[Ffmpeg-devel-irc] ffmpeg.log.20140921

[burek]

[02:23] <judu> Hi, I would like to know, in ffmpeg 2.4 what is supposed to replace the SWR_CH_MAX that's defined in libswresample/swresample.h?
[02:26] <judu> In the "Macro Definition Documentation" section of http://ffmpeg.org/doxygen/2.4/group__lswr.html, the SWR_CH_MAX is documented, but in the file it's not defined anymore -> "#if LIBSWRESAMPLE_VERSION_MAJOR < 1" is false since 2.4.
[05:54] <Mephistos> hi, quick question, I would like to merge two videos together, but concatenate the first one, the second video being a replacement for the credits, is this possible to do with ffmpeg?  My first attempt failed.
[05:54] <c_14> What do you mean "merge"?
[05:55] <Mephistos> two input files, one output file, converting from lossless x264 to a vbr x264, with the credits part of the video replaced at the end (second video source).
[05:56] <c_14> sure
[05:56] <Mephistos> I have most of the command line, but when I did this, it didn't appear to replace the credits.
[05:57] <c_14> ffmpeg -i 1 -i 2 -filter_complex '[0]trim=cred_start:cred_end[tmp];[tmp][1]concat=n=2:v=1:a=0[out]' -map '[out]' -c:v x264 -crf 18 out.mkv
[05:57] <c_14> or something
[05:58] <c_14> eh, no wait
[05:58] <c_14> replace cred_start with 0 and cred_end with cred_start
[05:58] <Mephistos> ffmpeg -i source1-1080p-09b-x264-lossless.mp4 -t 4:04 -i source2-1080p-09cF-x264-lossless.mp4 -c:v libx264 -crf 23 -tune film -acodec aac -strict experimental -b:a 256k -ac 2 -f mp4 -threads 3  merged-final-1080p-09d-ffmpeg-x264-`date +%Y%m%d%H%M%S`.mp4
[05:58] <Mephistos> That's my command line right now
[05:59] <c_14> source2 is the credits or the to be cut one?
[05:59] <Mephistos> the credits
[05:59] <Mephistos> I want to cut it at 4:04 seconds exactly
[05:59] <c_14> put the -t 4:04 in front of source1 not source2
[05:59] <Mephistos> ok
[05:59] <c_14> and you're not actually concatting
[06:00] <c_14> add -filter_complex '[0][1]concat=n=2:v=1:a=1[v][a]' -map '[a]' -map '[v]'
[06:01] <Mephistos> I'm getting a tonne of buffer overflows
[06:02] <c_14> probably not fatal
[06:02] <c_14> audio related?
[06:02] <Mephistos> no idea
[06:02] <Mephistos> it says it's dropping frames
[06:29] <Mephistos> I did it as a two step process copying the codecs to remove the credits at the end.
[06:31] <Mephistos> and I'm using -f concat now, it's not giving me errors or dropping frames.
[06:32] <c_14> If it works, it's good.
[06:34] <Mephistos> I have to wait and see, but I think it will work no problem.
[07:59] <goulard> What is the purpose of AAC/ADTS extradata?
[08:00] <goulard> I can't find any info about it.  A small chunk... usually 5 bytes of data in between each ADTS packet
[10:02] <frickenate> sacarasc: Haha damn. I was going with AAC for HLS streaming. Quick search seems to indicate that mp3 + aac are both supported on all worthy platforms.
[10:06] <frickenate> Ah then again, maybe not. Native html5 support varies for mp3 and aac. Of course.
[10:23] <brontosaurusrex> frickenate: ffmpeg -codecs | grep mp3 < if you get the "D" then you have a decoder
[10:24] <frickenate> Ooooh I like the D.
[10:24] <frickenate> Still compiling for the first time.libavcodec takes a loooong time to compile :)
[10:25] <brontosaurusrex> and a nice browser test page http://hpr.dogphilosophy.net/test/
[10:31] <brontosaurusrex> and I don't think firefox has any mp4/aac support, but don't quote me
[10:35] <frickenate> I looked that up already. Firefox has AAC now except on OS X, which is coming next month with ff 34.
[10:35] <brontosaurusrex> ok, I stand corrected
[10:35] <frickenate> But yeah, still fragmented. All hail flash player fallback.
[10:36] <brontosaurusrex> yeah, makes you wonder why bother with html5 if we have to flash fallback ...
[10:37] <brontosaurusrex> so ff 34 will have that cisco decoders working ... ?
[10:38] <frickenate> ooh fancy: http://areweplayingyet.org/
[12:09] <JazzCZ> Hey
[12:10] <JazzCZ> where can I find the TrueHD patch?
[12:23] <JazzCZ> anyone knows how to apply truehd patch?
[12:31] <sara> hello
[12:32] <sara> I have a quick q on ffmpeg, I keep getting "no such file or directory" when trying to encode from flv to mp4
[12:32] <sara> any help? :)
[12:36] <sacarasc> sara: If the file name has spaces in them, you'll have to either escape it or put quotes around it.
[12:37] <sacarasc> Or, the file doesn't exist in the directory you're calling ffmpeg from.
[12:37] <sara> I psted the file in the same directory as the ffmpeg executable files
[12:37] <sara> lemme try the quote thingy,
[12:38] <sara> nope, nothing
[12:39] <sara> this is the command I'm using: ffmpeg -i sample sample.mp4
[12:39] <sara> I've tried sample.flv too
[12:39] <sara> but nothing
[12:39] <sara> as my input i mean
[12:40] <sacarasc> Is there a file called "sample" in the directory?
[12:40] <sacarasc> Or sample.flv?
[12:40] <sara> yes
[12:40] <sara> the file i pasted in there
[12:40] <sara> named merely sample
[12:40] <sara> I pasted it in the "bin" folder
[12:41] <sara> where the 3 ffmpeg executable files are found
[12:41] <sara> when i downloaded ffmpeg
[12:41] <sacarasc> And you're in that directory trying to convert?
[12:42] <sara> yes
[12:42] <sara> (i think)
[12:43] <sacarasc> Which OS are you on?
[12:43] <sara> so the file is in the ffmpeg directory (under my c drive(
[12:43] <sara> and the command line i'm putting includes the ffmpeg directory
[12:43] <sara> is that right>?
[12:43] <sara> if I'm inserting as my command line
[12:43] <sara> ffmpeg -i sample sample.mp4
[12:43] <sara> windows#
[12:43] <sara> windows 7
[12:45] <sacarasc> You're in a directory similar to C:\ffmpeg\bin ?
[12:45] <sacarasc> If do you the command "dir", do you see ffmpeg.exe and the file sample?
[12:45] <sara> that's correct
[12:45] <sara> oops
[12:45] <sara> lemme check
[12:46] <sara> so merely type in the command line...
[12:46] <sara> dir ffmpeg?
[12:47] <sara> if i put dir simple
[12:47] <sara> *simpoly
[12:47] <sara> i don't see ffmpeg
[12:47] <sara> but i can my sample file
[12:48] <sara> ah, so I need to actually copy pasted the ffmpeg files into this directory?
[12:48] <sacarasc> Okay, change your command to something like C:\ffmpeg\bin\ffmpeg.exe -i sample sample.mp4
[12:48] <sacarasc> Replacing the "C:\ffmpeg\bin\" part with the directory that ffmpeg.exe is in.
[12:51] <sara> but ffmpeg.exe
[12:51] <sara> is in C:\ffmpeg\bin already
[12:51] <sara> though my executable file doesn't have the .exe postfix, is that normal?
[12:52] <sara> so it's such ffmpeg in the bin folder
[12:52] <sacarasc> Where did you download it from?
[12:53] <sara> http://ffmpeg.zeranoe.com/builds/
[12:53] <sara> i download the static version for my system
[12:54] <sacarasc> Okay. And on the command line "dir" showed the file "sample"? Not "sample.flv"?
[12:54] <sara> in dir it says "sample.flv
[12:55] <sacarasc> Okay, type this: C:\ffmpeg\bin\ffmpeg.exe -i sample.flv sample.mp4
[12:55] <sacarasc> That should work.
[12:59] <sara> nope....
[12:59] <sara> but
[12:59] <sara> I just rerun ffmpeg
[12:59] <sara> i mean I just re-did
[12:59] <sara> dir
[12:59] <sara> and ffmpeg is there
[13:00] <sara> I copy-pasted the entire ffmpeg folder into my home directory
[13:01] <sara> op!
[13:01] <sara> seems to be working
[13:02] <sara> ,,,,converting....
[13:02] <sara> is there a way to automate this for a batch process?
[13:02] <sara> or would I need to do the same thing x 100 for a 100 videos?
[13:03] <sacarasc> If you know how to batch script, you can do it. I don't know how, so can't help there.
[13:06] <sara> okey dokey
[13:06] <sara> thank you!
[13:39] <judu> The wI would like to know, in ffmpeg 2.4 what is supposed to replace the SWR_CH_MAX that's defined in libswresample/swresample.h?
[13:39] <judu> In the "Macro Definition Documentation" section of http://ffmpeg.org/doxygen/2.4/group__lswr.html, the SWR_CH_MAX is documented, but in the file it's not defined anymore -> "#if LIBSWRESAMPLE_VERSION_MAJOR < 1" is false since 2.4.
[14:31] <kaotiko> hi
[14:34] <hatsch> hi everybody
[14:35] <Sokolio> Hi
[14:36] <hatsch> i am trying to use the new icecast protocol in ffmpeg to send  vpx video and vorbis audio in a webm container to an icecast server. from the ffmpeg side of view everythinks seems to be okay, i get connected to the server and the input is processing. but receiving the video with eg vlc gives an error about a wrong header ( mp3)  http://pastebin.com/WGYLxZHS
[14:37] <hatsch> don't know for sure if it's an ffmpeg or icecast issue. but sending webm to icecast with gstreamer is working.
[15:57] <sara> hi
[15:57] <sara> is it possible to convert and compress a video file at the same time in ffmpeg?
[15:58] <sara> e.g. convert a mp4 to flv and compress from 100MB to a smaller size (ideally with minimum loss of quality)
[15:58] <sara> i know how the command line to convert
[15:58] <Suchiman> usually transcoding (convert) compresses
[15:58] <sara> could you give me the command line for compressing+converting at the same time?
[15:59] <Suchiman> though you can adjust Quality / Bitrate to Archive compression
[15:59] <brontosaurusrex> sara: flv and mp4 are containers
[15:59] <Suchiman> but compression is archived through reducing quality
[15:59] <sara> how can I adjust the quality/bitrate?
[16:00] <sara> is the bitrate value _b:v in the command line?
[16:00] <sara> woops sorry, just saw your comm on containers
[16:01] <sara> you lost me a bit...
[16:01] <brontosaurusrex> meaning flv can contain various codecs
[16:01] <brontosaurusrex> perhaps read this http://en.wikipedia.org/wiki/Flash_Video
[16:02] <brontosaurusrex> or to refraze; different compression formats
[16:02] <sara> I try the command line ffmpeg -i sample.mp4 -ar 22050 sample.flv
[16:02] <Suchiman> sara: yeah b:v sets Bitrate but depending on codec there are various Parameters to tweak
[16:02] <sara> ok
[16:02] <sara> and is the codec dependent on container?
[16:03] <sara> or could i use h.264 for all all containers?
[16:03] <brontosaurusrex> you could use h.264 for both mp4 and flv
[16:03] <Suchiman> sara: Containers can contain from 1 up to almost any codec
[16:03] <Suchiman> depends on the container
[16:03] <sara> aha
[16:04] <Suchiman> sara: the Player at the other end must support both, Container and codec
[16:06] <sara> of course, makes sense
[16:06] <sara> :)
[16:07] <brontosaurusrex> anyway, why would you need flv?
[16:45] <sruli> hi all, i hope someone can help me. I am trying to create a video from 3 clips, and an overlay on 1, only 1 of the clips have audio, but i get an error, everytime, please see http://pastebin.com/aCjmQnw5 for command i use and error
[17:16] <Lac3rat3d> Does ffmpeg support decoding DTS-ES? I found some threads about supporting it, with patches etc, but when I try to convert a file with DTS-ES to AAC native, I get an error message
[17:20] <Lac3rat3d> http://lists.ffmpeg.org/pipermail/ffmpeg-devel/2010-June/091970.html      http://lists.ffmpeg.org/pipermail/ffmpeg-devel/2014-April/156698.html      http://lists.ffmpeg.org/pipermail/ffmpeg-devel/2010-June/091970.html
[18:44] <bearish> hey guys, i'm getting real bad performance with libvpx compared to libx264, any pointers?
[18:46] <Nosomy|off> only performance?
[18:48] <bearish> So i have this H264 video piping in to ffmpeg
[18:49] <bearish> and i'm trying to get it working on a chromecast
[18:49] <bearish> which is picky about it's formats
[18:49] <bearish> i'm using ffserver to restream
[18:53] <bearish> So I can't use MPEG-TS container
[18:53] <bearish> can't use mp4, muxer does not support non seekable output
[19:00] <hatsch>  bearish: did you set the -threads option (it seems 0/auto is not working with libvpx )
[19:03] <bearish> it doesn't seem to work either way, only 1 core gets utilized
[19:04] <bearish> and i get 9-15 fps
[19:04] <bearish> where on half a core libx264 produces ~30-35fps
[19:04] <JEEB> libvpx only has threading with vp8
[19:04] <JEEB> not vp9
[19:04] <theholyduck> theres still no threading on vp9?
[19:05] <JEEB> of course not
[19:05] <JEEB> google doesn't need it
[19:05] <theholyduck> JEEB, well, there wasnt any threading on it like, many months ago
[19:05] <JEEB> they just do GOP-based encoding and stitch shit together
[19:05] <theholyduck> when i last tested it
[19:05] <theholyduck> so i just asumed they would add it at some point
[19:05] <theholyduck> to make it not painfully slow
[19:05] <JEEB> lol
[19:05] <theholyduck> i guess i was wrong
[19:07] <theholyduck> also 9-15fps is a lot better than what i got with, libvpx last time around, though, i asked them for some high quality settings to use.
[19:07] <theholyduck> it took 13 hours to encode 2000 frames
[19:07] <theholyduck> on my poor i7
[19:07] <JEEB> yeah, compression-side settings are lolslow
[19:07] <JEEB> but I guess libvpx can get some speed
[19:07] <JEEB> (even with vp9)
[19:08] <JEEB> vp8 should be kind of faster, and have threading
[19:08] <theholyduck> well, who knows, they might have implemented some optimizations, last time i tried using vp9 was right ater it was "released"
[19:08] <theholyduck> might not have any threading, but they could atleast write faster code
[19:09] <ubitux> am i reading that libvpx is now usable for vp9 encoding?
[19:09] <theholyduck> ubitux, i think usable is in the eye of the beholder
[19:09] <theholyduck> ubitux, its slow and doesnt offer massive quality gains over x264
[19:10] <theholyduck> so, in my mind, theres no reason to use it at all.
[19:10] <ubitux> no, no one except google can use libvpx
[19:10] <ubitux> several days to encode a few thousands frames is not usable
[19:10] <ubitux> :p
[19:10] <theholyduck> ubitux, well, you could reduce the settings so that it makes worse quality than x264
[19:10] <theholyduck> and it wont take several days
[19:11] <theholyduck> ;P
[19:11] <ubitux> will it be faster than x264?
[19:11] <theholyduck> i havent tested it in forever, but seeing as its still not threaded
[19:11] <theholyduck> id say no
[19:20] <Lac3rat3d> Does ffmpeg support decoding DTS-ES? I found some threads about supporting it, with patches etc, but when I try to convert a file with DTS-ES to AAC native, I get an error message
[19:54] <Lac3rat3d> no one?
[20:14] <hatsch> anyone got webm streaming to icecast2.4 working?
[20:19] <brian_ch> Hi, sorry if it's a dumb question, but are the FF_API_ defines really only for future reference, or it can happen that in some cases they are used otherwise? Is it safe to assume that all code between a FF_API_ ifdef will be compiled in that specific version of ffmpeg? Or the old defines are not always removed right away after the version bump?
[20:36] <brian_ch> I ask because I was taking a look at the Pascal translation of the headers, and was wondering if it would be ok to drop these defines, or if instead of simplifying this would bring even more problems.
[21:05] <frog_> hi. Some where i read that Ubuntu is changing from ffmpeg to avconv. Is this right?
[21:05] <JEEB> quite some time ago they switched from ffmpeg to libav (when libav forked)
[21:06] <JEEB> so if you want an up-to-date binary you use avconv on debian-based systems
[21:06] <JEEB> (or you install an up-to-date ffmpeg otherwise)
[21:06] <frog_> ok. thanks
[21:07] <Lac3rat3d> Does ffmpeg support decoding DTS-ES? I found some threads about supporting it, with patches etc, but when I try to convert a file with DTS-ES to AAC native, I get an error message
[21:26] <gcl5cp> how can i remove cover art in a M4A?
[21:27] <gcl5cp> COVER ART --> Stream #0:1: Video: mjpeg, yuvj420p, 600x600 [SAR 72:72 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
[21:33] <gcl5cp> ffmpeg -i original.m4a -map 0:0 -c:a copy non-cover.m4a. do nothing
[21:33] <c_14> -vn
[21:33] <c_14> probably
[21:33] <gcl5cp> do nothing
[21:35] <gcl5cp> in ID3Tag editors (easytag puddletag picard) i can't see the cover.
[21:36] <gcl5cp> i just need to replace to the correct cover.
[21:36] <gcl5cp> ffmpeg -i original.m4a -vn -map 0:0 -c:a copy non-cover.m4a. do nothing
[21:43] <gcl5cp> Stream #0:1 seems to be inside Metadata
[21:44] <gcl5cp> how can i erase metadata?
[21:45] <c_14> -map_metadata -0 or something
[21:51] <gcl5cp> only with this works. "-map_metadata -1 -vn -c:a copy"
[21:52] <gcl5cp> thank c_14
[21:56] <bearish> What am I doing wrong? All this hls stuff is hurting my brain
[21:56] <bearish> http://paste.pound-python.org/show/IBdwS5NzYBch44o8XjnH/
[22:08] <eago_> hello
[22:09] <EvolE> eago_: hi
[22:09] <eago_> I'm trying to use ffmpeg to stream a live video from a browser
[22:10] <eago_> using the webcam
[22:11] <eago_> was planning to use webrtc and node.js
[22:11] <eago_> but I'm having problems finding docuemtnation on how to do it
[22:13] <EvolE> so you gonna get video stream to your nodejs server and then you will direct it to ffmpeg?
[22:14] <eago_> yes
[22:14] <eago_> or is there a simpler way?
[22:15] <EvolE> what you plan to do next with that video after ffmpeg?
[22:15] <eago_> stream it using ffserver
[22:15] <eago_> basically on user will enter the website and broadcast ysing the web cam
[22:15] <eago_> then other users can see that video
[22:16] <EvolE> k, now i got it
[22:24] <EvolE> eago_: did you see this? https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC/RecordRTC-to-Nodejs
[22:25] <eago_> yes.... but that seems to record many small files
[22:25] <EvolE> eago_: but there video goes to node like a chunk. not sure if it can be modified to make endless recording and endless chunk sending
[22:25] <eago_> instead of a singel stream
[22:26] <eago_> ok.... so I call ffmpeg to use that file as input
[22:27] <eago_> or can I connect it directly to the data sent by node.js?
[22:28] <EvolE> eago_: seems like this is better to modify https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC/RecordRTC-over-Socketio
[22:28] <eago_> ah, yup
[22:28] <EvolE> you can make ffmpeg read from pipe. probably there should be something for nodejs also
[22:31] <eago_> those look good for first recording and then serving... but not sure how to use it for live constant stream
[23:19] <zenny1> Hi, when overlayed two videos side-by-side, the video on right disappears after a few seconds. I am using 'ffmpeg -i Audio.wav -i Rightvideo.mp4 -vf "[in] scale=iw/2:ih/2, pad=2*iw:ih [left]; movie=LeftVideo.mp4, scale=iw/2:ih/2, fade=out:300:30:alpha=1 [right]; [left][right] overlay=main_w/2:0 [out]" -b:v 768k final_output.mp4'. Any input on why the right video disappears after a few seconds?
[23:44] <zenny1> I also tried with 'ffmpeg -i Audio.mp2 -i Rightvideo.mp4 -vf "[in] scale=iw/2:ih/2, pad=2*iw:ih [left]; movie=LeftVideo.mp4, scale=iw/2:ih/2, fade=out:300:30:alpha=1 [right]; [left][right] overlay=main_w/2:0 [out]" -b:v 768k final_output.mpg', still the right video vanishes after a few seconds!!! What am I doing wrong?!!
[23:55] <benlieb> I'm trying to clean up the audio in a video. I've separated the audio to clean it up externally, but when I try to recombine with the video (with old audio removed) it doesn't play the sound .
[23:55] <benlieb> ffmpeg -i videomute.mp4 -i audio.mp3 -c copy found1.mp4
[23:56] <benlieb> The audio is in the file according to ffprobe
[00:00] --- Mon Sep 22 2014