[Ffmpeg-devel-irc] ffmpeg.log.20160525

[burek]

[01:39:55 CEST] <Aksen> Hi all, I'm trying to merge 8 mono wav files into a single 5.1+LtRt interleaved wav (SMPTE 302M format). I'm getting errors but when I google I can't find any examples of others using this encoder, so I'm not sure what my next step is. http://pastebin.com/exXQbhge
[01:43:46 CEST] <rrva> in mpeg-ts streams, for a certain GOP, is there anything else than PTS to correlate which audio packets belong to a certain GOP of video frames (packets)?
[02:05:46 CEST] <JerryT> Any suggestions on how to go about getting 5 frames from a video starting after the first minute of footage and then every 1/5th of the movie.  So say the movie is 5 minutes it would get 5 frames from the first minute, 2nd, 3rd, 4th, 5th minute.  Then put those images into 1 long image sprite horizontally.  And ideally have the images be high quality frames?
[02:26:08 CEST] <dsl42> JerryT, https://trac.ffmpeg.org/wiki/Create%20a%20thumbnail%20image%20every%20X%20seconds%20of%20the%20video should be the answer to the first part of your questions. no clue how to put them into one image with ffmpeg afterwards, i probably would just use gimp. would not be surprised if it would be possible with ffmpeg though
[02:27:30 CEST] <dsl42> ok..
[02:32:18 CEST] <llogan> Aksen: "s302m codec not supported in WAVE format"
[02:35:13 CEST] <llogan> try another output format. such as mpegts (.ts,.m2t,.m2ts,.mts, etc)
[02:35:33 CEST] <Aksen> thanks, i'll give it a shot
[02:36:38 CEST] <Aksen> the ts extension worked. I'll check to see if it's workable for my client
[03:31:20 CEST] <okanaganrusty> i'm playing with ffmpeg trying to overlay two videos and an audio stream but keep getting an error - Error initializing complex filters, can someone please let me know what i'm doing wrong, or point me in the direction to solve the filter issue i have (http://pastebin.com/Lqk7GqKs)
[03:34:51 CEST] <llogan> okanaganrusty: you're missing the complete output
[03:35:50 CEST] <okanaganrusty> the output is at http://pastebin.com/5Ht36HZH
[03:47:18 CEST] <llogan> okanaganrusty: didn't get to look closely, but use vstack instead of nullsrc+overlay. it's easier and faster
[03:47:26 CEST] <llogan> i mean hstack
[03:47:35 CEST] <okanaganrusty> okay, i'll look at that.  thanks
[03:47:48 CEST] <llogan> and you can use anullsrc or aevalsrc instead of /dev/zero if you prefer
[03:48:12 CEST] <llogan> got to go
[04:46:29 CEST] <asdfffdsa> Using ffmpeg to reduce the size of a video file.  I got about 90% done but then I ran out of space on the HD, causing ffmpeg to stop (process still running and taking up CPU, but the output file size remains the same so i'm not sure what it's doing).  Is there any way I can make ffmpeg "resume" without killing and restarting the command?
[04:46:45 CEST] <asdfffdsa> (I've sinced cleared up some more HD space)
[04:51:41 CEST] <asdfffdsa> well i have to restart but i will be back later
[04:58:31 CEST] <CoJaBo> hate it when that happens lol
[05:27:41 CEST] <sheenobu> I'm able to use ffmpeg to take webcam input and x11grab input, overlay them (yay!), and output to a dummy v4l2 device. The thing I want to do, after that, is be able to turn off the inputs and keep the dummy v4l2 device up.
[05:27:59 CEST] <sheenobu> gstreamer might an option
[05:29:49 CEST] <sheenobu> as is some sort of RTMP aware thing. (webcam, x11grab) -> smart rmtp server that can detect and switch inputs -> rtmp client -> v4l2 dummy device
[07:04:23 CEST] <kristofferR> I'm trying to record a web stream, but sometimes it's unstable so ffmpeg thinks the streams end prematurely. How can I make ffmpeg wait longer for incoming packets?
[07:58:06 CEST] <arbi> Hi! Can anyone recommend a good video composer for linux that can handle mp4 files? (Cinelerra has problems with mp4)
[08:04:40 CEST] <Betablocker> https://alternativeto.net/software/cinelerra/
[08:05:00 CEST] <Betablocker> try windows movie maker
[08:05:06 CEST] <Betablocker> kidding&.
[08:06:15 CEST] <arbi> :)
[08:26:42 CEST] <Me4502> Hi, I'm trying to use ffmpeg to convert a single PNG into an MP4 of a length of 1 second
[08:27:00 CEST] <Me4502> This is what I'm using now - ffmpeg -f image2 -framerate 1 -i input.png output.mp4
[08:27:19 CEST] <Me4502> However, the video doesn't play. In details it has the correct dimensions and length however
[08:27:38 CEST] <Me4502> If I press play in VLC - the screen flickers to the dimensions and goes back - no image appears
[08:34:11 CEST] <Me4502> Here's the video file if anyone wants it http://s000.tinyupload.com/index.php?file_id=75587366096982834330
[08:34:19 CEST] <Me4502> I re-did it with 10 seconds, just incase 1 was too short
[08:42:12 CEST] <thebombzen> Me4502: VLC automatically goes black when the video ends
[08:42:25 CEST] <Me4502> I also tried with the windows 10 photos/videos app
[08:42:28 CEST] <thebombzen> try playing it with ffplay. which freezes when the video ends.
[08:42:47 CEST] <thebombzen> also try setting the video codec. to something like mpeg4
[08:43:10 CEST] <Me4502> Is codec between input and output?
[08:43:40 CEST] <Me4502> whelp.. Setting codec fixed it
[08:43:41 CEST] <Me4502> Thanks
[08:44:41 CEST] <thebombzen> if you don't set the codec it will use libx264 profile High 4:4:4 Predictive
[08:44:49 CEST] <thebombzen> which windows probably can't read cause it's dumb
[08:44:58 CEST] <Me4502> Oh, I presumed it'd be able to
[08:45:01 CEST] <Me4502> Makes sense
[08:45:22 CEST] <thebombzen> vlc should display the video for one second, but it's probably programmed to be "done" when the last frame plays
[08:45:34 CEST] <Me4502> Possibly
[08:45:38 CEST] <thebombzen> and when it's done it goes to the black screen.
[08:45:39 CEST] <Me4502> And seeing as there is only one frame
[08:46:11 CEST] <Me4502> Actually yeah - VLC works if I have 2 frames, doesn't if I have 1
[08:46:31 CEST] <thebombzen> there you go
[08:46:51 CEST] <Me4502> Another quick question - https://trac.ffmpeg.org/wiki/Concatenate#demuxer
[08:47:02 CEST] <Me4502> Is that mylist.txt absolute paths only?
[08:47:05 CEST] <Me4502> Or can I use relative?
[08:52:05 CEST] <klaxa> as you can tell from the examples it's relative
[08:52:20 CEST] <Me4502> Okay - thanks
[09:23:26 CEST] <arbi> quit
[09:54:49 CEST] <prelude2004c> hey everyone, good day
[09:55:10 CEST] <prelude2004c> question.. if i have ffmpeg -f output .. how do i pipe this to send the hls files to a webdav server instead of a local output
[09:55:28 CEST] <prelude2004c> can't seem to find reliable info online
[10:14:20 CEST] <sheenobu> prelude2004c, i wouldn't be surprised if ffmpeg couldn't output webdav. are you on a *nix?
[10:14:32 CEST] <sheenobu> rather..output /to/ webdav
[10:14:35 CEST] <prelude2004c> yup nix
[10:14:52 CEST] <sheenobu> then your best bet may be to mount the webdav mount and just output to a file on the mount
[10:15:44 CEST] <sheenobu> prelude2004c, actually https://www.ffmpeg.org/ffmpeg-protocols.html#http
[10:18:32 CEST] <prelude2004c> i have a remote mount setup but i am having issues where sometimes if the remote mount lags it messes up the output
[10:18:53 CEST] <prelude2004c> so i am trying to just push it to an http:// where it can retransmit and keep a queue
[10:21:03 CEST] <BtbN> Run ffmpeg on the target machine instead.
[10:21:03 CEST] <sheenobu> ahh
[10:22:20 CEST] <sheenobu> i can actually use webdav as an output but i'm not seeing the output
[10:23:07 CEST] <sheenobu> you can even use pipe as an output to wget -X POST prelude2004c ?
[10:23:40 CEST] <prelude2004c> yup.. just have to learn how to do it
[10:23:52 CEST] <sheenobu> figuring it myself
[10:23:53 CEST] <BtbN> Why not just run nginx-rtmp on the target machine?
[10:23:56 CEST] <prelude2004c> will it execute the segment_delete as well?
[10:28:23 CEST] <prelude2004c> maybe i run inotify wait.. and then use curl to grab data and push to webdav ?
[10:28:26 CEST] <prelude2004c> would that be easier ?
[10:30:24 CEST] <prelude2004c> maybe i am over thinking it
[10:31:26 CEST] <BtbN> nginx-rtmp can output hls and dash from an rtmp input, or run ffmpeg to do the conversion.
[10:31:39 CEST] <BtbN> No need for strange "curl webdav streaming" hacks.
[10:45:31 CEST] <prelude2004c> this is odd.
[10:45:41 CEST] <prelude2004c> 016/05/25 04:44:41 [error] 18637#0: *28484444510 open() "/var/www/citshd/citshd2M69.ts" failed (2: No such file or directory), client: 10.0.200.219, server: , request: "POST /webdav/citshd/citshd2M69.ts HTTP/1.1", host: "10.0.200.2"
[10:45:53 CEST] <sheenobu> you'll want a PUT
[10:46:10 CEST] <prelude2004c> so i pushed http:// and webdav is running on nginx.. ( actually pushing data from other location ) .. odd how its erroring out
[10:46:22 CEST] <prelude2004c> you mean request PUT instead of POST ?
[10:46:26 CEST] <sheenobu> yes
[10:46:30 CEST] <sheenobu> PUT creates files in webdav
[10:46:32 CEST] <prelude2004c> how do i change that ?
[10:46:37 CEST] <prelude2004c> ffmpeg have an option ?
[10:46:56 CEST] <prelude2004c> i am simply outputing to http://server/webdav/$files
[10:47:22 CEST] <sheenobu> ahh
[10:48:27 CEST] <sheenobu> http://ffmpeg.org/ffmpeg-all.html#http
[10:48:29 CEST] <sheenobu> there is a 'method' option
[10:48:40 CEST] <prelude2004c> yup reading ont hat.. but how to use it :)
[10:48:44 CEST] <prelude2004c> on the output i guess
[10:49:27 CEST] <sheenobu> -a copy -f type -method PUT http://...
[10:49:37 CEST] <prelude2004c> checking
[10:50:02 CEST] <Elirips> Hello. I'm using ffmpeg to have fun with some rtsp-streams. Question: When will ffmpeg disconnect from an rtsp stream if no data is received? never? 60 seconds? 10 seconds? is there any param to fine-tune when ffmepg will disconnect? I know about -stimeout - is there anyhing else?
[10:51:21 CEST] <prelude2004c> wow, look at that :)
[11:42:24 CEST] <sheenobu> prelude2004c, working?
[11:42:38 CEST] <prelude2004c> yup..
[11:42:39 CEST] <prelude2004c> thank you
[11:42:44 CEST] <sheenobu> great, i'm glad i could help
[11:43:01 CEST] <prelude2004c> thank you very much.. just tweaking some scripts for automation.. but looks good
[11:43:57 CEST] <sheenobu> great
[12:04:22 CEST] <sheenobu> hrm. this is a problem if your WebDAV server is under a non-valid TLS cert
[12:36:40 CEST] <wallbroken> ffmpeg -i input -vf "hflip,vflip,scale=1024:-2"
[12:36:44 CEST] <wallbroken> ffmpeg -i input -vf "transpose=1,scale=576:-2"
[12:36:51 CEST] <wallbroken> is there some difference between those commands?
[12:41:36 CEST] <c_14> wallbroken: yes
[12:41:55 CEST] <wallbroken> the first rotates of 90 degrees
[12:41:58 CEST] <wallbroken> the seconds of 180
[12:42:07 CEST] <c_14> yes
[12:42:11 CEST] <c_14> Also, the scales are different
[12:42:12 CEST] <wallbroken> but whi the first uses hflip, vplip and the second uses transpose?
[12:42:38 CEST] <c_14> maybe hflip,vflip is faster than transpose,transpose
[12:44:22 CEST] <sheenobu> transpose=1 is 90 degrees, but scale=576:-2 rotates and scales?
[12:44:48 CEST] <sheenobu> to get 180?
[12:45:05 CEST] <c_14> no?
[12:45:09 CEST] <c_14> scale doesn't rotate
[12:45:12 CEST] <c_14> it scales
[12:45:38 CEST] <sheenobu> right. how does transpose=1,scale=576:-2 make 180?
[12:45:52 CEST] <c_14> it doesn't
[12:46:00 CEST] <c_14> hflip,vflip does 180
[12:46:15 CEST] <sheenobu> ah. wallbroken said it backwards then, and I was /very confused/
[12:46:27 CEST] <sheenobu> or my IRC client got the messages backwards.
[12:51:33 CEST] <wallbroken> c_14 the question is why is not used the same thing for both 90 and 180
[12:52:02 CEST] <c_14> 10:42 <c_14> maybe hflip,vflip is faster than transpose,transpose
[12:52:56 CEST] <wallbroken> and is not possible to make transpose 180 ?
[12:54:48 CEST] <furq> https://ffmpeg.org/ffmpeg-filters.html#transpose
[13:09:36 CEST] <misuzu__> I'm using hls stream as input for ffmpeg and server returns 403 for some reason. I found that ffmpeg sends Range header and this may be the reason it does not work. I found this on mail list: https://ffmpeg.org/pipermail/ffmpeg-cvslog/2012-October/055212.html and not sure how to use it. Is this patch already on upstream? If yes what command is used
[13:09:37 CEST] <misuzu__>  to disable range header?
[13:14:17 CEST] <c_14> If it's in cvslog it's upstream, set -seekable 0
[13:15:11 CEST] <misuzu__> c_14: thanks!
[13:15:23 CEST] <c_14> It should do that automatically though
[13:15:27 CEST] <c_14> What version of ffmpeg are you running?
[13:17:59 CEST] <misuzu__> 3.0.2
[13:18:08 CEST] <misuzu__> It works!
[13:18:16 CEST] <misuzu__> c_14: Thank you!
[14:02:11 CEST] <Hababam> hi, does any of yo guys know how to set the SO_RCVBUF within ffmpeg ?
[14:02:25 CEST] <Hababam> i have tried to edit the network core files, but it didn;t work
[14:04:03 CEST] <DHE> for udp, add "?buffer_size=1048576" or whatever value you want
[14:04:12 CEST] <DHE> tcp makes a distinction between send_buffer_size and recv_buffer_size
[14:07:46 CEST] <Hababam> hmm i will try that
[14:15:55 CEST] <Hababam> hmm, still getting the rtm missed X packets error
[14:16:02 CEST] <Hababam> rtp *
[14:17:04 CEST] <BtbN> if you're processing slower than you're getting data that will happen at some point.
[14:17:59 CEST] <DHE> current versions of ffmpeg do have a speed multiplier displayed. you should be getting something >= 1.00x
[14:18:44 CEST] <Hababam> hmm getting around 0.9 to 1
[14:19:20 CEST] <DHE> would you say it's going up or down?
[14:19:33 CEST] <DHE> at the end of the day 1.00 is where you want to be for live content
[14:19:53 CEST] <Hababam> it bouncing between them
[14:20:12 CEST] <DHE> that does not leave me with a warm fuzzy feeling...
[14:20:30 CEST] <Hababam> yeahh i know that feel -.- :/
[14:20:58 CEST] <DHE> Can you paste[bin] your commandline? go ahead and censor out any details that are private (rtmp URLs or passwords, etc) but leave all parameters present
[14:21:22 CEST] <Hababam> sure 1 sec
[14:22:28 CEST] <BtbN> usualy for a live stream you'd expect to start somewhere above x1.00, slowly approaching 1.00 from above.
[14:22:58 CEST] <DHE> typically, since ffmpeg will be buffering the input before starting and hence be able to process the buffer first.
[14:23:34 CEST] <Hababam> http://pastebin.com/raw/RJxEprrU
[14:24:10 CEST] <BtbN> speed=0.562x
[14:24:22 CEST] <BtbN> you're processing _way_ too slow.
[14:25:04 CEST] <DHE> computer over
[14:25:43 CEST] <Hababam> hmm,
[14:25:52 CEST] <Hababam> let me check on antoher server i have
[14:25:53 CEST] <DHE> it's just H264->raw which is a surprise. it's not CPU-bound, which means you're probably maxing out something else.
[14:25:59 CEST] <BtbN> it's not related to the server
[14:26:05 CEST] <BtbN> it's getting the stream just fine.
[14:26:06 CEST] <DHE> 1080p at 25fps in raw pixels is probably too much
[14:26:14 CEST] <BtbN> What you are doing with the stream locally is what is too slow.
[14:26:42 CEST] <Hababam> hmm, i need the stream out of a decklink card at 1080p 25 fps
[14:26:55 CEST] <BtbN> What is the input video?
[14:27:04 CEST] <BtbN> so, resolution and framerate
[14:27:22 CEST] <DHE> 1080p, 25fps apparently
[14:27:23 CEST] <Hababam> 720p 25
[14:27:27 CEST] <Hababam> input
[14:27:28 CEST] <BtbN> ah
[14:27:31 CEST] <BtbN> Stream #0:0: Video: h264 (Main), yuv420p(tv, bt709), 1280x720, 25 fps, 25 tbr, 90k tbn
[14:27:45 CEST] <BtbN> so you are upscaling and doing a pix_fmt conversion
[14:27:50 CEST] <DHE> ah, input
[14:27:55 CEST] <Hababam> yup
[14:27:59 CEST] <BtbN> and one that doesn't look too optimized to me
[14:28:04 CEST] <BtbN> so that could easily be your bottleneck
[14:28:39 CEST] <Hababam> i just trew it together, so no optmimizing yet
[14:29:10 CEST] <BtbN> I'm talking about yuv420p->uyvy422
[14:29:22 CEST] <DHE> software scaling alone is actually pretty good...
[14:29:24 CEST] <DHE> this on linux?
[14:29:29 CEST] <Hababam> yup ubuntu
[14:29:36 CEST] <Hababam> 16.04
[14:29:43 CEST] <DHE> custom build by the looks of it?
[14:29:49 CEST] <Hababam> yup
[14:30:04 CEST] <Hababam> cause in the standard build there's no decklnk support
[14:30:05 CEST] <DHE> if you run the "ffmpeg_g" version (with debug symbols) you can profile it running "perf top" (as root) in another terminal
[14:30:18 CEST] <DHE> press 'q' to quit perf
[14:31:08 CEST] <furq> you could maybe try using zscale instead of swscale
[14:31:33 CEST] <Hababam> ffmpeg_g how would i run that ?
[14:31:37 CEST] <Hababam> jsut ./ffmpeg_g
[14:31:39 CEST] <Hababam> ?
[14:31:49 CEST] <DHE> no, same ffmpeg commandline, just use the ffmpeg_g binary instead
[14:33:45 CEST] <Hababam> i'm not quite sure how to do that
[14:46:08 CEST] <DHE> ./ffmpeg_g -i rtsp://****@87.195.159.0:554/mode1 ....
[14:48:51 CEST] <Hababam> bash: ./ffmpeg_g: No such file or directory
[14:48:54 CEST] <Hababam> is the response
[14:57:38 CEST] <DHE> it's in the compilation directory
[15:03:31 CEST] Action: gmh is away: Idle
[15:04:29 CEST] Action: gmh is back (gone 00:00:05)
[15:06:10 CEST] <Elirips> Hello. I'm connection to rtsp stream over vpn using tcp. This works fine 2 out of 3 times, but sometimes I get '[rtsp @ 003fc100] Could not find codec parameters for stream 0 (Video: h264, 1 reference frame, none): unspecified size' and the hint to increase -analyzeduration. If set -analyzeduration and -probesize to 6000M, but still get the error sometimes. Any hints? see http://pastebin.com/pb4U30vM for the full command-line and ou
[15:07:54 CEST] <Elirips> Even more confusing is, that it seems if I omit '-stimeout 30000' it will never fail - but also stall forever if the vpn connection is not enabled
[15:10:31 CEST] <Elirips> also interesting, if watching traffic in wireshark, everytime when ffmpeg fails wireshark reports a retransmission of at least one package..
[16:21:04 CEST] <kauchman> Hello, I'm trying to live stream video from IP camera via ffmpeg and ffserver. Input from camera is rtp(h264) and i want to transcode it into webm so i can include it in web page. Now everything seems to work except that when i play the video, it plays from beggining of the feed file (that is when the ffmpeg command started). Is there a way to start playing "live" video? Thank you very much for any help.
[16:25:00 CEST] <DHE> well webm isn't intended for real-time streaming like that. more like for youtube
[16:27:25 CEST] <kauchman> Thank you. Is there maybe a more suited format? I tried Ogg(theora/vorbis) but it  wouldnt load video at all, only the loader kept spinning
[16:31:35 CEST] <kepstin> for live video in html5, you probably want to look into using DASH or HLS with a javascript player that handles the format.
[16:33:06 CEST] <kauchman> Thank you, I will look into that.
[16:56:19 CEST] <P4Titan> Hello all. I'd like to ask: what would the best way be to chose a specific C compiler to build ffmpeg?
[16:56:31 CEST] <c_14> The first one you run across that isn't broken.
[16:56:48 CEST] <P4Titan> Is there a way to specify directly
[16:56:54 CEST] <c_14> --cc= afaik
[16:56:55 CEST] <P4Titan> rather than rely on the $PATH variable
[16:57:00 CEST] <c_14> in configure
[16:57:08 CEST] <P4Titan> hmm, interesting
[16:57:11 CEST] <c_14> configure should probably also read the environment CC variable
[16:58:23 CEST] <P4Titan> Yes, --cc= does exist
[16:59:15 CEST] <furq> there's also --cross-prefix if you're cross-compiling
[17:00:07 CEST] <c_14> looks like it doesn't read CC
[17:02:05 CEST] <P4Titan> what does --cross-prefix take
[17:02:20 CEST] <P4Titan> a directory where gcc and other executables can be found?
[17:04:41 CEST] <DHE> more like a prefix to the gcc name. like arm-gnu-linux-[gcc]
[17:04:46 CEST] <DHE> (I made that up, not a real prefix)
[17:06:18 CEST] <furq> P4Titan: all the cross compiler toolchain binaries will have a prefix
[17:06:29 CEST] <furq> like i686-w64-mingw32
[17:06:34 CEST] <furq> if you're not cross compiling then don't use that
[17:06:54 CEST] <P4Titan> my case is a little specific
[17:07:37 CEST] <P4Titan> basically, I am using gcc and other tools that are specifically optimized to take C code and create a flash executable instead of a regular executable
[17:07:51 CEST] <P4Titan> I just wish to route the ffmpeg compilation through that gcc compiler
[17:08:16 CEST] <P4Titan> currently I set $PATH to include that gcc before the system one
[17:08:22 CEST] <P4Titan> but that is rather bad practice
[17:11:55 CEST] <vade> hi all. Im trying to get an AVFilterChain to do audio format conversion for me, and am having duration issues in my output stream. Im using the libavfilter api, using a buffer source filter  configured with my input streams time base, sample rate, sample format, an aformat filter set up to my output streams sample rate and format, and finally a buffer sink filter configured for output format, sample rate, and linked in that order. Looking t
[17:11:56 CEST] <vade> ffmpeg_filter.c, it *appears* unless I am reading it wrong, it actually goes buffer -> sink -> format ? and sink isnt configured at all except for channels?
[17:23:58 CEST] <andross> hey ho
[17:24:11 CEST] <andross> been a while but i decided to return to my project
[17:24:26 CEST] <andross> i just compiled ffmpeg on ubuntu#
[17:24:34 CEST] <andross> which files will i need for development on windows?
[17:25:12 CEST] <andross> i cross compiled of course
[17:29:00 CEST] <JEEB> andross: easiest is to do --prefix=/home/my_username/my_win_prefix
[17:29:03 CEST] <JEEB> and then do make install
[17:29:19 CEST] <JEEB> it will install all the "development" files into that prefix in the directory structure required
[17:30:22 CEST] <andross> i think i might have done something like that already
[17:30:27 CEST] <andross> im not sure, it was a while ago
[17:30:53 CEST] <andross> do you remember when i was here before JEEB ?
[17:33:43 CEST] <andross> JEEB: how do i do this prefix command, i tried it with configure but it said no such command
[17:34:01 CEST] <andross> i currently instaleld everything into the same folder by mistake(?) i guess
[17:34:13 CEST] <andross> although does that matter?
[17:34:48 CEST] <andross> i installed into the same folder as the source code
[17:35:27 CEST] <andross> (reminder im not a linux user so dont really know what im doing)
[17:35:33 CEST] <JEEB> andross: ./configure --prefix=/one/two/something
[17:35:40 CEST] <JEEB> when you configure your FFmpeg build
[17:35:52 CEST] <andross> ah forgot the ./ i guess
[17:35:58 CEST] <JEEB> uhh
[17:36:02 CEST] <JEEB> how did you compile FFmpeg then? :D
[17:36:06 CEST] <JEEB> or even more
[17:36:12 CEST] <JEEB> how did you succeed in cross-compilation?
[17:36:46 CEST] <andross> i was slightly more familiar with linux shell then
[17:36:49 CEST] <andross> but promptly forgot
[17:36:56 CEST] <andross> as i didnt use it
[17:37:08 CEST] <andross> and the full set of options was provided by someone here
[17:37:11 CEST] <JEEB> and of course you still require all the other configuration parameters
[17:37:19 CEST] <JEEB> you just add the prefix
[17:37:22 CEST] <andross> god damn id rather not go through all that again
[17:37:30 CEST] <andross> i already have it all compiled within the source code folder
[17:37:43 CEST] <JEEB> but that isn't in the form other things use it
[17:38:09 CEST] <JEEB> also you don't need to necessarily re-compile, you just need to re-configure to set the prefix :P
[17:38:30 CEST] <andross> i had a text document somewhere with the full configuration
[17:38:44 CEST] <JEEB> and then you do make/make install so that the required stuff gets done and you should have a prefix that you can just tar/zipball and move to a windows host for usage
[17:41:26 CEST] <andross> i found a text file i saved with a configuration: http://hastebin.com/tepigusaqa.lua
[17:41:30 CEST] <andross> but i dont believe it's the right one
[17:41:39 CEST] <andross> i mean it doesnt look like a windows build
[17:42:27 CEST] <andross> im sure i saved the right one in a text file somewhere
[17:42:47 CEST] <andross> is there a way i can see recent text files with gedit?
[17:44:13 CEST] <andross> okay i think i found it from config.log
[17:45:13 CEST] <andross> how doe sthis look JEEB:
[17:45:17 CEST] <andross> ./configure --cross-prefix=i686-w64-mingw32- --arch=i686 --target-os=mingw32 --disable-static --enable-shared --pkg-config='pkg-config --static' --extra-ldflags=-static-libgcc --enable-libmp3lame --prefix=/home/andross/ffmpeg-win
[17:47:35 CEST] <andross> what the hell, the command doesnt work any longer
[17:47:50 CEST] <andross> ERROR: libmp3lame >= 3.98.3 not found
[17:50:07 CEST] <andross> i feel like i remember getting that error before but i have no idea how i resolved it
[17:50:12 CEST] <andross> i def have lame installed
[17:53:59 CEST] <andross> anyone??
[17:59:01 CEST] <andross> JEEB :(
[17:59:31 CEST] <andross> do i need to do that path garbage
[18:07:58 CEST] <vizier> Hey i have ffmpeg on mac and i just want to setup a script so that whenever i download a file to a folder it will convert it and put it in a new folder
[18:08:32 CEST] <andross> this is driving me absolutely insane
[18:08:53 CEST] <andross> i cant work like this, i cant just stare at a screen completely dumbfounded
[18:08:54 CEST] <vizier> Can anyone help with this i dont know much about what im doing but i read that this was a fast and easy background process to setup so it automatically converts
[18:09:21 CEST] <thebombzen> andross
[18:09:37 CEST] <thebombzen> did you try --pkg-config='i686-w64-mingw32-pkg-config --static'
[18:10:08 CEST] <thebombzen> I think --pkg-config overrides --cross-prefix
[18:10:33 CEST] <thebombzen> also try --enable-cross-compile
[18:10:41 CEST] <thebombzen> so it doesn't try to execute the binaries.
[18:13:28 CEST] <thebombzen> well okay
[18:13:48 CEST] <thebombzen> vizier: does wget exist on a mac?
[18:13:54 CEST] <thebombzen> otherwise try using cURL
[18:14:05 CEST] <vizier> i used homebrew to install ffmpeg
[18:14:17 CEST] <furq> vizier: https://github.com/emcrisostomo/fswatch
[18:17:11 CEST] <vizier> i think i can use automator app like this guy suggested http://apple.stackexchange.com/questions/129929/automating-ffmpeg-using-automator-service
[18:17:34 CEST] <thebombzen> I automate stuff with cron. but that's just me. fswatch sounds like exactly what you're looking for.
[18:18:14 CEST] <vizier> ok i will look that that i didnt know this was going to be so hard lol
[18:19:06 CEST] <vizier> ffmpeg just converts right you cant move or rename stuff to a suggested format like sickrage or something right
[18:19:58 CEST] <vizier> i hardly know anything about what im doing im a biochemist
[18:23:18 CEST] <andross> hiii
[18:23:24 CEST] <andross> sorry someone highlighted me right before my pc crashed
[18:23:38 CEST] <andross> think his name began with a v?
[18:23:45 CEST] <andross> voltagex ?
[18:24:57 CEST] <andross> fuck my stupid fucking pc for crashing at the worst possible moment
[18:25:49 CEST] <andross> and fuck irssi for not keeping logs
[18:28:42 CEST] <andross> you there JEEB ?
[18:28:56 CEST] <andross> vizier ?
[18:29:58 CEST] <iive> <andross> i cant work like this, i cant just stare at a screen completely dumbfounded
[18:29:58 CEST] <iive> <vizier> Can anyone help with this i dont know much about what im doing but i read that this was a fast and easy background process to setup so it automatically converts
[18:30:17 CEST] <andross> hey iive
[18:30:27 CEST] <andross> someone highlighted me just after that, then my pc crashed
[18:30:54 CEST] <iive> <thebombzen> andross
[18:30:54 CEST] <iive> <thebombzen> did you try --pkg-config='i686-w64-mingw32-pkg-config --static'
[18:30:54 CEST] <iive> <thebombzen> I think --pkg-config overrides --cross-prefix
[18:30:54 CEST] <iive> <thebombzen> also try --enable-cross-compile
[18:30:54 CEST] <iive> <thebombzen> so it doesn't try to execute the binaries.
[18:31:08 CEST] <andross> hi thebombzen
[18:31:57 CEST] <thebombzen> HELLO
[18:31:58 CEST] <thebombzen> oops
[18:32:02 CEST] <thebombzen> hello. did you try that
[18:32:10 CEST] <andross> one second
[18:32:33 CEST] <furq> there's no need for mingw to have its own pkg-config
[18:32:42 CEST] <furq> just set PKG_CONFIG_PATH to point to your mingw libs
[18:32:51 CEST] <andross> im almost certain that command i pasted worked previously
[18:33:03 CEST] <andross> at least when building inside of the source code folder
[18:33:08 CEST] <thebombzen> vizier: ffmpeg takes an input file name and an output filename. you can let those be whatever you want. and in different directories
[18:33:10 CEST] <andross> the only difference is the prefix line
[18:33:19 CEST] <furq> like i said, you probably need to set PKG_CONFIG_PATH if it can't find libraries
[18:33:42 CEST] <andross> hey furq it may have been you that originally helped me with this
[18:33:46 CEST] <furq> probably
[18:34:06 CEST] <andross> can you remember roughly the PKG_CONFIG_PATH command i needed?
[18:34:25 CEST] <andross> because i dont know where the mingw libs even are
[18:34:29 CEST] <vizier> if someone would be rich if they made a gui where you can select an input folder, convert file inside it, apply metadata, rename it given a selected format, and move it to an output folder
[18:34:32 CEST] <furq> there should be a lib/pkgconfig directory somewhere which is full of .pc files
[18:34:37 CEST] <furq> presumably in the directory where you built ffmpeg
[18:34:51 CEST] <furq> `find . -name *.pc` if you can't find it
[18:35:04 CEST] <furq> then just set PKG_CONFIG_PATH to that directory before running configure
[18:35:27 CEST] <andross> and that will fix the lame not found issue?
[18:36:58 CEST] <furq> it should do
[18:38:26 CEST] <andross> here's the output when i run the find command furq: http://hastebin.com/tohohijiwe.js
[18:39:02 CEST] <furq> replace . with /
[18:40:19 CEST] <andross> furq: http://hastebin.com/jidizaxuku.avrasm
[18:40:23 CEST] <andross> thanks for helping btw
[18:42:43 CEST] <andross> is it /usr/local/lib/pkgconfig ?
[18:43:13 CEST] <andross> since it has stuff like libavcodec.pc
[18:43:32 CEST] <furq> nvm i just checked and lame doesn't bother installing a .pc
[18:43:36 CEST] <furq> search for lame.h instead
[18:44:20 CEST] <andross> furq: http://hastebin.com/caxodipahu.rb
[18:45:20 CEST] <furq> --extra-cflags=-I/home/andross/build/include --extra-ldflags=-L/home/andross/build/lib
[18:45:23 CEST] <furq> should do it
[18:45:40 CEST] <andross> on the configure command?
[18:45:42 CEST] <furq> yeah
[18:45:47 CEST] <andross> great thanks
[18:45:49 CEST] <furq> add the ldflags to your existing --extra-ldflags
[18:46:00 CEST] <furq> if you specify it twice the second one will override the first
[18:47:04 CEST] <furq> also thank you for reminding me that i still need to sort out this mingw build system ;_;
[18:47:17 CEST] <furq> stupid libgme
[18:48:08 CEST] <N_1ck> hi there, i have a n00bie question : i'm trying to do this : https://trac.ffmpeg.org/wiki/Create%20a%20mosaic%20out%20of%20several%20input%20videos However when i run the provided example script i end up with : " -i : command not found " ... I understand this is because the file is multi-line... what should i do ?
[18:48:25 CEST] <andross> furq: what's the syntax to add flags to the existing option?
[18:48:26 CEST] <furq> remove the newlines?
[18:48:32 CEST] <N_1ck> to keep it 'readable'
[18:48:48 CEST] <furq> --extra-ldflags="-static-libgcc -L/home/andross/build/lib"
[18:49:57 CEST] <andross> how does this look furq:
[18:50:00 CEST] <andross> ./configure --cross-prefix=i686-w64-mingw32- --arch=i686 --target-os=mingw32 --disable-static --enable-shared --pkg-config='pkg-config --static' --extra-ldflags="-static-libgcc -L/home/andross/build/lib" --extra-cflags=-I/home/andross/build/include --enable-libmp3lame --prefix=/home/andross/ffmpeg-win
[18:50:19 CEST] <furq> looks fine
[18:51:43 CEST] <andross> i think it worked :O
[18:51:58 CEST] <andross> now doing make install
[19:01:40 CEST] <andross> this is taking longer than i thought
[19:01:47 CEST] <andross> i think it is compiling...
[19:01:57 CEST] <furq> you can cancel it and run make -j4 (or however many cores you have)
[19:02:01 CEST] <furq> it'll pick up where it left off
[19:02:22 CEST] <andross> well this is a virtual machine and i think its just using one core
[19:02:45 CEST] <furq> i'm getting deja vu here
[19:03:07 CEST] <andross> by the way will this delete the compiled files in the source folder or will there be two copies now
[19:03:57 CEST] <furq> it'll overwrite them
[19:04:57 CEST] <andross> woah i think its doen
[19:05:02 CEST] <andross> that was quicker than i thought
[19:05:32 CEST] <andross> so i now have a 25.8mb folder called ffmpeg-win
[19:05:50 CEST] <andross> and i just copy this to my win pc?
[19:05:57 CEST] <furq> sure
[19:06:11 CEST] <furq> and then hope it doesn't segfault
[19:06:27 CEST] <furq> or more likely hope it doesn't complain about missing dependencies
[19:06:39 CEST] <furq> it should be fine though
[19:13:20 CEST] <andross> aaaright
[19:16:10 CEST] <andross> so i notice it builds both .a and .lib files
[19:16:15 CEST] <andross> which do i want to include?
[19:17:08 CEST] <furq> i don't think you want either if you're linking against the shared libs
[19:17:17 CEST] <furq> iirc mingw should spit out .dll.a import libraries
[19:17:26 CEST] <andross> yeah thats what i mean
[19:17:32 CEST] <andross> so i want to include the .dll.a files?
[19:17:34 CEST] <furq> yeah
[19:17:46 CEST] <furq> i think .lib is msvc's name for those, but i don't know why mingw would produce those
[19:28:35 CEST] <andross> hmm, seems using ffmpeg in Qt might not be as straightforward as i thought
[19:31:08 CEST] <andross> maybe its simpler if i use ffmpeg binaries and run them as an external process heh
[19:32:34 CEST] <furq> if you just need to encode pcm to mp3 then use lame directly
[19:32:38 CEST] <furq> the library, not the binary
[19:32:50 CEST] <andross> want to do aac as well
[19:32:58 CEST] <andross> and who knows what else
[19:40:52 CEST] <andross> how come the files in bin have these number suffixes
[19:40:58 CEST] <andross> e.g. avcodec-57.dll
[19:42:32 CEST] <JEEB> it's a way of telling which API version you're linking against
[19:42:50 CEST] <JEEB> every time a change in API is done that might require a re-linking the number gets +1'd
[19:43:01 CEST] <andross> ah
[19:43:57 CEST] <JEEB> on *nix usually development packages have f.ex. avcodec.so (read: like dll) which is a symbolic link to the versioned one
[19:44:22 CEST] <JEEB> so you can link against avcodec but the final binary will depend on avcodec-57 f.ex.
[19:45:02 CEST] <JEEB> (that way you can also have multiple versions of the library in your system in case of global library locations - on windows the DLLs tend to be located next to the binaries)
[20:10:40 CEST] <andross> god maybe ive made a huge mistake
[20:12:53 CEST] <grotesk> Hi @ all, maybe someone of you could help me, i'm trying to write an ios app in swift, to get the live stream from my garmin virbxe and stream it to youtube live, the camera is sending an rtsp stream with h264 video and aac audio.
[20:13:06 CEST] <grotesk> here is my output http://pastebin.com/uJKdWEaS
[20:13:14 CEST] <andross> furq: when i do the command av_register_all()
[20:13:17 CEST] <andross> it just crashes
[20:14:08 CEST] <andross> error code 0xc0000135
[20:25:30 CEST] <andross> furq: is there an easy way to verify the libraries built correctly? as right now im unsure if the problem is with the libraries or with QtCreator
[20:25:55 CEST] <furq> run the binary?
[20:26:36 CEST] <andross> interesting
[20:26:45 CEST] <andross> cant start because libmp3lame-0.dll is missing
[20:27:29 CEST] <furq> if you built a shared lib then you'll need that in the same directory
[20:27:39 CEST] <furq> if you built a static lib of lame then shrug
[20:27:51 CEST] <andross> yeah its share
[20:29:24 CEST] <andross> okay, the executable seems to work as in, it displays the help text
[20:30:18 CEST] <andross> i dont know how much i need to test, or if that alone confirms it built properly?
[20:30:53 CEST] <furq> if it gets that far then it should be fine
[20:31:04 CEST] <andross> ill be be back in a bit  need to get food
[20:48:07 CEST] <grotesk> Is there an easy way to figure out, how to use this command line as c code? ffmpeg -i rtsp://url -acode copy -vcode copy c:/name.flv
[21:09:31 CEST] <andross> alright im back
[21:10:00 CEST] <andross> furq: tbh i think i jumped the gun and im now way in over my head prob
[21:10:17 CEST] <andross> it somehow never even occured to me that ffmpeg is in c while my project is c++
[21:11:22 CEST] <JEEB> yes, the way to correctly use the C interfaces in C++ is noted in the FAQ
[21:13:54 CEST] <andross> well yeah im using extern "C" already
[21:14:07 CEST] <JEEB> yup
[21:14:26 CEST] <JEEB> I used to make a simple thing to decode the first frame of a matroska file and have explorer show it as a thumbnail
[21:14:46 CEST] <JEEB> the funny part was that it worked for a while on win7, until some random windows update made it start crashing
[21:14:58 CEST] <JEEB> this was circa 2013
[21:18:01 CEST] <kristian_on_linu> hi
[21:18:19 CEST] <JEEB> ohai
[21:18:40 CEST] <kristian_on_linu> I have an mp3 file that is 30 minutes long, made up of ... 12 individual pieces or so. I know how long each song is. Can I somehow cut it out?
[21:18:50 CEST] <kristian_on_linu> I mean, in a smart way ...
[21:19:25 CEST] <JEEB> ffmpeg -ss [seekpoint timestamp] -i input.mp3 -t [for how long] -c copy out.mp3
[21:19:40 CEST] <JEEB> also see ffmpeg-all.html for documentation on these options on ffmpeg.org
[21:23:42 CEST] <kristian_on_linu> so I have to do it X number of times?
[21:23:50 CEST] <JEEB> yes
[21:24:08 CEST] <JEEB> if you know the spots you can script
[21:25:37 CEST] <kristian_on_linu> thanks
[21:31:18 CEST] <andross> already having trouble with the example
[21:31:31 CEST] <andross> where is CODEC_ID_MP2?
[21:31:48 CEST] <andross> its not recognized
[21:32:24 CEST] <JEEB> ./include/libavcodec/avcodec.h:    AV_CODEC_ID_MP2
[21:32:34 CEST] <JEEB> you're looking at something rather old if you don't have the AV_ prefix
[21:32:41 CEST] <JEEB> I recommend looking at the docs/examples dir
[21:32:47 CEST] <JEEB> that has examples for demuxing, decoding etc
[21:32:56 CEST] <andross> hmm
[21:33:16 CEST] <andross> yeah i was looking here https://ffmpeg.org/doxygen/trunk/encoding-example_8c-source.html
[21:33:36 CEST] <furq> andross: if you don't need to decode a wide variety of formats then i'd probably just use libmp3lame/libfdk-aac directly
[21:33:56 CEST] <JEEB> uhh
[21:33:59 CEST] <furq> the only advantage of using libavcodec for this is if you need to decode anything other than pcm
[21:34:00 CEST] <JEEB> fdk-aac can't be distro'd
[21:34:40 CEST] <andross> furq: my understanding is that ffmpeg has lots of powerful wrapper functions that makes things easier or something
[21:34:41 CEST] <JEEB> unless he or his employer complies to the software license's binary redistribution part
[21:34:43 CEST] <andross> or so i was told
[21:35:17 CEST] <JEEB> so for AAC I'd almost just go with the latest internal libavcodec aac encoder unless you need HE-AAC)
[21:35:29 CEST] <JEEB> that way you're clearly in LGPL territory
[21:35:45 CEST] <furq> oh yeah i forgot lame is lgpl
[21:36:01 CEST] <JEEB> I actually just thought of that too
[21:36:15 CEST] <JEEB> yeah, it is
[21:37:02 CEST] <JEEB> andross: funny how that example still uses the non-AV-prefixed thing
[21:37:08 CEST] <furq> andross: it's simpler for a lot of things but for pcm->mp3 using lame directly would be much easier
[21:37:27 CEST] <JEEB> true
[21:37:36 CEST] <JEEB> if that's all you do, just using LAME's API could be simpler
[21:37:41 CEST] <furq> likewise with fdk but you can't distribute a binary that links to lame and fdk
[21:37:51 CEST] <furq> so you'd be stuck with libfaac which sucks
[21:37:54 CEST] <JEEB> uhh
[21:37:56 CEST] <furq> actually is that even gpl-compatible
[21:38:00 CEST] <JEEB> faac breaks its own license
[21:38:03 CEST] <JEEB> it's supposed to be GPL
[21:38:05 CEST] <furq> yeah i remembered something like that
[21:38:07 CEST] <drv> where did you find this doxygen link? it's from 2011 (not sure why it's still on the web server)
[21:38:09 CEST] <JEEB> but it's not really because it uses non-GPL-compatible code
[21:38:10 CEST] <furq> it'd be libvo-aacenc then
[21:38:14 CEST] <furq> which is even worse
[21:38:18 CEST] <JEEB> well, no
[21:38:25 CEST] <JEEB> in that case you build a limited libavcodec
[21:38:29 CEST] <JEEB> with the AAC encoder
[21:38:32 CEST] <JEEB> and use that
[21:38:35 CEST] <furq> i meant if you were doing this without using libav*
[21:38:47 CEST] <JEEB> well that makes sense if there are good alternatives to libavcodec
[21:38:51 CEST] <JEEB> like LAME itself
[21:38:52 CEST] <furq> exactly
[21:38:59 CEST] <furq> but since fdk is out, there's no good aac encoder
[21:39:04 CEST] <JEEB> uhh
[21:39:09 CEST] <furq> other than the ffmpeg one obv
[21:39:11 CEST] <JEEB> even the current lavc internal one?
[21:39:13 CEST] <JEEB> yes
[21:39:21 CEST] <JEEB> which is when it becomes the "alternative"
[21:39:39 CEST] <JEEB> you can still use LAME straight and lavc separately
[21:39:44 CEST] <JEEB> or use lavc for both
[21:39:49 CEST] <furq> you might as well use the same code path for both though
[21:39:57 CEST] <JEEB> yup
[21:41:05 CEST] <furq> i really ought to figure out why ffmpeg keeps stopping for no reason
[21:41:10 CEST] <furq> i bet it's mintty's fault
[21:41:11 CEST] <andross> drv: i just googled ffmpeg encoding example or something like that and it was the first thing that came up
[21:41:33 CEST] <drv> anyway, that is just a listing of a source file from the ffmpeg tree, so you can find an up-to-date copy there
[21:42:05 CEST] <JEEB> the github mirror has probably the best web view https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples
[22:35:23 CEST] <vade> what is the latest on hardware encode for ffmpeg via libavcodec API? is still mostly up to you to enable partcular paths in code, or does avcodec_encode_video2 handle this for you now-a-days if you pass right codec ?
[22:35:42 CEST] <andross> sorry im
[22:35:43 CEST] <andross> back
[22:35:50 CEST] <andross> got interrupted by a phonecall
[22:37:53 CEST] <andross> god damn it
[22:38:00 CEST] <andross> as soon as i try to use any function furq it just crashes
[22:38:22 CEST] <andross> during program startup
[22:38:43 CEST] <andross> im sure theres something wrong with the linker
[22:39:28 CEST] <andross> in my .pro i have unix|win32: LIBS += -L$$PWD/../../ffmpeg-win/lib/ -lavcodec -lavformat -lavutil
[22:39:42 CEST] <andross> INCLUDEPATH += $$PWD/../../ffmpeg-win/include
[22:57:55 CEST] <andross> was just missing the dlls in the build folder apparently
[23:06:53 CEST] <andross> all seems to be working now JEEB furq just need to work through the examples i guess, thanks again
[23:09:38 CEST] <andross> good way to learn c a bit more too i guess
[23:19:54 CEST] <DHE> vade: while there's a new API available, the avcodec_encode_video2 API is still working. and yes, for hardware support it's just a  matter of selecting the right encoder (eg: nvenc rather than libx264)
[23:20:15 CEST] <vade> DHE - any documentation on this new API?
[23:20:22 CEST] <vade> apologies if Ive missed it :)
[23:24:49 CEST] <DHE> vade: http://ffmpeg.org/doxygen/trunk/group__lavc__decoding.html#ga9395cb802a5febf1f00df31497779169  it's still pretty new though. it's not in the 3.0 release
[23:25:56 CEST] <vade> ah. interesting. Any options for encode too ?
[23:26:18 CEST] <DHE> avcodec_(send|receive)_(frame|packet) as required
[23:26:24 CEST] <vade> ah I see
[23:26:28 CEST] <DHE> but that's all API stuff. if you use the ffmpeg tool who cares?
[23:27:29 CEST] <vade> Well, im using the API stuff :)
[23:27:38 CEST] <DHE> well then
[23:30:11 CEST] <andross> welp
[23:30:17 CEST] <andross> the following code appears to cause a segfault:
[23:30:20 CEST] <andross> const enum ffmpeg::AVSampleFormat *p = codec->sample_fmts;
[23:30:47 CEST] <andross> ffmpeg:: is because i wrapped the header include directives within an ffmpeg namespace
[23:31:29 CEST] <vade> is codec null? have you registered your formats and codecs via register_all or whatever it is
[23:31:55 CEST] <DHE> that sounds like something easily identified with a debugger
[23:32:32 CEST] <andross> codec = ffmpeg::avcodec_find_encoder(ffmpeg::AV_CODEC_ID_MP2);
[23:33:26 CEST] <andross> you're right tho i havent done avcodec_register_all();
[23:34:05 CEST] <JEEB> which means nothing is initialized
[23:34:11 CEST] <andross> got it
[23:35:25 CEST] <andross> i guess that fixed it
[23:36:22 CEST] <vade> DHE: is h264_videotoolbox /videotoolboxenc.c scheuled for the next release?
[23:37:05 CEST] <vade> and btw JEEB I figured out my resampling / timing issues. Everything is pretty much dead on now. Wooop woop :)
[23:37:26 CEST] <andross> what is 's16' exactly
[23:37:36 CEST] <DHE> I would imagine so, but I don't do releases. I'm just another API user
[23:37:39 CEST] <vade> short 16 bit
[23:37:40 CEST] <JEEB> signed, 16
[23:37:42 CEST] <DHE> andross: signed
[23:37:44 CEST] <vade> oops sorry
[23:37:47 CEST] <andross> oh duh
[23:41:24 CEST] <andross> tbh i dont think i've ever played an mp2 file in my life lol
[23:41:27 CEST] <vade> is the new recieve / send packet frame functionally the same as encode video2 audio 4, but just a unified path for audio / video ?
[00:00:00 CEST] --- Thu May 26 2016