[Ffmpeg-devel-irc] ffmpeg.log.20151228

[burek]

[00:03:29 CET] <furq> angeldorado: are you running ubuntu 14.04 or 15.10
[00:04:24 CET] <furq> if you're going to be using ffmpeg/libavxxx then it's better to use 15.10
[00:06:18 CET] <angeldorado> ubuntu 15.10?
[00:06:49 CET] <furq> whichever ubuntu derivative you want
[00:07:15 CET] <furq> for various convoluted reasons ffmpeg isn't in 14.04 or 14.10
[00:07:22 CET] <angeldorado> oh I see
[00:07:41 CET] <furq> if you're using 15.10 then you can install it from apt and save yourself a lot of bother
[00:07:54 CET] <angeldorado> well, good news is ubuntu studio 15.10
[00:07:59 CET] <furq> well yeah
[00:08:01 CET] <furq> apt-get install ffmpeg
[00:08:12 CET] <furq> unless you absolutely need ffmpeg 2.8
[00:08:21 CET] <angeldorado> well I have to install it, I have the previous version. but it does exist
[00:08:28 CET] <angeldorado> great stuff. what's the ppa?
[00:08:45 CET] <furq> i don't know of a ppa for 2.8
[00:08:49 CET] <furq> 2.7 is in the main repo
[00:09:39 CET] <furq> don't take that to mean that there isn't a ppa, i don't use ubuntu
[00:09:59 CET] <angeldorado> well I've been looking for hours
[00:10:11 CET] <angeldorado> and its not like any other thing ive installed in linux
[00:10:13 CET] <furq> do you specifically need 2.8
[00:10:46 CET] <angeldorado> im not sure exactly which version I need. I just know that almost every forum post I find has someone mentioning ffmpeg
[00:10:56 CET] <angeldorado> trying to solve some video/audio render problems im having
[00:11:13 CET] <furq> you might need to add universe to your /etc/apt/sources.list
[00:11:54 CET] <angeldorado> how do I do that?
[00:12:02 CET] <angeldorado> ill look google it
[00:22:34 CET] <rjp421> http://cdn.c3voc.de/s1_native_slides.webm
[02:53:58 CET] <salviadud> Hi
[02:54:04 CET] <salviadud> I want to share what I'm doing
[02:54:12 CET] <salviadud> which I feel is kind of odd
[02:54:25 CET] <salviadud> I got this compressed mp4 video in h264 format
[02:54:45 CET] <salviadud> The ratio is wrong, it should be 1080p
[02:54:54 CET] <salviadud> but it's 720x486
[02:55:04 CET] <salviadud> so, I'm stretching it and changing the aspect ratio
[02:56:20 CET] <salviadud> the command goes something like this: ffmpeg -i input.mp4 -s 1920x1080 -aspect 16:9 -b:v 3000k -vcodec libx264 -an out.mp4
[02:56:31 CET] <salviadud> now, it's doing it
[02:56:35 CET] <furq> you don't need to reencode to change the aspect ratio
[02:56:46 CET] <salviadud> but, at certain intervals
[02:56:50 CET] <salviadud> it asks me for a target
[02:57:00 CET] <salviadud> it's weird
[02:57:02 CET] <furq> you do if you want the output to be 1080p but that's pointless if the source is 480p
[02:57:25 CET] <salviadud> oh, but this is no ordinary video
[02:57:38 CET] <salviadud> It has something ciphered in it...
[02:57:40 CET] <pzich> it's extraordinary
[02:58:05 CET] <salviadud> It has to do with a native algorithm inside handbrake
[02:58:14 CET] <furq> it probably won't have something ciphered in it once you've reencoded it
[02:58:28 CET] <salviadud> why not furq?
[02:58:46 CET] <furq> i can't see how that would work
[02:59:09 CET] <salviadud> well, I might be wrong, and there's nothing there
[02:59:18 CET] <salviadud> but it is a misterious video, nonetheless
[02:59:43 CET] <salviadud> Now, my only doubt is why ffmpeg asks for a command
[02:59:46 CET] <salviadud> every now and then
[02:59:52 CET] <salviadud> it's version 2.3
[02:59:53 CET] <furq> anyway if you just want to change the AR then use -aspect 16:9 -c copy
[03:00:23 CET] <furq> and if you really think scaling it to 1080p will help anything then at least use -crf instead of -b:v
[03:00:33 CET] <salviadud> what's -crf ?
[03:00:50 CET] <furq> https://trac.ffmpeg.org/wiki/Encode/H.264#crf
[03:01:51 CET] <salviadud> ok, I get it. thanx for the crf
[03:02:02 CET] <salviadud> But still, is it common for ffmpeg to pause
[03:02:09 CET] <JodaZ> salviadud, what do you think stretching a 720x486 to 1080p will acomplish? you are upscaling...
[03:02:09 CET] <furq> it's not something i've ever seen
[03:02:11 CET] <salviadud> and ask for arguments, while doing a re-encode
[03:02:57 CET] <salviadud> another thing...
[03:04:30 CET] <salviadud> The output goes something like this: I get a constant line that repeats itself, in blue cyan, libx264@[somememoryaddress] and the usual frame by frame line
[03:04:48 CET] <salviadud> Is that common when reencoding h264?
[03:05:09 CET] <furq> that sounds like a warning which is being overwritten by the progressbar
[03:05:17 CET] <furq> it's probably something you can safely ignore
[03:05:33 CET] <furq> although maybe not if this is a mysterious james bond diagnosis murder video
[03:05:41 CET] <salviadud> Its xxx
[03:05:44 CET] <salviadud> haha
[03:05:52 CET] <salviadud> the original file is all wrong
[03:05:56 CET] <salviadud> straight from source
[03:06:01 CET] <salviadud> the dvd looks awful
[03:06:07 CET] <salviadud> It's like hacker xxx
[03:06:14 CET] <salviadud> you need to reencode it to see the original
[03:06:47 CET] <furq> i don't see how scaling would make a difference
[03:07:09 CET] <salviadud> It's an experiment of mine.
[03:07:22 CET] <salviadud> I lost ffmpeg on an old computer that has a particular processor
[03:07:31 CET] <salviadud> I'm running a gentoo live dvd on it
[03:07:43 CET] <salviadud> Had to compile ffmpeg 2.3 to get it working
[03:07:57 CET] <salviadud> I disabled the mmx flag
[03:08:13 CET] <salviadud> It's doing the most basic of processor optimizations.
[03:08:27 CET] <salviadud> That I belive is the key to decipher the vid
[03:08:51 CET] <salviadud> Soooo, I'll find out in a couple of days
[03:08:56 CET] <salviadud> since it's taking forever.
[03:09:07 CET] <salviadud> I will report back to you guys
[03:10:32 CET] <pzich> maybe use -t # to see if the first bit is looking right?
[03:33:17 CET] <Li> How to cut 1 minute from the begining of all video files in a given folder? I know for a single file it goes like ffmpeg -i input.filename -ss time output.filename
[03:33:53 CET] <pzich> you just run that for each file?
[03:33:53 CET] <c_14> for file in *; do ffmpeg -i <blargh>; done
[03:34:00 CET] <c_14> or any other scripting language
[03:52:43 CET] <Li> c_14: what is <blargh> how to replace it?
[03:53:13 CET] <c_14> -i "$file" -ss time "output$file"
[06:51:41 CET] <thomedy> hello
[06:52:29 CET] <thomedy> im not entirely sure how to get this very specific encoding
[06:52:32 CET] <thomedy> plese help
[06:52:44 CET] <thomedy> h264 main level 3 avc1 and aac
[06:55:18 CET] <pinPoint> ffmpeg -i input -c:v libx264 -profile:v baseline -level 3.0 -b:v 1000k output.mp4
[06:55:40 CET] <pinPoint> https://trac.ffmpeg.org/wiki/Encode/H.264
[06:58:23 CET] <rjp421> pick a width/height/fps within the level/profile youre encoding for. baseline to play on real old devices or main otherwise
[07:02:14 CET] <rjp421> https://sonnati.wordpress.com/2011/08/30/ffmpeg-%E2%80%93-the-swiss-army-knife-of-internet-streaming-%E2%80%93-part-iv/
[07:02:19 CET] <thomedy> thats funny i thought it would have been alot harder but i found it in my serach finally
[07:03:57 CET] <thomedy> now ijust have to figure out how to do the width height and fps
[07:04:49 CET] <pinPoint> https://trac.ffmpeg.org/wiki/Scaling%20(resizing)%20with%20ffmpeg
[07:04:58 CET] <pinPoint> thomedy: ^^^
[07:05:26 CET] <pinPoint> hell, https://trac.ffmpeg.org/wiki
[07:06:39 CET] <thomedy> oh my god i did it
[07:07:22 CET] <thomedy> im less concerned about the ffmpeg conversion but i needed it to accomplish my goal
[07:07:27 CET] <thomedy> and it just happened
[07:07:39 CET] <pinPoint> which is
[07:09:31 CET] <thomedy> i am building a website where i can scroll through the video in forwrd or reverse based on the window position
[07:09:43 CET] <thomedy> its not all that technical honestly
[07:09:49 CET] <thomedy> but i really liked the desing
[07:09:54 CET] <thomedy> design
[07:10:28 CET] <thomedy> now the more that i think about it ... it may have been less about the encode and more about the frame size
[07:10:50 CET] <thomedy> whatever i have it now
[07:10:55 CET] <thomedy> thank you
[07:11:20 CET] <pinPoint> ya
[07:26:32 CET] <sati> hi anybody there which can help me
[07:27:08 CET] <sati> I want use ffmpeg for video recording from analog camera
[07:28:00 CET] <sati> I want know for more than 8 recording (h.264) which hardware is compatible for ffmpeg
[07:36:36 CET] <rjp421> sati v4l2-ctl -d /dev/video0 --list-formats-ext --list-fields
[07:36:57 CET] <rjp421> or something along those lines
[07:40:04 CET] <sati> rjp421 : no Actually our pc i5 intel have limitation , that processor reach 100% usages on 3 h.264 video encoding
[07:40:14 CET] <sati> I want more than 8
[07:40:51 CET] <sati> so there have any another hardware solution which have low cost
[07:41:10 CET] <sati> because intel processor is a high cost pc
[07:41:45 CET] <rjp421> sati keep the lowest bitrate, fps and resolution you can stand
[07:44:14 CET] <rjp421> which os?
[07:44:39 CET] <sati> rjp421 : yes bro I do alot of R & D
[07:44:49 CET] <sati> that is linux Os
[07:44:52 CET] <sati> Ubuntu
[07:45:51 CET] <sati> rjp421 : now I think , we can use external hardware for video encoding
[07:45:57 CET] <sati> is it possible
[07:46:23 CET] <rjp421> ffmpeg -f v4l2 -list_formats all -i /dev/videoX
[07:46:55 CET] <rjp421> you may get lucky and have cams that can output h264
[07:47:18 CET] <rjp421> like raspberry pi cams
[07:47:45 CET] <rjp421> that will save cpu transcoding
[07:51:42 CET] <sati> rjp421 : No rasberry pi is very small board that device can not handle video encoding and decoding more than 1 channel
[07:53:04 CET] <sati> I mean to say that there have any external h.264 chip set available which is supported by ffmpeg and that will be attached with my pc
[07:53:25 CET] <rjp421> i mean h264 is one of the cameras output formats, and doesnt need to be transcoded..  i think youre going to have to skimp on some of the quality to get 8 smooth h264 streams
[07:54:16 CET] <sati> no I have analog camera . these camera producing raw data
[07:54:21 CET] <rjp421> especially if theyre all usb devices, yikes
[07:54:44 CET] <sati> like ??
[07:55:00 CET] <sati> or any device serial no if you know please
[07:55:06 CET] <rjp421> like your usb bud being overwhelmed
[07:55:09 CET] <rjp421> bus
[07:57:23 CET] <sati> rjp421 : bro I don't understand what you want to say
[07:58:46 CET] <sati> I want know there have a dedicated hardware which can perform encoding operation under ffmpeg  externally and my cpu consumption decreases
[08:04:50 CET] <flux> USB2.0 should be plenty for even 10 h264-encoded video streams? USB3.x even more so.
[08:06:53 CET] <flux> sati, you may be looking for some professional equipment to do hardware-based h264-encoding capture from 8 channels.. perhaps a device that talks IP as-is already, no need for ffmpeg?
[08:08:31 CET] <flux> sati, just did an ebay search on h264 encoder, and on the paper this seems applicable: http://www.ebay.com/itm/150772072175 (you would need two)
[08:08:51 CET] <flux> I hope though those 8 video streams don't need to be synchronized, though :)
[08:09:39 CET] <sati> flux : if you know intel have quick sync technology which is paid and that is operated by Our ffmpeg  , I want like that  device , but make a low cost device
[10:43:38 CET] <Disturbed1_away> gm, does anyone know of a decent guide for compiling ffmpeg with nvenc?
[10:54:14 CET] <saste_> Disturbed1_away, you only need the nvenc header from the NVidia SDK, then you build ffmpeg the usual way
[10:54:25 CET] <saste_> with --enable-nvenc
[10:55:44 CET] <saste> Disturbed1_away, https://trac.ffmpeg.org/wiki/HWAccelIntro#NVENC
[10:59:36 CET] <Disturbed1_away> ka, i'll give it a go over.... thought i was being smart buying a tesla s870... come to learn after purchase finding files for a old legecy device are impossible thru nvidia site....
[11:18:20 CET] <jonascj> Hi all. I am doing h264 two pass encoding and I want to scale my video at the same time (1080p to 720p).
[11:18:40 CET] <jonascj> Do I need "-vf scale=-1:720" in both "-pass 1" and "-pass 2" command?
[11:20:02 CET] <jonascj> if pass1 is just going through the file looking for statistics to use in pass 2 then it seems "stupid" to also scale it in that pass. So any scaling could be ignored when "-pass 1" is there, but I am not sure ...
[11:53:26 CET] <relaxed> jonascj: both
[11:59:37 CET] <Li> c_14: I tried this but it didn't work ... for file in *; do ffmpeg -t "$file" -ss 18 -vcodec "output$file"; done
[12:00:59 CET] <jonascj> relaxed, both?
[12:03:53 CET] <relaxed> jonascj: you need to scale on both passes
[12:05:02 CET] <jcay> Li, -t is time AFAIK , you need -i before filename
[12:18:42 CET] <Li> jcay: yes I changed it but it still not working
[12:18:49 CET] <Li> for file in *; do ffmpeg -i "$file" -ss 00:00:18 -vcodec "output$file"; done
[12:22:50 CET] <jcay> you didnt give the vcodec
[12:24:37 CET] <rjp421> http://streaming.media.ccc.de/32c3/hallg/#schedule
[12:44:51 CET] <jonascj> relaxed, alright, that is also what I am doing
[12:46:12 CET] <jonascj> after some time of -pass 1 it begins with "Invalid UE golomb code=25.0"
[13:01:22 CET] <jonascj> The bitrate it reports in "-pass 1" for libx264 encoding, (frame=384754 fbps=58 q=25 time=03:04:03.23 bitrate=4300 kbits/s speed=2.4x), what bitrate is that?
[13:02:00 CET] <jonascj> In my two-pass I am targeting 4100kbit/s video and 250kbit/s audio, which is 4350kbit/s, is that it?
[13:22:15 CET] <rjp421> http://cdn.c3voc.de/s2_native_sd.webm
[13:23:10 CET] <dystopia_> bitrate will fluctuate even with a target
[13:23:21 CET] <dystopia_> 2pass encoding to x264 is kinda pointless
[13:23:35 CET] <dystopia_> use single pass and correct crf for content type
[13:28:29 CET] <Li> How to delete 5 seconds from the end of the file without having to substraction operation to calculate duration -t
[13:29:20 CET] <Li> in other words an equivalent for the switch -ss to delete from file's end
[13:42:01 CET] <rjp421> damnit wheres their irc... someone should remind them of rtmfp and its ability to handle the udp packet loss
[13:50:44 CET] <Li> huh
[14:23:10 CET] <Disturbed1_away> does the latest build of ffmpeg not support nvenc 4.0 ? (
[14:26:11 CET] <Disturbed1_away> trying to compile ffmpeg with cuda kit 5 & nvenc 4 (and x264) so i can use legecy nvidia S870 hardware...
[14:35:16 CET] <Disturbed1_away> if so, what version of ffmpeg does support nvenc4 >?
[14:42:40 CET] <saste> Disturbed1_away, from the check in configure, NVENC API version 4 or older is not supported
[14:43:13 CET] <saste> basically, ffmpeg only supports nvenc API >= 5
[15:00:11 CET] <Disturbed1_away> so from what i have figured out nvenc5 isn't compatible with the driver 340.xx required to run the s870... only nvenc4 is....
[15:01:00 CET] <Disturbed1_away> so i'm kinda forced to use a legecy copy of ffmpeg if i'm ever going to use s870 to encode with
[15:03:27 CET] <Disturbed1_away> what i need to know is at what ffmpeg version did nvenc4 get dropped so i can compile a version before that...
[16:01:57 CET] <Disturbed1_away> ugh
[16:08:49 CET] <jonascj> dystopia_, two-pass is recommended when you have a target file size as far as the documentation goes.
[16:27:30 CET] <LEOXD> hi
[16:28:55 CET] <waressearcher2> LEOXD: hallo
[16:29:14 CET] <waressearcher2> LEOXD: wie geht's es dir ?
[16:30:39 CET] <LEOXD> waressearcher2: Kenn ich dich?
[16:31:19 CET] <saste> Disturbed1_away, see commit b08caa87c35a768ec0abb16b1e99c3a85f1df28e
[16:31:50 CET] <waressearcher2> LEOXD: nein, ich glaube nicht
[16:33:08 CET] <saste> Disturbed1_away, I think it requires nvenc >= 5 since its first commit
[16:33:44 CET] <LEOXD> anyways, I was wondering whether dale's info from Jan 9 about ffvp9 are still correct https://groups.google.com/a/chromium.org/forum/#!msg/chromium-dev/2Dz5Q5gCTXQ/Up6eD7ti3VYJ
[16:36:51 CET] <durandal_170> lot have changed in ffvp9
[16:39:38 CET] <LEOXD> durandal_170: yeah, I thought so. But I can't find anything useful regarding what exactly changed.
[16:40:15 CET] <durandal_170> see git log
[16:40:50 CET] <LEOXD> link?
[16:42:16 CET] <durandal_170> source.ffmpeg.org
[16:42:38 CET] <durandal_170> locate vp9 in libavcodec
[16:44:12 CET] <LEOXD> thanks
[17:31:49 CET] <jonascj> stopping and resuming an ffmpeg encoding, is that possible?
[17:32:08 CET] <c_14> As long as you don't reboot, sure
[17:32:49 CET] <c_14> Use SIGSTOP and SIGCONT
[17:33:13 CET] <jonascj> ah yes, how would sigstop followed by a suspend to disk work?
[17:33:24 CET] <jonascj> okay I suppose
[17:33:33 CET] <c_14> Should be fine afaik
[17:34:23 CET] <jonascj> and of course this job is running on windows :(
[17:34:36 CET] <jonascj> thanks, i'll go look at windows sigstop sigcont capabilities
[17:44:06 CET] <kepstin> you should be able to just suspend to disk with ffmpeg running, the os will take care of stopping/resuming everything.
[17:50:53 CET] <jonascj> kepstin, good point.
[18:23:56 CET] <JodaZ> hmm, theres these hacks to suspend a single process to disk on linux
[18:24:50 CET] <JodaZ> otherwise, i am kind of very interested in ways to stop/start encoding at a specific point in a file because that would make HLS (encoding and caching) much more elegant
[18:25:59 CET] <JodaZ> but last time i tried there was no way to encode from a specific frame (and have the audio not somehow generate half filled chunks and stuff)
[19:15:52 CET] <yongyung> So I'm trying to encode a video so that sony vegas can read it, I've used this options for different sources already and vegas could always read them no problem. But somehow it can't with this source, even though the source shouldn't even influence it right? Vegas thinks it's audio-only and the audio just sounds like random data, it's definitely not reading the actual audio stream. If I use a different container it doesn't recognize the
[19:15:52 CET] <yongyung> file at all. Any ideas what I could try? The exact command line is
[19:15:55 CET] <yongyung> ffmpeg -ss 00:02:44 -i "clip01.mp4" -t 30 -c:a copy -c:v libx264 -qp 16 -preset veryfast -tune fastdecode -movflags +faststart test.mp4
[19:17:20 CET] <fritsch> depends on the source
[19:17:25 CET] <fritsch> post a mediainfo of the source
[19:17:36 CET] <fritsch> it might be interlaced
[19:18:12 CET] <yongyung> Like this? http://codepad.org/tkiiG0Oq
[19:18:45 CET] <fritsch> its 1080p60
[19:18:55 CET] <fritsch> not sure the device can keep up with 60 progressive frame
[19:18:57 CET] <fritsch> s
[19:19:30 CET] <yongyung> I've edited 1080p180 with qp=12 before so I don't think it should be a problem
[19:19:38 CET] <yongyung> Or what do you mean
[19:19:41 CET] <ChocolateArmpits> transcode the audio too
[19:20:02 CET] <ChocolateArmpits> maybe it expects aac and then goes crazy when there's mp3 there
[19:28:23 CET] <yongyung> ChocolateArmpits: It works! But vegas only likes mp4, and mp4 doesn't support pcm, and I don't want to get another audio reencoding step... well let me try to just strip the audio
[19:28:45 CET] <fritsch> yongyung: that sentence makes no sense
[19:29:00 CET] <yongyung> fritsch: ?
[19:29:13 CET] <fritsch> that sentence has nothing to with with what ChocolateArmpits said
[19:29:14 CET] <ChocolateArmpits> yongyung: just use high bitrate if you want transparency
[19:30:42 CET] <Li> what is the wrong with this syntax and how to correct it? ffmpeg -ss 00:00:10 -i file1.webm -c copy file2.webm
[19:31:00 CET] <Li> I'm trying to cut 10 second intro from audo file
[19:31:03 CET] <Li> audio*
[19:32:08 CET] <ChocolateArmpits> Li: do you want intro removed or preserved ?
[19:32:25 CET] <Li> ChocolateArmpits: nope I want to get rid of it
[19:33:58 CET] <yongyung> Well I don't have libfdk_aac and the aac encoder doesn't seem that great.. but it doesn't matter, stripping audio seems to work
[19:34:43 CET] <ChocolateArmpits> yongyung: aac is pretty good now, people say, at least than libvo definitely
[19:35:00 CET] <yongyung> ChocolateArmpits: Any idea why I get tons of "past duration too large" yellow messages when I try to encode the file without -ss, but with -ss 00:00:02 it works just fine? Is the first frame somehow not correctly aligned in the source file or something?
[19:35:46 CET] <ChocolateArmpits> yongyung:  is this the same transcode?
[19:35:52 CET] <yongyung> Yeah
[19:35:58 CET] <yongyung> with -an
[19:36:47 CET] <ChocolateArmpits> Probably some problem with initial timestamps, the warning message is very generic
[19:39:50 CET] <yongyung> Hmm well good to know that I can avoid it at least. It comes up so much the i/o actually significally slows encoding lol
[19:40:43 CET] <ChocolateArmpits> You can hide those messages using -loglevel error I think
[19:41:14 CET] <ChocolateArmpits> or using setpts filter to completely reset timestamps for each frame
[19:45:53 CET] <yongyung> Yeah the problem with loglevel is that it also hides all the information output since it's technically "less important". The other method I never tried, if -ss doesn't work some day I'll remember that
[20:13:45 CET] <gabi__> hi, i have an audio file (adts mp3) and i am wondering how i could transform that to a regular mp3 file, because my car can not play adts, just regular mp3. can someone tell me how to do that with ffmpeg?
[20:15:20 CET] <gabi__> even when i convert it to .wav and then reconvert to mp3 with sox/lame/ffmpeg etc. i always get a mp3 in adts
[20:15:33 CET] <gabi__> Audio file with ID3 version 2.3.0, contains: MPEG ADTS, layer III, v1, 192 kbps, 44.1 kHz, Stereo
[20:32:59 CET] <yongyung> gabi__: Can you post the ffprobe output of your file to pastebin and the link here?
[20:43:54 CET] <gabi__> yongyung, jep, one second
[20:44:27 CET] <gabi__> yongyung, http://pastebin.com/k4Urr9bA
[20:44:53 CET] <BtbN> That's a perfectly normal mp3 file.
[20:45:19 CET] <gabi__> hm ok, i used file before and that reports adts
[20:45:43 CET] <gabi__> Adele - Hello.mp3: Audio file with ID3 version 2.3.0, contains: MPEG ADTS, layer III, v1, 192 kbps, 44.1 kHz, Stereo
[20:49:05 CET] <gabi__> BtbN, so is file reporting something wrong_
[20:49:14 CET] <furq> file reports that for every mp3 i just tried
[20:49:28 CET] <gabi__> ah ok :) so its a feature of file...
[20:54:00 CET] <furq> yeah it looks like libmagic always refers to mp3 as "MPEG ADTS, layer III"
[21:00:32 CET] <yongyung> gabi__: So does your car play the file?
[21:06:46 CET] <gabi__> yongyung, no
[21:06:57 CET] <gabi__> unfortunately not, i thought maybe thats the reason
[21:07:13 CET] <furq> is it vbr
[21:07:22 CET] <furq> some very old hardware mp3 players will only play cbr
[21:07:28 CET] <gabi__> how can i check that?
[21:09:56 CET] <furq> mediainfo will tell you
[21:12:29 CET] <furq> http://sprunge.us/YfXI
[21:13:07 CET] <yongyung> gabi__: Did you buy the music? If so, are you able to download a .flac or .wav version of the file or do you have a CD? Because in that case you could just encode the music yourself and try out what works on your car. Re-encoding usually doesn't sound as good, though I guess it's not the end of the world for car speakers
[21:13:41 CET] <gabi__> no i downloaded it from youtube with a online converter
[21:15:03 CET] <gabi__> http://pastebin.com/jLWaZEZg
[21:15:19 CET] <gabi__> so i guess thats CBR, as it says constant bit rate
[21:15:23 CET] <yongyung> gabi__: Alright if you want to go the youtube route (instead of.. ehem more shady ways to get the music), use a program or website to just download the raw video file, and convert that with ffmpeg. What you should try in your car is
[21:15:52 CET] <yongyung> ffmpeg -i fromyoutube.mp4 -c:a libmp3lame -c:b 320k test1.mp3
[21:16:05 CET] <furq> use youtube-dl
[21:16:08 CET] <yongyung> ffmpeg -i fromyoutube.mp4 -c:a libmp3lame -c:b 128k test2.mp3
[21:16:27 CET] <yongyung> ffmpeg -i fromyoutube.mp4 -a:a libmp3lame -aq 0 test3.mp3
[21:16:34 CET] <furq> -q:a 0
[21:16:57 CET] <yongyung> ffmpeg -i fromyoutube.mp4 -c:a copy test4.aac
[21:17:00 CET] <furq> also -b:a, not -c:b
[21:17:05 CET] <gabi__> i have a document which says that the radio should work with regular mp3 files with 96...256k, if i remember correctly
[21:17:10 CET] <furq> and please use .m4a, not .aac
[21:17:11 CET] <yongyung> uhh yeah mit b:a LOL
[21:17:40 CET] <furq> with hardware players there could be a million things that it doesn't like
[21:18:09 CET] <yongyung> gabi__: Well, then try ffmpeg -i fromyoutube.mp4 -c:a libmp3lame -b:a 256k test1.mp3
[21:18:16 CET] <furq> i take it there's no line-in port for you to plug a real mp3 player into
[21:18:44 CET] <gabi__> no, there is no line in
[21:18:52 CET] <gabi__> only usb
[21:19:38 CET] <gabi__> i will generate a mp3 from the raw file and see if that works. if not, i will be back :)
[21:20:02 CET] <furq> i can't help but notice "Mode extension: MS Stereo" in that mediainfo
[21:20:06 CET] <yongyung> gabi__: Good and the parameters are -c:a for codec selection and -b:a for bit rate selection, I mistyped that a couple of times
[21:20:18 CET] <furq> i'm not sure what that is but it sure sounds like something which could break an elderly player
[21:39:21 CET] <gabi__> ok, i have the new .mp3 file without ms stereo extension. will test this tomorrow. many thanks :)
[21:39:36 CET] <gabi__> bye
[21:55:04 CET] <BtbN> A lot of players also play AAC, which you can download straight from youtube, without any transcoding
[23:13:03 CET] <jonascj> "Invalid UE golomb code=27" while encoding x264. It started around half-way through the encoding (framewise). Anything to be worried about? Not much can be seen on the mailing list.
[23:13:57 CET] <jonascj> Mostly people / developers talk about polluting the log files with it, and worry that certain checks pertaining to this UE golomb code may slow down x264 encoding which they don't want because it is a popular encoding proceduire...
[00:00:00 CET] --- Tue Dec 29 2015