struggling with this one for weeks
[15:44:57 CEST] <momomo> maybe it's a bug
[15:45:08 CEST] <momomo> I am trying to update version now
[15:48:39 CEST] <tezogmix> <furq> -r doesn't do anything as an input option if the input is mp4 >>> but if the input is a different fps and we wanted a different fps output, then the "-r" would be appropriate furq ?
[15:51:30 CEST] <tezogmix> also furq with the "pause" line that I had below example 1 + example 2, if I'm combing all 4 examples together, then the "pause" would be right below the last command right? (so that I wouldn't have to be there to press any key to continue the next action found in the batch file)?
[15:51:55 CEST] <tezogmix> combing=combining*
[15:55:27 CEST] <tezogmix> so something like this (where all 4 commands are in 1 single batch) > https://pastebin.com/3dG9uBzb
[19:44:45 CEST] <hrm99> Hi! Isn't there a Windows installer for ffmpeg?
[19:46:15 CEST] <hrm99> Ok so I copy the .exe's to C:\Windows\System32
[19:46:46 CEST] <hrm99> No sorry. Can I paste some?
[19:46:49 CEST] <hrm99> C:\yt>youtube-dl https://twitter.com/TRAKGIRL/status/868159765502939136
[19:46:49 CEST] <hrm99> [twitter] 868159765502939136: Downloading webpage
[19:46:49 CEST] <hrm99> [twitter:card] 868159765502939136: Downloading webpage
[19:46:49 CEST] <hrm99> [twitter:card] 868159765502939136: Downloading m3u8 information
[19:46:49 CEST] <hrm99> [download] Destination: TRAKGIRL - That time @flyinglotus took us to church.-868159765502939136.mp4
[19:46:52 CEST] <hrm99> ERROR: m3u8 download detected but ffmpeg or avconv could not be found. Please install one.
[19:47:21 CEST] <JEEB> put it next to youtube-dl -.-
[19:47:27 CEST] <hrm99> I put ffmpeg.exe, ffplay.exe and ffprobe.exe into C:\Windows\System32
[19:47:44 CEST] <JEEB> and make sure they are not blocked as downloaded files
[19:47:59 CEST] <hrm99> JEEB: Thanks man
[19:48:07 CEST] <hrm99> That sounds like the sane way of doing it
[19:48:22 CEST] <hrm99> They can be blocked? Hold up
[19:48:37 CEST] <JEEB> right click and properties, the unlock thing is there if it is there
[19:48:39 CEST] <hrm99> YES!!
[19:48:44 CEST] <hrm99> we have ACTION here!! :D
[19:49:09 CEST] <hrm99> it is time to sample my favorite part of my favorite beat of today :)
[19:49:18 CEST] <hrm99> thanks JEEB
[19:50:15 CEST] <hrm99> * beat of the day
[20:08:29 CEST] <james999> what is your favorite beat?
[20:12:06 CEST] <hrm99> james999: i shall export it
[20:12:12 CEST] <hrm99> for EVERYONE TO SEE!!
[20:12:18 CEST] <hrm99> (but mainly for you to see)
[20:18:32 CEST] <hrm99> james999: https://soundcloud.com/fuckdeg/trakgrl-flylo i just looped up my youtube-dl ffmpeg clip
[20:21:35 CEST] <hrm99> sorry had to eq the sound a bit, reuploaded
[20:21:56 CEST] <hrm99> thanks again guys!
[21:32:47 CEST] <Soni> can I rotate a JPEG with ffmpeg?
[21:35:50 CEST] <dystopia_> you should be able to with transpose
[21:36:59 CEST] <dystopia_> ffmpeg -i in.jph -vf "transpose=1" out.jpg
[21:37:16 CEST] <dystopia_> 0 = 90CounterCLockwise and Vertical Flip (default), 1 = 90Clockwise, 2 = 90CounterClockwise, 3 = 90Clockwise and Vertical Flip
[21:37:23 CEST] <Soni> cool, thanks
[21:37:27 CEST] <dystopia_> np
[21:37:34 CEST] <dystopia_> there is also vflip and hflip
[21:37:40 CEST] <dystopia_> which do basically the same thing
[21:52:52 CEST] <Exairnous> Hi, when using ffmpeg to split a stereo audio file to two mono files ffmpeg amplifies a very soft part in the right channel.  How can I stop it doing this?
[21:55:07 CEST] <Exairnous> terminal output: https://pastebin.com/tgfGcwHx
[22:10:10 CEST] <durandal_1707> Exairnous: use lavfi filters instead
[22:11:01 CEST] <Exairnous> durandal_1707: got an example handy?
[22:11:41 CEST] <durandal_1707> read docs and wiki entry
[22:12:26 CEST] <Exairnous> ok
[22:13:31 CEST] <durandal_1707> basically use channelsplit filter
[22:14:26 CEST] <Exairnous> thanks
[22:20:25 CEST] <marquisor> lol what a nickname
[22:21:20 CEST] <bob_twinkles> is it normal for filter performance to degrade the further it gets in to a job?
[22:25:31 CEST] <durandal_1707> bob_twinkles: no, what filter ?
[22:27:46 CEST] <Exairnous> durandal_1707: Using this command I got the same:
[22:27:51 CEST] <bob_twinkles> durandal_1707: I'm not sure exactly where the slowdown is coming from, but this is the chain https://pastebin.com/z5aMgsPG
[22:27:54 CEST] <Exairnous> ffmpeg -i "Keep the Whole World Singing - Tenor Right.mp3" -lavfi "[0:a]channelsplit[a1][a2]" -map "[a1]" -ac 2 f.l.mp3 -map "[a2]" -ac 2 f.r.mp3
[22:28:47 CEST] <bob_twinkles> but it's gone from 9FPS to ~2 over the course of 5k frames
[22:29:28 CEST] <durandal_1707> Exairnous: ffmpeg does not amplifies anything out of nothing, make sure its not libmp3lame issue
[22:30:37 CEST] <Exairnous> durandal_1707: I tested using sox and it doesn't have this issue, give me a sec, let me link an image to illistrate
[22:31:18 CEST] <durandal_1707> because sox doesnt utilize lame
[22:32:48 CEST] <Exairnous> so if I convert it first to wave and then run it through ffmpeg it shouldn't run into that problem?
[22:32:56 CEST] <durandal_1707> bob_twinkles: move fps after scale
[22:33:37 CEST] <durandal_1707> Exairnous: no mater what you do lame encoder will do it
[22:34:15 CEST] <Exairnous> but if I use sox to convert to wav and then run it through ffmpeg that should eliminate lame, yes?
[22:34:49 CEST] <furq> why are you running it through ffmpeg
[22:34:55 CEST] <durandal_1707> Exairnous: sox does same thing albeit slower than ffmpeg
[22:36:19 CEST] <furq> that was a rhetorical question but i notice the input is mp3
[22:36:37 CEST] <furq> lame isn't a decoder
[22:36:40 CEST] <furq> in spite of its name
[22:36:50 CEST] <furq> if you encode it to mp3 with ffmpeg then it's using libmp3lame
[22:36:59 CEST] <Exairnous> furq: ok, what is lame?
[22:37:05 CEST] <furq> it's an encoder
[22:37:10 CEST] <furq> that's why it's called "lame ain't an mp3 encoder"
[22:37:14 CEST] <furq> because it is an mp3 encoder
[22:37:38 CEST] <furq> i was sort of hoping they'd change the name to lime since the mp3 patents expired, but never mind
[22:38:38 CEST] <Exairnous> right, so if I change the file to a wav file enclosing pcm and work with that I won't be using lame, right?
[22:38:54 CEST] <furq> sure
[22:39:04 CEST] <furq> if you mean the output files
[22:39:11 CEST] <furq> it makes no difference what you do with the input file
[22:39:25 CEST] <Exairnous> right
[22:39:43 CEST] <bob_twinkles> hrm
[22:40:19 CEST] <bob_twinkles> if I'm willing to write some code against the ffmpeg API, would that execute faster than using the command line tool?
[22:40:45 CEST] <Exairnous> nope, same problem encoding to wav
[22:41:15 CEST] <furq> well yeah that might be an ffmpeg issue then
[22:41:19 CEST] <furq> or your ears are wrong
[22:41:20 CEST] <DHE> bob_twinkles: yes, but the codec work involved is probably going to be the real CPU bottleneck
[22:41:49 CEST] <Exairnous> one second I'll grab a screenshot of the waveforms :)
[22:42:12 CEST] <furq> bob_twinkles: maybe
[22:42:24 CEST] <furq> it depends on what you're doing
[22:42:38 CEST] <furq> it sounds like you're doing something pretty typical for ffmpeg, so i doubt it'll make an appreciable difference
[22:42:56 CEST] <bob_twinkles> I linked the filter chain above, but basically taking N input videos and aranging them in a grid
[22:43:07 CEST] <furq> are you encoding them
[22:43:58 CEST] <bob_twinkles> I'm not changing the pixel format (I think?) but I do have to reencode the video yes
[22:44:13 CEST] <furq> the slowdown is probably just caused by increasing complexity then
[22:44:16 CEST] <bob_twinkles> since it's merging 4 video streams in to 1
[22:44:30 CEST] <durandal_1707> furq: thers no bug in channelsplit, you are propagating gold earphones
[22:44:36 CEST] <furq> durandal_1707: don't blame me
[22:45:14 CEST] <furq> if he's splitting to pcm and it sounds wrong then either his ears are wrong or you should probably be interested
[22:45:32 CEST] <bob_twinkles> applying durandal_1707's suggestion and updating ffmpeg seems to have stabilized the FPS estimates
[22:45:33 CEST] <durandal_1707> bob_twinkles: merging is fast,its just memcpy
[22:46:06 CEST] <bob_twinkles> yeah, I assumed so
[22:46:20 CEST] <bob_twinkles> would doing the vstacks/hstacks have an impact or is that just being silly?
[22:46:28 CEST] <bob_twinkles> s/doing/changing the order of
[22:46:31 CEST] <furq> that's what he means by merging
[22:46:55 CEST] <bob_twinkles> I'm using hstack/vstack already
[22:47:00 CEST] <zerodefect> Trying out libavfilter through the C API.  I've created my first graph for video processing: buffer -> format -> colorspace -> fieldorder -> buffersink. Noticing an oddity with custom threading.
[22:47:02 CEST] <furq> it would have an impact if you moved them before/after the scale
[22:47:10 CEST] <furq> i assume it would have a negative impact if you moved them before but i don't really know
[22:47:50 CEST] <zerodefect> In AVFilterGraph, I set the 'execute' value to my function, but it never gets called.  I'm not sure if that's because none of the filters use threading?
[22:47:53 CEST] <durandal_1707> doing multiple stacks just uses more memcpy,but thats fast on modern machines, if you want ultra speed for merging  stuff in grids use shaders
[22:49:08 CEST] <Exairnous> Here is an image of the right channel before ffmpeg and after ffmpeg: https://pasteboard.co/bs5bZgCaF.jpg
[22:49:47 CEST] <bob_twinkles> that would involve having a GPU =/
[22:50:08 CEST] <bob_twinkles> moving the scale after the stacks is way slower lol
[22:50:54 CEST] <furq> you probably want to try outputting rawvideo and see how fast it is
[22:51:30 CEST] <furq> make sure that filtering is actually a noticeable bottleneck compared to encoding
[22:51:33 CEST] <durandal_1707> Exairnous: how you obtained original?
[22:52:15 CEST] <Exairnous> it's from a learning track from the Barbershop Harmony Society
[22:53:20 CEST] <durandal_1707> Exairnous: what tool you used to extract original channel?
[22:54:48 CEST] <bob_twinkles> it's about twice as fast with -c:v rawvideo
[22:55:27 CEST] <bob_twinkles> stil only ~8 FPS though
[22:55:53 CEST] <bob_twinkles> the machine this is running on is honestly not powerful enough to be doing serious video work so that's not entierly unexpected
[22:56:01 CEST] <durandal_1707> bob_twinkles: that needs disk, -use -f null -
[22:56:18 CEST] <bob_twinkles> the machine this is running on is honestly not powerful enough to be doing serious video work so that's not entierly unexpected
[22:56:19 CEST] <Exairnous> ffmpeg.  I only showed the right channel in the screenshot because it's the only one that has the problem.  What I'm doing is extracting the right channel from the stereo file  and outputing it to a stereo file of it's own
[22:56:36 CEST] <Exairnous> durandal_1707: ^^
[22:57:40 CEST] <durandal_1707> Exairnous: can you give original mp3 , upload somewhere?
[22:58:04 CEST] <Exairnous> the extra blip shown on the new file in the screenshot is actually there on the original, but it's so quiet you can't see it
[22:58:34 CEST] <Exairnous> durandal_1707: maybe, just a second
[23:00:12 CEST] <Exairnous> durandal_1707: here: http://www.barbershop.org/files/documents/freeandeasy/Keep%20the%20Whole%20World%20Singing%20-%20tenor.mp3
[23:01:48 CEST] <furq> i love #ffmpeg
[23:02:00 CEST] <furq> this must be the only channel on freenode where someone can get help with becoming the tenor in a barbershop quartet
[23:03:11 CEST] <Exairnous> furq: actually I'm doing this for a whole barbershop chorus :)
[23:03:57 CEST] <furq> nvm then there are loads of channels for that
[23:05:20 CEST] <Exairnous> the file is made so that the main part is in the right channel and the three other parts are mixed in the left channel, so I separate them out for easier learning so you get a, part none (to sing against), part only (to sing with), and the original right left mix (for a little of both)
[23:06:40 CEST] <Exairnous> furq: but I agree, #ffmpeg has lots of neat and diverse conversations :)
[23:10:30 CEST] <durandal_1707> i see two separate channels, they are like dual mono
[23:10:43 CEST] <Exairnous> yeah
[23:12:24 CEST] <Exairnous> if you split them using ffmpeg compare the right channel of the original and the output from ffmpeg
[23:12:46 CEST] <Exairnous> and you'll see what's in my screenshot
[23:13:18 CEST] Action: Exairnous really hopes this isn't happening only to me
[23:20:36 CEST] <durandal_1707> Exairnous: its ffmpeg decoder vs libmad one i think
[23:22:07 CEST] <Exairnous> it's a problem with the decoder?
[23:28:09 CEST] <Exairnous> durandal_1707: yep, looks like it's a problem with the decoder.  I just converted the mp3 to a wav using sox then used the wav as an input to ffmpeg and tested the output and everything was fine
[23:32:01 CEST] <furq> are you using a really old ffmpeg or something
[23:32:06 CEST] <furq> i didn't even know ffmpeg could use libmad
[23:32:32 CEST] <durandal_1707> furq: audacity uses libmad
[23:32:46 CEST] <furq> oh
[23:33:31 CEST] <Exairnous> furq: ffmpeg version 3.2.2 Copyright (c) 2000-2016 the FFmpeg developers built with gcc 6.2.1 (GCC) 20160830
[23:35:05 CEST] <Exairnous> furq: so it's slightly old, I haven't updated in a few months
[23:44:17 CEST] <Exairnous> durandal_1707: can I make ffmpeg us the libmad decoder?
[23:44:57 CEST] <durandal_1707> currently not
[23:45:33 CEST] <Exairnous> --enable-lib won't do it?
[00:00:00 CEST] --- Sun May 28 2017