[Ffmpeg-devel-irc] ffmpeg.log.20160927

[burek]

[01:57:02 CEST] <TwinTailed> anyone available to help? Or all away?
[01:57:52 CEST] <llogan> TwinTailed: you should just ask your question and if someone knows they will answer but it may take a while just like in any IRC channel
[01:58:24 CEST] <TwinTailed> well ok
[01:59:26 CEST] <TwinTailed> I am using libav to demux and decode then mux and encode to another media file, the generated mp4 video however plays normally on VLC, but on Windows Media Player and Quicktime, the video is black screen, just the sound plays
[02:00:16 CEST] <llogan> as in libav I assume you mean the FFmpeg libav* libraries.
[02:26:40 CEST] <TwinTailed> llogan: yes ffmpeg's
[02:28:06 CEST] <llogan> TwinTailed: libav-user mailing list may be your best opion. not many library related answers here.
[02:29:05 CEST] <TwinTailed> llogan: the libav-user mailing list still haven't answered my last two emails more than a week ago
[02:29:17 CEST] <TwinTailed> I have fixed the problems I asked them from the help of people here
[02:29:53 CEST] <TwinTailed> you've got BtbN, JEEB, flux, kepstin
[02:30:45 CEST] <llogan> from what you told me I'll guess that your pixel format is not yuv420p so dumb players can't play it
[02:30:54 CEST] <llogan> but i haven't seen your code or any info about your output file
[03:40:12 CEST] <TwinTailed> llogan: I am copying the pixel format from the encoder
[03:40:52 CEST] <TwinTailed> if (encoder->pix_fmts)
[03:40:52 CEST] <TwinTailed>                     oCtx->pix_fmt = encoder->pix_fmts[0];
[03:40:52 CEST] <TwinTailed>                 else
[03:40:52 CEST] <TwinTailed>                     oCtx->pix_fmt = iCtx->pix_fmt;
[09:32:21 CEST] <the_gamer> is it possible to mix 2 mp3s as the new audio for my new video?
[09:33:24 CEST] <furq> the_gamer: https://ffmpeg.org/ffmpeg-filters.html#amix
[09:33:36 CEST] <the_gamer> thank you
[10:32:04 CEST] <mbarisa> hi guys, after I successfully compiled ffmpeg with --enable-openssl I still cannot read https links, I get this kind of issue:
[10:32:26 CEST] <mbarisa> CANNOT LINK EXECUTABLE: could not load library "libssl.so.1.0.0" needed by "/data/data/....
[10:33:07 CEST] <mbarisa> can anyone point me in direction how to pack libssl.so.1.0.0 in ffmpeg binary as well
[10:34:24 CEST] <mbarisa> http://pastebin.com/xiNJvgwa --> here is how the script looks like
[10:35:33 CEST] <mbarisa> and /usr/local/ssl/android-18/lib -> in this path libssl.so.1.0.0 exists, so I am really not sure what to do.. please point me in some direction
[10:36:21 CEST] <furq> mbarisa: --extra-ldflags=-static
[10:39:04 CEST] <mbarisa> I should remove my --extra-ldflags?
[10:39:16 CEST] <mbarisa> or just add this one extra
[10:40:21 CEST] <furq> add that one
[10:41:36 CEST] <mbarisa> giving it a try :)
[10:41:36 CEST] <mbarisa> thanks
[10:43:07 CEST] <mbarisa> now it says openssl not found :/
[10:46:00 CEST] <furq> do you have libssl.a in your lib path
[10:46:14 CEST] <mbarisa> yes
[10:46:20 CEST] <furq> check config.log
[10:47:55 CEST] <mbarisa> http://pastebin.com/cLsNKEmb
[10:47:56 CEST] <mbarisa> last lines
[10:50:04 CEST] <CruX|> this is build for android ?
[10:50:09 CEST] <mbarisa> yes :(
[10:50:10 CEST] <furq> well you're missing -ldl
[10:50:19 CEST] <furq> but for some reason that's trying to link against libws2_32 and libgdi32
[10:50:19 CEST] <CruX|> -lws2_32 is windows library
[10:50:22 CEST] <furq> which are both windows libraries
[10:51:16 CEST] <furq> if those are in libssl.pc or libcrypto.pc then something's fucked p
[10:51:34 CEST] <mbarisa> darn
[10:52:35 CEST] <furq> i'm guessing stuff from a mingw toolchain has got mixed in with your android toolchain
[10:54:18 CEST] <mbarisa> is there some simple way to compile ffmpeg with ssl for android D
[10:54:36 CEST] <mbarisa> i tried everything i could find on internetz
[10:55:08 CEST] <furq> recompile openssl?
[10:55:14 CEST] <mbarisa> i compiled it myself
[10:56:19 CEST] <furq> pastebin libcrypto.pc and libssl.pc
[10:56:29 CEST] <mbarisa> http://pastebin.com/VCvj2ctw
[10:56:58 CEST] <bencoh> .44
[10:57:00 CEST] <bencoh> woops
[10:57:16 CEST] <mbarisa> http://pastebin.com/erpSu4EU
[10:57:28 CEST] <furq> well shit
[10:57:30 CEST] <furq> those look fine to me
[10:58:01 CEST] <furq> pastebin your ffmpeg configure line i guess
[10:59:31 CEST] <mbarisa> http://pastebin.com/4f1MvUky
[10:59:40 CEST] <mbarisa> you can see on top i pasted my flags
[10:59:59 CEST] <mbarisa> LD and CF
[11:00:16 CEST] <furq> you can't specify --extra-ldflags twice
[11:00:40 CEST] <furq> --extra-ldflags="-static -L/usr/local/ssl/android-18/lib $LDFLAGS"
[11:01:10 CEST] <furq> also you should probably use `set -e` to quit on error instead of having `|| exit 1` everywhere
[11:01:56 CEST] <mbarisa> still same thing, i tried to add it in single line on front
[11:02:41 CEST] <mbarisa> what about those CFLAGS and LDFLAGS I have...
[11:02:45 CEST] <furq> check the configure line after variable expansion (at the top of config.log)
[11:02:51 CEST] <mbarisa> shoulsd I remove those
[11:02:52 CEST] <furq> make sure all those variables are set to what you expect them to be
[11:03:43 CEST] <furq> i've never cross-compiled for android so i have no idea whether those are ok
[11:03:51 CEST] <furq> but i can pretty confidently say they wouldn't cause a dependency on winsock
[11:04:18 CEST] <mbarisa> it worked fine for other libs
[11:04:31 CEST] <mbarisa> that were in before, i removed everything and left only ssl
[11:04:51 CEST] <furq> it looks an awful lot like the wrong pkg-config files are being read
[11:04:58 CEST] <furq> make sure your PKG_CONFIG_PATH etc are still correct
[11:05:41 CEST] <mbarisa> PKG_CONFIG_PATH is correct
[11:05:46 CEST] <mbarisa> if i remove -static it compiles
[11:05:53 CEST] <mbarisa> and doesn't say anything about openssl
[11:06:24 CEST] <furq> oh
[11:06:34 CEST] <furq> try --pkg-config="${2}/ffmpeg-pkg-config --static"
[11:07:06 CEST] <mbarisa> http://pastebin.com/wZwcWqAA
[11:08:28 CEST] <mbarisa> with this --static it is compiling
[11:09:00 CEST] <furq> hopefully the ws2_32 thing was a red herring
[11:09:54 CEST] <furq> also i'm pretty sure you can just use the system pkg-config as long as PKG_CONFIG_PATH is set
[11:10:22 CEST] <Kiicki> I thought it was not possible to change container without re-encoding the codec. I added a MP4 file to MKVmerge and the output became MKV in seconds?
[11:10:40 CEST] <furq> what made you think that wasn't possible
[11:11:01 CEST] <Kiicki> people always said that. I have really never found a fast way to change from mp4 to mkv
[11:11:15 CEST] <mbarisa> furq: thanks, waiting on compilation to finish, if it works we can celebrate :D
[11:11:20 CEST] <furq> ffmpeg -i foo.mp4 -c copy -map 0 bar.mkv
[11:12:06 CEST] <Kiicki> dude, you are just amazing with those codes.
[11:12:23 CEST] <Kiicki> well I guess it is possible then
[11:19:59 CEST] <flux> does FFmpeg support decoding into a texture with VDPAU?
[11:20:54 CEST] <flux> also, does it support video encoding from a texture using nvidia acceleration?
[11:21:52 CEST] <mbarisa> furq: compiled successfully, and it is still same issue
[11:26:49 CEST] <mbarisa> mbarisa at ubuntu:~/ffmpeg-android/build/armeabi-v7a/bin$ pkg-config --libs openssl
[11:26:49 CEST] <mbarisa> -L/usr/local/ssl/android-18/lib -lssl -lcrypto
[11:26:56 CEST] <mbarisa> i think this is correct
[11:27:24 CEST] <mbarisa> I have no idea why libssl.so.1.0.0 is missing
[11:29:01 CEST] <furq> it shouldn't be linked against anything with --extra-ldflags=-static
[11:31:13 CEST] <mbarisa> yeah but it doesn't work with --extra-ldflags=-static
[11:31:59 CEST] <mbarisa> then it cannot find openssl
[11:32:23 CEST] <mbarisa> and without that it can find it.. but it doesn't link libssl.so.1.0.0 inside
[11:32:44 CEST] <furq> did you use --extra-ldflags=-static and --pkg-config="pkg-config --static"
[11:34:13 CEST] <mbarisa> only --pkg-config
[11:34:21 CEST] <mbarisa> when i add static in --extra-ldflags
[11:34:29 CEST] <mbarisa> it breaks, says there is no openssl
[11:34:38 CEST] <mbarisa> when i don't add -static there it passes
[11:35:42 CEST] <mbarisa> and my configure has --enable-static
[11:36:38 CEST] <furq> yeah you need to use pkg-config --static with --extra-ldflags=-static
[11:36:57 CEST] <furq> enable-static is a different thing
[11:37:28 CEST] <mbarisa> but how to add --extra-ldflags=-static when it then says ERROR: openssl not found
[11:37:33 CEST] <mbarisa> so I cannot compile it
[11:38:26 CEST] <furq> --pkg-config="pkg-config --static"
[11:38:44 CEST] <mbarisa> that is set
[11:38:46 CEST] <mbarisa> and that is fine
[11:39:21 CEST] <mbarisa> http://pastebin.com/u71rwguU
[11:39:24 CEST] <mbarisa> this is how it looks
[11:39:39 CEST] <mbarisa> but with this ldflags=-static it is not compiling
[11:39:53 CEST] <furq> pastebin config.log again
[11:41:53 CEST] <mbarisa> I can say it is same
[11:41:54 CEST] <mbarisa> http://pastebin.com/JxKmswBq
[11:47:16 CEST] <furq> /home/mbarisa/ffmpeg-android/toolchain-android/bin/../lib/gcc/arm-linux-androideabi/4.9.x/../../../../arm-linux-androideabi/bin/ld: error: cannot find -ldl
[11:47:19 CEST] <furq> oh
[11:47:20 CEST] <furq> i guess you don't have libdl.a
[11:48:57 CEST] <furq> it's a bit hacky but you should be able to remove --extra-ldflags=-static if you remove libssl.so and libcrypto.so
[11:49:09 CEST] <furq> it'll link against the static libs if there are no shared libs available
[11:50:00 CEST] <mbarisa> where to remove them from?
[11:50:18 CEST] <mbarisa> from /usr/local/ssl/android-18/lib?
[11:50:23 CEST] <mbarisa> how can that work?
[11:51:10 CEST] <furq> it's probably better to rename them so you don't have to recompile them in future
[11:51:24 CEST] <furq> -lcrypto will link against libcrypto.a if there's no libcrypto.so
[11:51:26 CEST] <mbarisa> already removed :D but i have them copied in other places :)
[11:51:54 CEST] <mbarisa> compiling again
[11:52:13 CEST] <furq> you would normally do something like -Wl,-Bstatic -lcrypto -lssl -Wl,-Bdynamic -ldl
[11:52:38 CEST] <furq> but good luck getting ffmpeg's configure script to do that
[11:53:50 CEST] <mbarisa> thanks
[11:56:37 CEST] <mbarisa> why ssl is not included by default
[11:56:42 CEST] <mbarisa> :D
[12:03:40 CEST] <mbarisa> it is still not working ffs
[12:04:40 CEST] <mbarisa> it is in byte identical as old one
[12:04:53 CEST] <mbarisa> still asking for libssl.so.1.0.0
[12:05:28 CEST] <ronald245> hey everyone, im having some issues compiling ffmpeginterop make install
[12:05:28 CEST] <ronald245> make: *** No rule to make target 'install'.  Stop.
[12:05:42 CEST] <BtbN> what's ffmpeginterop?
[12:06:09 CEST] <ronald245> basically ffmpeg for windows
[12:06:22 CEST] <BtbN> Never heard of it, you'll have to ask whoever made it.
[12:07:36 CEST] <SouLShocK> ronald that's Microsoft, right?
[12:07:53 CEST] <SouLShocK> > This project is licensed from Microsoft under the Apache 2.0 License
[12:07:53 CEST] <SouLShocK> https://github.com/Microsoft/FFmpegInterop
[12:08:17 CEST] <BtbN> make install is usualy a bad idea anyway
[12:08:21 CEST] <ronaldp> yea, it´s m$
[12:08:39 CEST] <BtbN> on windows it's not even somewhat well defined.
[12:10:57 CEST] <ronaldp> figures, i´m trying to create a h264 streaming app in Uwp, but i´ll keep looking, thanks :)
[12:19:07 CEST] <ronaldp> join #WPDev
[12:19:25 CEST] <ronaldp> well this is embarrasing :)
[12:31:30 CEST] <Timster> Hi - could somebody please tell me what's wrong with this line to batch convert on Windows: http://pastebin.com/U6G7SiGq
[12:31:50 CEST] <Timster> FFmpeg says: Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
[12:35:43 CEST] <mbarisa> furq: i compiled it with flags you sent -Wl,-Bstatic -lcrypto -lssl -Wl,-Bdynamic -ldl and added -static in front as well, and it passed
[12:36:07 CEST] <mbarisa> but now i don't get error, but my command doesn't work now.. it worked before with http at least :D
[13:23:16 CEST] <TwinTailed>  I am using libav to demux and decode then mux and encode to another media file, the generated mp4 video however plays normally on VLC, but on Windows Media Player and Quicktime, the video is black screen, just the sound plays
[14:25:25 CEST] <kepstin> Timster: please pastebin the complete output. there should be an earlier log message saying exactly what the incorrect codec parameter is
[14:25:52 CEST] <kepstin> Timster: but the issue is probably that your .mov files use an audio codec that's not supported in .mp4, so the copy fails.
[14:26:58 CEST] <Timster> kepstin: sec
[14:28:18 CEST] <Timster> kepstin: http://pastebin.com/3ZAi4Ej8
[14:28:46 CEST] <kepstin> there you go: [mp4 @ 000000000262f340] Could not find tag for codec pcm_s16le in stream #1, codec not currently supported in container
[14:29:05 CEST] <kepstin> can't put pcm audio in mp4, you'll have to encode it to something, or stick with mov container
[14:30:20 CEST] <furq> if you want it to be lossless then your choices are mp4als and alac
[14:30:25 CEST] <kepstin> (andd your probably want to use -pix_fmt yuv420p to convert the color range and improve compatibility.
[14:31:31 CEST] <furq> [mp4 @ 000000000052f400] Could not find tag for codec alac in stream #0, codec not currently supported in container
[14:31:34 CEST] <furq> oh nice
[14:31:51 CEST] <furq> i guess that's to be expected but it begs the question of what container does support alac
[14:32:59 CEST] <kepstin> hmm, i think it only works with -f ipod?
[14:33:07 CEST] <furq> yeah it works with m4a
[14:33:12 CEST] <furq> how annoying
[14:33:21 CEST] <furq> or it would be annoying if anyone cared about alac
[14:33:28 CEST] <kepstin> it's an apple-specified extension, so it basically only works with apple stuff :/
[14:33:35 CEST] <kepstin> (and ffmpeg, of course)
[14:37:24 CEST] <Timster> kepstin: Thank you!
[14:41:14 CEST] <Timster> furq: Thanks also :)
[15:08:53 CEST] <nostrora> Hi! this is my command : ffmpeg -vn -i "$i" -codec:a libvorbis -qscale:a 5 "${i%.mp3}.ogg"
[15:09:08 CEST] <nostrora> but mp3 have cover (mjpeg) how to don't add it in .ogg ?
[15:09:23 CEST] <nostrora> So all ogg have theora and i don't want. Thanks
[15:10:42 CEST] <kepstin> nostrora: add the option '-map 0:a' and it will only copy the audio.
[15:11:14 CEST] <nostrora> kepstin: Thanks, and what about -vn ?
[15:12:14 CEST] <kepstin> oh. '-vn' would have worked, but you have to put it in the correct place. it's an "output" option
[15:12:24 CEST] <kepstin> so it goes before the output file, not before the input file
[15:13:40 CEST] <nostrora> kepstin: Thanks :) and how to set same quality as MP3 ?
[15:13:51 CEST] <nostrora> i don't want to set quality to 9 if mp3 only have 64kbit/s
[15:14:52 CEST] <kepstin> nostrora: the concept of "same quality" between two different codecs is hard to define
[15:15:14 CEST] <kepstin> lossy transcoding will always make quality lower, too
[15:15:35 CEST] <kepstin> so best to always encode to high quality, so it doesn't sound much worse than original
[15:15:44 CEST] <nostrora> kepstin: Yes i know :/ i want to entirely convert my Music directory to ogg (because is better,free etc.)
[15:16:04 CEST] <kepstin> nostrora: doing a transcode will not make it sound better - only worse :)
[15:16:10 CEST] <kepstin> nostrora: best to just leave it as-is.
[15:16:20 CEST] <nostrora> kepstin: Yes i know i just want ogg because is better
[15:16:51 CEST] <kepstin> it's only better if you do a new encode from lossless source
[15:16:52 CEST] <nostrora> kepstin: worse or same :p not only worse
[15:16:58 CEST] <kepstin> if you start with mp3, it's worse
[15:17:11 CEST] <nostrora> kepstin: not only worse, it may be same
[15:17:17 CEST] <kepstin> it will never be the same, unless you use really high quality, so the file is bigger.
[15:17:48 CEST] <nostrora> kepstin: Yes, if i use quality 9 on a 64kbit/s mp3 the quality is same
[15:17:53 CEST] <nostrora> but not worse
[15:19:28 CEST] <relaxed> I would keep them as mp3s
[15:19:59 CEST] <nostrora> relaxed: but mp3 is not free (as freedom) :/
[15:20:21 CEST] <nostrora> relaxed: why do you keep as mp3s ?
[15:20:31 CEST] <kepstin> all the good mp3 encoders/decoders are free software, and afaik, all the patents on it have expired by now?
[15:21:14 CEST] <relaxed> nostrora: like kepstin said, you're going to lose quality if you transcode to another lossy format
[15:21:27 CEST] <kepstin> in general, you should keep lossy formats in the original codec, because each time you convert, you lose quality
[15:21:45 CEST] <kepstin> (unless you convert to a lossless codec, of course, but that's just a waste of space)
[15:21:45 CEST] <nostrora> ok ok, thanks for information :)
[15:22:09 CEST] <relaxed> nostrora: but if you're hellbent on "freedom", you could use flac (which is lossless)
[15:22:24 CEST] <relaxed> but your files will be larger
[15:22:31 CEST] <nostrora> Do you know how to speed up ffmpeg conversion ? my cpu is not at 100% when i start the conversion :/
[15:24:54 CEST] <nostrora> i've set threads to 8 but speed is not better
[15:25:09 CEST] <kepstin> nostrora: depends exactly which conversion. audio codecs are all single-threaded, so you should probably just do multiple files at the same time (run multiple ffmpeg)
[15:25:31 CEST] <nostrora> kepstin: ok :) i'l doing this (multiple ffmpeg)
[15:26:01 CEST] <nostrora> So now i use MP3 for music i have already on my computer. Flac for store new music and ogg to convert the flac for my smartphone or car
[15:26:49 CEST] <furq> 14:20:31 ( kepstin) all the good mp3 encoders/decoders are free software, and afaik, all the patents on it have expired by now?
[15:26:53 CEST] <furq> december 2017 according to wikipedia
[15:27:12 CEST] <kepstin> hmm, so not yet, but soon. probably depends on jurisdiction too.
[15:27:13 CEST] <furq> only in the US of course
[15:27:25 CEST] <kepstin> nostrora: keep in mind that mp3->flac->ogg is the exact same quality loss as mp3->ogg, and you want to avoid mp3->flac->mp3 as well :)
[15:27:36 CEST] <nostrora> kepstin: yes i know this
[15:27:47 CEST] <furq> you want to avoid mp3 to anything really
[15:28:53 CEST] <nostrora> furq: Me yes, because vorbis are better or opus (better than vorbi) but i don't transcore my old mp3 because i loss quality
[15:31:16 CEST] <kepstin> I'm kind of annoyed that android 6 supports opus in ogg, but the media library doesn't support opus in ogg with the .opus extension (it doesn't index them) :/
[15:31:32 CEST] <furq> but yeah if you really have to transcode audio with ffmpeg then xargs is your friend
[15:32:01 CEST] <kepstin> i'm a crazy person with a shell script that runs a makefile to parallelize my ffmpeg audio transcodes ;)
[15:32:19 CEST] <furq> find . -name "*.mp3" printf "%f\0" | xargs -0 -n1 -P8 -I{} ffmpeg -i "{}" -c:a vorbis "{}".ogg
[15:33:18 CEST] <furq> then you get to enjoy 8 ffmpeg statuses in the same terminal
[15:33:19 CEST] <tontonth> generated makefiles for parallel processing is damn kwol !
[15:35:45 CEST] <nostrora> furq: my for is good ? or there is a better way to convert lot of file with ffmpeg ?
[15:35:57 CEST] <nostrora> for i in *.mp3; do ffmpeg -i "$i" -vn -c:a libvorbis -q:a 8 "${i%.mp3}.ogg"; done
[15:36:52 CEST] <furq> that'll do it one at a time
[15:36:58 CEST] <furq> xargs -P8 will run 8 processes at once
[15:37:15 CEST] <nostrora> i don't understand what do you mean
[15:37:26 CEST] <furq> the downside is that you can't strip the extension so you'll end up with "foo.mp3.ogg"
[15:38:38 CEST] <furq> nostrora: the for loop will encode each file one after the other
[15:38:48 CEST] <furq> and libvorbis is single threaded so that will only use one core
[15:38:55 CEST] <nostrora> furq: Yep
[15:39:20 CEST] <furq> xargs -P8 will run 8 at a time, so it'll use 8 cores
[15:39:42 CEST] <nostrora> furq: i have to execute xargs -P8 for .... ?
[15:39:55 CEST] <furq> 14:32:19 ( furq) find . -name "*.mp3" printf "%f\0" | xargs -0 -n1 -P8 -I{} ffmpeg -i "{}" -c:a vorbis "{}".ogg
[15:42:21 CEST] <nostrora> find: paths must precede expression: printf
[15:46:39 CEST] <nostrora> Bye bye vorbis, hello Opus :D
[15:46:53 CEST] <furq> oh
[15:46:55 CEST] <furq> it should be -printf
[15:48:40 CEST] <nostrora> What is the difference between -c:a libvorbis and -c:a vorbis
[15:50:36 CEST] <furq> vorbis is the internal encoder which is apparently a thing now
[15:50:42 CEST] <furq> libvorbis is presumably better
[15:52:31 CEST] <nostrora> >< bullshit
[16:20:16 CEST] <mbarisa> furq: workssssss now :0 thanks for your hints
[17:27:06 CEST] <TwinTailed>  I am using libav to demux and decode then mux and encode to another media file, the generated mp4 video however plays normally on VLC, but on Windows Media Player and Quicktime, the video is black screen, just the sound plays
[18:09:51 CEST] <jpacker1337> Hello, currently trying to alter the volume of my inputs in a amix call but I can't seem to get the syntax right with the outputs. Can someone help or guide me in the right direction?
[18:09:52 CEST] <jpacker1337> http://pastebin.com/JhQQGH0x
[18:10:44 CEST] <TwinTailed>  I am using libav to demux and decode then mux and encode to another media file, the generated mp4 video however plays normally on VLC, but on Windows Media Player and Quicktime, the video is black screen, just the sound plays
[18:26:52 CEST] <Chloe> TwinTailed: share code?
[18:27:51 CEST] <TwinTailed> Chloe: sure
[18:28:42 CEST] <jpacker1337> Trying to adjust audio levels of inputs using this code
[18:28:43 CEST] <jpacker1337> -ss 0.300 -i /data/user/0/me.rapchat.rapchat/cache/recording.m4a -i /data/user/0/me.rapchat.rapchat/cache/beats/05a0f287-4183-4ab0-8233-049b95ca38fa.m4a -filter_complex [0:a]volume=1.5[a0];[1:a]volume=1.5[a1]; [a0][a1]amix=inputs=2:duration=shortest:dropout_transition=3 -preset ultrafast /data/user/0/me.rapchat.rapchat/cache/rap.m4a
[18:29:02 CEST] <TwinTailed> Chloe: http://pastebin.com/u7TB2cDr
[18:45:52 CEST] <TwinTailed> Chloe: Anything wrong with my code?
[19:13:39 CEST] <TwinTailed> its weird this is happening to freenode
[19:14:59 CEST] <DHE> no, it's rare to go a month without this happening
[19:15:57 CEST] <TwinTailed> DHE: I used to use freenode excessively a few years ago and this never happened
[19:16:33 CEST] <TwinTailed> DHE: Have you got the time to help me right now?
[19:19:09 CEST] <DHE> you're doing libav stuff, so likely not
[19:20:18 CEST] <TwinTailed> DHE: you helped me before with libav as far as i remember :P
[19:33:59 CEST] <mbarisa_> again?
[19:36:56 CEST] <TwinTailed> that seems a DDoS attack not overload :P
[19:36:56 CEST] <TwinTailed> whats amusing is that I haven't been disconnected :P
[19:58:12 CEST] <mgpai_> When using ffplay, is it possible to control the window size both for audio and video using the same command? I am able to control the video size using -vf scale=iw*0.5:--1, but when playing audio file it increases the screen size.
[20:16:13 CEST] <ibisr> Hi there, I'm creating fragmented MP4 files and the box structure looks nice except I'm not getting a sidx box. Maybe I'm missing something? Here's the command I'm using: ffmpeg -i myInputMovie.mov -y -c:v libx264 -x264opts "keyint=24:min-keyint=24:noscenecut" -b:v 4800k -maxrate 4800k -vf "scale=1920:1080" -an -frag_duration 2000000 myFragmentedOutputVideo.mp4
[20:53:24 CEST] <jesseg> Hmm, ffplay is missing from the static linux 64 build
[20:53:43 CEST] <BtbN> complain to the creator of that build.
[20:55:40 CEST] <llogan> jesseg: i don't recall relaxed ever providing ffplay with his builds
[20:59:16 CEST] <jesseg> llogan, it was a couple years ago but I'm pretty sure I used to get ffplay from the statics but anyway it doesn't matter. :D
[20:59:27 CEST] <jesseg> I'll try compiling from source see if I can make t hat go
[21:04:38 CEST] <jesseg> Hmmm, ffplay was not built when I compiled either...
[21:05:05 CEST] <jesseg> maybe I need SDL2 support
[21:10:50 CEST] <jesseg> HA! compiled SDL2, rebuilt ffmpeg, now I have ffplay :D
[21:12:22 CEST] <JEEB> jesseg: yeah the SDL1 stuff was recently rewritten for SDL2
[21:13:08 CEST] <jesseg> JEEB, ahh OK.
[21:26:54 CEST] <jesseg> JEEB, have you done anything with melt? It compiles for me but can't play video files even though ffmpeg can read them
[21:28:23 CEST] <JEEB_> jesseg: nope
[21:31:35 CEST] <jesseg> ty
[21:36:24 CEST] <llogan> jesseg: melt probably requires sdl_image for playback
[21:37:53 CEST] <llogan> let me restate that: try sdl1
[21:39:25 CEST] <llogan> or, if you're lazy, use the static build that comes with shotcut. execute it via the script instead of directly executng the binary
[21:47:01 CEST] <vklimkov> hello. I want to extract part of mp3 file into separate file programmatically. I.e. I have linked ffmpeg static lib against my C++ code and I want to do "ffmpeg -i in.mp3 -ss $start -c copy -t $dur out.mp3"
[21:48:50 CEST] <vklimkov> I can think of decoding, selecting only frames in the time range I want and encoding back, but there should be better way, right?
[21:51:13 CEST] <persina> Can someone explain the difference between the input and output framerates for practical purposes?
[21:51:37 CEST] <DHE> while it might be different for MP3, you should be able to ignore any AVPackets whose PTS and duration is outside  your interesting range and only start decoding when you hit the time you want
[21:51:55 CEST] <jesseg> persina, can you explain some more background?
[21:52:06 CEST] <DHE> persina: overriding the input framerate is the same as modifying the input file to change its framerate field. setting the output framerate will drop and duplicate frames to maintain the requested framerate.
[21:52:20 CEST] <DHE> intentionally playing with values will allow you to turn a long movie into a short time-lapse for example
[21:54:06 CEST] <vklimkov> DHE: Thanks for reply. but I don't want to decode. I mean if I decode and than encode back, there will be quality degradation. Is there a way to avoid it?
[21:54:10 CEST] <persina>  DHE "overriding the input framerate is the same as modifying the input file to change its framerate field" I'm not suing an input file so I'm not sure what the :framerate field" does.
[21:55:03 CEST] <jesseg> persina, what is your video source?
[21:55:04 CEST] <persina> jesseg: I'm just trying to figure out the difference between them so I can understand which of them (perhaps both?) I need to change.
[21:55:19 CEST] <llogan> nobody knows what you're referring to
[21:55:27 CEST] <persina> jesseg: I have a bunch of png's in a dir and trying to convert them to a mp4.
[21:55:46 CEST] <jesseg> persina, well, what DHE said is the answer, but evidentally you have some special circumstances which is why I'm asking about more background info
[21:56:16 CEST] <llogan> the image file demuxer uses -framerate as an input option. the default is -framerate 25.
[21:56:22 CEST] <jesseg> persina, ok just specify the input frame rate for how fast you want the pics to play
[21:56:40 CEST] <jesseg> persina, so if you want the pictures to play at 10 a second, set the frame rate to 10
[21:57:21 CEST] <llogan> if you add the output option -r then it will drop/duplicate frames if it differs from your input -framerate
[21:58:05 CEST] <jesseg> persina, basically the input frame rate flag tells it how many fps it's supposed to be played at. The output frame rate flag then inserts extra frames or deletes some frames to change the frame rate above or below the input frame rate.
[21:59:13 CEST] <persina> jesseg: Why would someone want to overwrite the input framerate using -r instead of just changing the -framerate flag?
[22:00:16 CEST] <llogan> some players don't like -framerate 1 so you'll have to make a standard output framerate such as -r 25.
[22:01:34 CEST] <jesseg> persina, lots of reasons, depending on the situation. You can make a video play faster or slower without actually adding or deleting frames, by simply changing the fps flag.
[22:03:19 CEST] <jesseg> persina, for example, let's say you have a 30fps video and you want to make it play at 1fps so it becomes slow motion: Changing the fps flag just slows it down without increasing file size. However, if you change the frame rate on the output, then it'll take every frame and duplicate it 30 times. so you'll end up with 30 times the number of frames.
[22:05:22 CEST] <jesseg> Likewise, if you have a 1fps timelapse type video and you want to speed it up, you just change the input frame rate to 30, causing it to play every frame but 30 times faster, which is better than duplicating each output frame 30 times to bring the fps up to 30, which wouldn't even speed up the video
[22:06:16 CEST] <jesseg> persina, ultimately, adjusting the input fps changes the play speed without changing the frame count or filesize, while the output frame rate setting changes the number of frames and filesize without changing the play rate.
[22:07:26 CEST] <persina> jesseg: OK, that makes sense! +1 for the clear explanation :)
[22:08:05 CEST] <jesseg> :D
[22:08:18 CEST] <TwinTailed> Hi, I am transcoding using Libav (FFMPEG), but the output H264 MP4 video only plays sound on Windows Media Player and QuickTime Player, however it plays normally on VLC and Classic Media Player
[22:17:29 CEST] <acamargo_> TwinTailed, which audio codec is being used?
[22:17:46 CEST] <flux> twintailed, I understand you're using FFmpeg as a library. have you tried if FFmpeg as a command line tool is able to make files you can play?
[22:18:08 CEST] <TwinTailed> acamargo_: aac
[22:18:25 CEST] <TwinTailed> flux: yes it does generate playable media, i know the problem is in my code
[22:18:28 CEST] <acamargo_> TwinTailed, use ffprobe or mediainfo on the output file to get all info
[22:19:03 CEST] <acamargo_> TwinTailed, which audio profile?
[22:27:18 CEST] <TwinTailed_> acamargo_: http://pastebin.com/Saz1srXd
[22:27:30 CEST] <TwinTailed_> flux: http://pastebin.com/Saz1srXd
[22:28:12 CEST] <TwinTailed_> I no longer which of u wasn't kicked from the last server overload
[22:31:21 CEST] <flux> twintailed_, not sure if you're going to get a lot of progress here unless we see the code and/or the file it produces
[22:31:38 CEST] <acamargo_> TwinTailed_, Stream #0:1(und), 1, 1/44100: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 130 kb/s (default)
[22:31:43 CEST] <acamargo_> TwinTailed_, seems ok to me
[22:31:59 CEST] <flux> the sound worked, video didn't :)
[22:31:59 CEST] <TwinTailed_> acamargo_: The video is the problem not the audio
[22:32:28 CEST] <TwinTailed_> flux: I can send u the code, but people like JEEB keep telling me "We don't have time to fix your code for you" so i stopped sending my code
[22:32:47 CEST] <acamargo_> TwinTailed_, Stream #0:0(und), 2, 1/12800: Video: h264 (High), 1 reference frame (avc1 / 0x31637661), yuv420p(left), 640x360 (640x368), 0/1, 1312 kb/s, 25 fps, 25 tbr, 12800 tbn, 100 tbc (default)
[22:33:09 CEST] <TwinTailed_> flux: http://pastebin.com/fxhvwUH8
[22:33:13 CEST] <flux> twintailed_, well, I don't actually have time ;-) to look at it tonight, but perhaps later..
[22:33:35 CEST] <TwinTailed_> flux: can you just see line 100?
[22:33:36 CEST] <flux> oh, it's not super long
[22:33:45 CEST] <TwinTailed_> flux: To know if I have copied all pars
[22:33:51 CEST] <TwinTailed_> required pars of course
[22:34:24 CEST] <acamargo_> TwinTailed_, where didn't play?
[22:34:27 CEST] <flux> twintailed_, btw, I expect memory management issues later on because of the verbatim copy of the extradata pointer
[22:34:36 CEST] <flux> I think yuo need to allocate memory for it etc
[22:34:47 CEST] <TwinTailed_> acamargo_: windows media player and quicktime player
[22:35:48 CEST] <TwinTailed_> flux: doesn't it already allocate everything upon avcodec_alloc_context3?
[22:36:27 CEST] <flux> twintailed_, even if it does, you're overwriting the extradata pointer with one originating from another structure
[22:36:35 CEST] <flux> twintailed_, I expect the extradata ends up being free'd twice
[22:37:03 CEST] <flux> it seems unlikely this is anyhow related to your problem
[22:37:24 CEST] <TwinTailed_> flux: what function should I use to allocate extradata?
[22:37:57 CEST] <flux> maybe malloc if av_malloc isn't part of the public api
[22:38:20 CEST] <TwinTailed_> ah, just a normal malloc, not a specific function
[22:38:26 CEST] <flux> yes
[22:38:42 CEST] <flux> but I'll be getting some sleep now. happy hacking, hope you get it fixed, you've been on it for quite a while already ;)
[22:39:45 CEST] <TwinTailed_> flux: ok good night, thank you :)
[22:39:46 CEST] <flux> who knows, if you get that working, it might make a decent doc/example for ffmpeg. it's quite simple compared to some of the examples..
[22:40:28 CEST] <TwinTailed_> flux: if it works i'll try sending it to the ffmpeg-devel
[22:42:41 CEST] <TwinTailed_> acamargo_: can you help?
[22:43:28 CEST] <acamargo_> TwinTailed_, I don't have any of them installed here :-/
[22:43:46 CEST] <TwinTailed_> ah ok :)
[22:44:29 CEST] <acamargo_> TwinTailed_, you can search if those players have codec h264 compatible with your output file
[22:44:40 CEST] <TwinTailed> acamargo_: They do
[22:44:57 CEST] <acamargo_> TwinTailed, well... then they will play hehehe
[22:45:05 CEST] <acamargo_> :-)
[22:45:18 CEST] <TwinTailed> acamargo_: Yeah, the problem is with my code i know that :P But its due to lack of documentation to be honest
[00:00:00 CEST] --- Wed Sep 28 2016