[Ffmpeg-devel-irc] ffmpeg.log.20150701

burek burek021 at gmail.com
Thu Jul 2 02:05:01 CEST 2015

[04:20:24 CEST] <chocoladedevelop> Hi
[04:20:56 CEST] <chocoladedevelop> I'm using c# with ffmpeg and pipes and i'm pushing frames and create in real time compressed mp4 video file.
[04:21:26 CEST] <chocoladedevelop> The problem is that after few seconds the ffmpeg.exe memory usage is raising to over 1GB of memory and the hard disk raise up to over 53MB/s
[04:21:53 CEST] <chocoladedevelop> Is that a known problem ?
[04:25:20 CEST] <chocoladedevelop> I posted my code in csharp in the ffmpeg forum here: http://ffmpeg.gusari.org/viewtopic.php?f=11&t=2204 and also here: http://stackoverflow.com/questions/31149752/why-when-using-ffmpeg-to-create-video-file-in-real-time-the-ffmpeg-exe-take-over
[05:32:47 CEST] <fred1807> I have to need to always convert mp4 to raw h264 for my application. I use this cmd , ffmpeg -i input.mp4 output.h264  (sometimes with resolution scale included).  What would be the best way to automate this process ?  Can I create a shell window where I can simply throw my input files over it?
[05:50:35 CEST] <godofgrunts> I'm struggling to figure out how to burn .ass subtitles into an mp4. I suspect my problem is that I don't know the name of the .ass file, but I'm not sure how to get it. trac.ffmpeg.org tells me to do ffmpeg -i video.mkv -vf "ass=subtitle.ass" out.mkv but I get the following output http://pastebin.com/eujhxdYx
[05:51:19 CEST] <godofgrunts> BTW ffmpeg.pastebin.org can't be resolved for me
[05:54:47 CEST] <godofgrunts> as far as the video info in concerned http://pastebin.com/BKbGxGfM
[06:35:02 CEST] <chungy> godofgrunts: I think you're going to have to extract it to a separate file first; ffmpeg -i video.mkv -c copy -map 0:2 video.ass
[06:36:27 CEST] <chungy> (-c copy is probably unnecessary, but it keeps the ass stream as-is)
[06:56:53 CEST] <godofgrunts> chungy, seems to be working, I'll let you know once the video is done
[07:01:30 CEST] <godofgrunts> just produces a black screen for some reason
[07:19:14 CEST] <Mista_D> Looking for old FFmpeg source files, git version 31792
[07:20:37 CEST] <anoop_r> anybody know how to use x265 multilib with ffmpeg
[07:25:00 CEST] <godofgrunts> anoop_r, do you have --enable-libx265?
[07:25:18 CEST] <anoop_r> yes
[07:25:37 CEST] <anoop_r> anything else needed
[07:27:12 CEST] <godofgrunts> Forgive my ignorance, but are you trying to compile ffmpeg or just encode something with x265?
[07:27:55 CEST] <anoop_r> x265 mulilib means 8bit and 10 bit builds together
[07:28:20 CEST] <anoop_r> so i want to add x265 multilib to ffmpeg
[07:28:28 CEST] <godofgrunts> Oh I see. Sorry, I have no idea
[07:29:11 CEST] <anoop_r> i too have no idea
[07:32:07 CEST] <godofgrunts> anoop_r, this is old and for x264, but maybe it'll help https://ffmpeg.org/pipermail/ffmpeg-user/2013-January/012769.html
[08:38:29 CEST] <anoop_r> does ffmpeg encodes audio and video simultaniously ?
[08:47:56 CEST] <anoop_r> i had encountered a problem
[08:48:12 CEST] <anoop_r> i am using x265 10bit + ffmpeg
[08:49:15 CEST] <anoop_r> the encoding process will only finish after some times even after the frame count reaches the maximum farme count
[08:49:45 CEST] <chocoladedevelop> ffmpeg.exe have a memory leak ?
[08:50:10 CEST] <anoop_r> but for a certain build encoding process outs output file exactly when frames reaches maximum farme of out put
[08:50:29 CEST] <anoop_r> file
[08:50:40 CEST] <anoop_r> i don't know
[08:51:58 CEST] <anoop_r> does ffmpeg have any performance builds
[08:52:54 CEST] <anoop_r> it's very hard to explain
[08:53:00 CEST] <anoop_r> let me try again
[08:53:17 CEST] <anoop_r>  see the progress is show as frame=   30
[08:53:36 CEST] <anoop_r> let say the media have 300 frames
[08:54:11 CEST] <anoop_r> when frame= 300 it should immediatly complete encoding and output file
[08:54:35 CEST] <anoop_r> but its taking some more time to finish
[08:54:54 CEST] <godofgrunts> I've had that issue with VP9 encoding as well
[08:55:34 CEST] <anoop_r> but a certain previous source was able to provide output at the same time when frame=   300
[08:55:37 CEST] <anoop_r> any idea
[08:55:50 CEST] <godofgrunts> I'm 99% sure that ffmpeg does lazy parrel pocessing in that it will x cores to encode video y cores to encode audio, etc
[08:56:19 CEST] <godofgrunts> I don't know enough about muxing to be certain, but I feel the delay comes in the remuxing
[08:56:43 CEST] <anoop_r> so which version of  ffmpeg source will give performance
[08:57:22 CEST] <godofgrunts> I'm encoding a video with 3 streams right now and I'm using 6 cores so I assume each stream is assigned two cores
[08:57:58 CEST] <godofgrunts> I don't know that to be honest. I'm using static builds from zeranoe
[08:58:53 CEST] <anoop_r> which ffmpeg source gives optimum performance
[08:59:09 CEST] <godofgrunts> I couldn't tell you.
[08:59:43 CEST] <chungy> x265 is slow in general, you'll just have to be patient with it
[09:00:07 CEST] <anoop_r> i know that
[09:00:26 CEST] <anoop_r> but i got a build which gives zero delay
[09:01:00 CEST] <anoop_r> but after using number of updated source i lost it
[09:01:18 CEST] <anoop_r> anybody else encountered any such issue
[09:02:28 CEST] <pandb> I keep getting the error: "unknown type name AVPixelFormat" and I can't figure out why. I know it's an enum defined in libavutil/pixfmt.h, but including that or avutil.h doesn't seem to make a difference
[09:06:17 CEST] <chocoladedevelop> I remember asking about 2-3 years ago about the ffmpeg.exe memory leak but nothing. No one gave a solution or any info.
[09:06:24 CEST] <chocoladedevelop> And now i have the same problem.
[09:06:57 CEST] <anoop_r> x265-43afbde189f3
[09:07:06 CEST] <chocoladedevelop> Strange it seems i'm the only one with this problem.
[09:07:30 CEST] <anoop_r> this commit gives more performance but it will not give output without delays
[09:08:29 CEST] <chungy> chocoladedevelop: what leak
[09:09:29 CEST] <chocoladedevelop> chungy in c# using pipe to compress batch of images and create mp4 video file in real time make the ffmpeg.exe file memory usage to jump after 3-5 seconds from 0.1MB to over 1GB usage.
[09:09:38 CEST] <chocoladedevelop> Also the hard disk usage raise to over 53MB/s
[09:10:05 CEST] <chungy> not necessarily a leak.
[09:10:08 CEST] <chocoladedevelop> I see it in the Task Manager the ffmpeg.exe just jump in seconds to over 1GB memory usage.
[09:10:28 CEST] <chocoladedevelop> What can it be else ?
[09:10:36 CEST] <chungy> right, if the uncompressed image data takes over 1GB, then that's what it is... that wouldn't be a leak.
[09:11:09 CEST] <chungy> you may want to ask the mailing list though to be certain
[09:13:36 CEST] <chocoladedevelop> Ok i will try to ask. But if what you say is true then what can i do ? not sure.
[09:13:50 CEST] <chocoladedevelop> How do i get ot the mailing list ?
[09:15:17 CEST] <pandb> ahhh 'enum AVPixelFormat...' :)
[09:15:49 CEST] <chungy> https://ffmpeg.org/contact.html#MailingLists
[09:15:55 CEST] <pandb> im not even mad that everyone was silent :)
[09:17:55 CEST] <chocoladedevelop> Thanks,
[09:54:15 CEST] <equanox> Hey, i have a question about interlaced encoding
[09:55:09 CEST] <equanox> i try to reencode a video from a interlaced source in a mts container
[09:55:27 CEST] <equanox> to a interlaced mp4
[09:56:15 CEST] <equanox> after reencoding the metadata of the output file is 'scan_type' is progressive
[09:56:57 CEST] <equanox> i'm not able to see the parameter 'scan_type' with ffprobe
[09:57:15 CEST] <equanox> but mediaarea shows this meta params
[09:57:51 CEST] <equanox> the fottage inside the output mp4 file is still interlaced
[09:58:17 CEST] <Mavrik> IIRC that's expected
[09:58:37 CEST] <equanox> but due to the metadata winmedia player will not detect it correctly
[09:59:29 CEST] <equanox> is there a way to set the metadata manually?
[10:03:30 CEST] <Mavrik> equanox, does manually setting field order with "-field_order tt" help?
[10:03:46 CEST] <Mavrik> do you explicitly set interlaced output for the encoder?
[10:19:08 CEST] <claz> if i try to cut the last frame of a video by seeeking to it's duration it often doesn't work
[10:19:17 CEST] <claz> it doesn't error or anything
[10:19:24 CEST] <claz> just doesn't create the output file
[10:19:35 CEST] <claz> is there any way to cut the last frame without probing to see it's timestamp?
[11:20:55 CEST] <equanox> i found a answer to my question
[11:20:58 CEST] <equanox> -flags +ildct+ilme -x264opts tff=1
[11:21:02 CEST] <equanox> is doing the job
[11:21:18 CEST] <Mavrik> ah
[11:21:24 CEST] <Mavrik> so you were actually encoding a progressive video
[11:21:30 CEST] <equanox> the problam is . mp4 does not traditionall interlaced
[11:21:54 CEST] <equanox> instead they use MBAFF
[11:22:28 CEST] <equanox> no the video was interlaced
[11:23:01 CEST] <JEEBsv> if it was interlaced and your source noted it correctly, it should be encoded through interlaced
[11:23:14 CEST] <JEEBsv> MBAFF is interlaced if the tag is there in the input bit stream
[11:23:25 CEST] <JEEBsv> both PAFF and MBAFF should come up as interlaced in case it is interlaced
[11:23:36 CEST] <JEEBsv> if it is not, well then you can blame the creator of the original file :P
[11:23:40 CEST] <Mavrik> ffmpeg doensn't do PAFF at all afaik
[11:23:43 CEST] <Mavrik> or was it x264?
[11:23:45 CEST] <JEEBsv> x264
[11:23:56 CEST] <JEEBsv> since the benefits of PAFF are like none
[11:24:08 CEST] <JEEBsv> and the amount of work needed to add PAFF coding is on the other hand...
[11:28:20 CEST] <equanox> the sources "scan_type" was named as interlaced
[11:28:58 CEST] <equanox> but the output files "scan_type" was progressive (but containing the interlaced video)
[11:29:09 CEST] <equanox> yes the encoder was x264
[11:34:27 CEST] <JEEBsv> how old is your ffmpeg?
[11:35:04 CEST] <equanox> only a few months old
[11:36:13 CEST] <JEEBsv> ffmpeg -version ?
[11:37:30 CEST] <equanox> ffmpeg version N-69972-g6c91afe Copyright (c) 2000-2015 the FFmpeg developers
[11:37:37 CEST] <Mavrik> equanox, did you set the field type flag?
[11:37:48 CEST] <JEEBsv> Mavrik: shouldn't it grab that from the decoded picture?
[11:38:02 CEST] <JEEBsv> unless ffmpeg is retarded and will not take that into mention when passing things into libx264
[11:38:16 CEST] <Mavrik> When in doubt, I rely on ffmpeg being retarded
[11:38:25 CEST] <Mavrik> I see that it reads the field_type AVContext in movenc.c
[11:38:31 CEST] <equanox> i tried setting the field type flag
[11:38:36 CEST] <Mavrik> But I'm not entirely sure if it can reliably set that on H.264
[11:38:36 CEST] <JEEBsv> movenc!?
[11:38:43 CEST] <equanox> but it doesn't had any effect
[11:38:46 CEST] <JEEBsv> the container should have nothing to do with it
[11:38:52 CEST] <JEEBsv> it's a bit stream thing
[11:39:18 CEST] <Mavrik> Unless you want the container flag set :)
[11:39:56 CEST] <JEEBsv> no, I mean the decoder should output pictures with the interlacism flag set, and then libx264 should be passed that information
[11:40:02 CEST] <JEEBsv> (unless ffmpeg is retarded enough not to do that)
[11:40:25 CEST] <JEEBsv> if the decoded picture is not interlaced then of course there's no way to pass that information :P
[11:40:32 CEST] <RobertNagy> Does ffmpeg somehow encode gapless meta-data for aac in mp4? i.e. front padding, end padding and real samples?
[11:41:28 CEST] <JEEBsv> RobertNagy: if the encoder outputs negative PTS at first, that will be written in the mp4 file
[11:41:33 CEST] <RobertNagy> similar to how itunes uses "iTunSMPB" meta headers. I can see that ffmpeg parses "iTunSMPB" but I can't see any such data being written when muxing.
[11:41:36 CEST] <JEEBsv> as that would be the encoder delay
[11:41:48 CEST] <JEEBsv> yeah, ffmpeg uses standard stuff in the container
[11:42:01 CEST] <RobertNagy> so how would I get those 3 values?
[11:42:15 CEST] <RobertNagy> front padding = 0 - first pts?
[11:42:31 CEST] <RobertNagy> real samples = duration?
[11:42:34 CEST] <RobertNagy> end padding = ???
[11:42:42 CEST] <JEEBsv> there shouldn't be any padding at the end
[11:42:49 CEST] <JEEBsv> the stream has duration X and then you have the encoder delay
[11:43:09 CEST] <RobertNagy> I assume the encoder delay is not included in the duration?
[11:43:37 CEST] <JEEBsv> I don't exactly remember the way ffmpeg outputs the stuff, but IIRC it does take it into mention
[11:44:02 CEST] <JEEBsv> the encoder delay is for the encoder's padding that's in the front, basically
[11:44:20 CEST] <JEEBsv> so anything sane will not play those samples
[11:44:24 CEST] <JEEBsv> (usually silence)
[11:44:36 CEST] <JEEBsv> and then ffmpeg doesn't do by itself any other checking of what you feed to it
[11:44:44 CEST] <JEEBsv> so it's the stream length and that's it
[11:44:50 CEST] <Mavrik> JEEBsv, it does, but equanox is complaining about MP4 file flags, not the actual video - he is saying that it's indeed interalced :)
[11:44:53 CEST] <RobertNagy> yea, I need to know the padding for playback using Media Extensions in Chrome
[11:45:19 CEST] <JEEBsv> Mavrik: of course, but he has also not given any info
[11:45:40 CEST] <JEEBsv> but as I noted, I have no idea if ffmpeg does the Right Thing
[11:45:49 CEST] <JEEBsv> it could have some retarded logic that encodes everything as progressive
[11:46:04 CEST] <JEEBsv> but on the other hand, it could just be that his bit stream is actually flagged as progressive
[11:46:47 CEST] <RobertNagy> JEEBsv: actually there is one problem. If the media doesn't start a pts=0 I have no way of knowing the padding.
[11:47:08 CEST] <RobertNagy> Though, I could guess it based on the previous segment. Maybe that would work.
[11:51:49 CEST] <RobertNagy> Actually, ffprobe gives me 'start: 0.000000', which can't be right...
[11:56:00 CEST] <RobertNagy> will have to use afconvert for now then...
[11:56:43 CEST] <JEEBsv> I only know that since around 2013 encoders that actually give negative PTS should have their encoder delay stored correctly
[11:56:47 CEST] <JEEBsv> not how it's shown
[12:07:04 CEST] <RobertNagy> so how can I read it?
[12:07:22 CEST] <RobertNagy> bruteforce with ashowinfo?
[12:10:25 CEST] <JEEBsv> I will probably test later and check how it's shown
[12:10:43 CEST] <JEEBsv> for API users it should be rather simple, your decoded audio packets would have a negative PTS until the encoder delay finishes
[12:11:11 CEST] <RobertNagy> yea, it just feels overkill to create a whole new application just for this
[12:11:26 CEST] <RobertNagy> and you are sure there is no end padding with libfdk_aac?
[12:11:30 CEST] <RobertNagy> nvm
[12:11:36 CEST] <RobertNagy> use duration...
[12:11:48 CEST] <RobertNagy> given that the duration is sample accurate...
[13:13:25 CEST] <theeboat> Hello, im using ffmpeg to capture video from a multicast stream. Does anybody know if its possible to force a black frame when there are no frames from the multicast stream?
[14:26:39 CEST] <zhanshan> hi
[14:27:12 CEST] <zhanshan> can mkv take wav format? and If input is 32bit float what should I use to make the resulting file properly playable?
[14:28:11 CEST] <zhanshan> given command line is: $ ffmpeg -f image2 -pattern_type glob -i '*.jpg' -i 'sound-in.wav' -pix_fmt yuv420p -c:v libx264 -crf 18 -x264opts "keyint=50" -c:a wav 'output.mkv'
[14:30:45 CEST] <Mavrik> zhanshan, you probably mean PCM not WAV
[14:30:54 CEST] <Mavrik> raw audio right?
[14:31:38 CEST] <zhanshan> it's lossless wav 32bit float
[14:31:54 CEST] <zhanshan> I didn't know wav is called "raw"
[14:32:08 CEST] <zhanshan> but if that's so I've learned something new
[14:32:36 CEST] <Mavrik> huh
[14:32:43 CEST] <Mavrik> WAV is a container
[14:32:51 CEST] <Mavrik> that contains non-compressed audio
[14:33:02 CEST] <Mavrik> and non-compressed audio is usually referred to as PCM
[14:33:27 CEST] <zhanshan> maybe I should use -c:a pcm_s16le ?
[14:33:33 CEST] <Mavrik> if you have 32-bit float
[14:33:41 CEST] <Mavrik> you should use the format that sets that :)
[14:33:44 CEST] <Mavrik> pcm_f32le
[14:33:57 CEST] <Mavrik>  DEA..S pcm_f32le            PCM 32-bit floating point little-endian
[14:34:02 CEST] <zhanshan> original recording it was 24bit/48KHz
[14:34:32 CEST] <zhanshan> that'd make pcm_s24le ?
[14:34:37 CEST] <zhanshan> does that exist?
[14:34:58 CEST] <Mavrik> yes
[14:35:04 CEST] <Mavrik> look at ffmpeg -codecs
[14:37:06 CEST] <zhanshan> so many!!
[14:37:19 CEST] <zhanshan> 4 different pcm_s24
[14:37:38 CEST] <zhanshan> be, le, daud, le_planar
[14:37:45 CEST] <zhanshan> what's the meaning of all that
[14:37:51 CEST] <godofgrunts> btw, mkv is the only container that can hold pretty much anything.
[14:37:52 CEST] <zhanshan> so many choices
[14:38:01 CEST] <zhanshan> godofgrunts thanks for that!
[14:38:04 CEST] <chungy> if your input file is a *.wav, you can just use "-i file.wav -c:a copy"
[14:38:07 CEST] <zhanshan> I love mkv I guess
[14:39:20 CEST] <zhanshan> chungy that's a good one, too, but since I drove a bit too far by exporting the audio from blender to 32bit float
[14:39:55 CEST] <zhanshan> and the recorded source was "only" 24bit I want to keep it was little as possible with the about "original" quality
[14:40:21 CEST] <JEEBsv> well if it's in float you did something to it, right?
[14:40:23 CEST] <chungy> fwiw, flac is usually a good choice. It's lossless but often is much smaller than the raw stream :P
[14:40:45 CEST] <JEEBsv> in that case you've already derped it and you should not decide anything on the original bit depth
[14:40:57 CEST] <zhanshan> blender doesn't seem to have to ability to export audio in 24bit?
[14:41:00 CEST] <JEEBsv> it's the same issue as with people wanting to re-encode shit with the same bit rate :P
[14:41:09 CEST] <zhanshan> so what should I've done to do it right?
[14:41:22 CEST] <zhanshan> export it to 32bit signed or 16bit signed instead?
[14:41:28 CEST] <JEEBsv> whatever you goddamn like :)
[14:41:34 CEST] <zhanshan> to be as close as possible to 24bit signed?
[14:41:45 CEST] <zhanshan> JEEBsv that's a good hint, too
[14:42:03 CEST] <chungy> pcm_s24le is as close as possible to 24-bit.
[14:42:21 CEST] <JEEBsv> unless you did only lossless changes to the source, then just don't try to decide something through 'the original bit depth was this'
[14:42:25 CEST] <zhanshan> JEEBsv "same" in that case I guess means to not loss too much of the original holyness of sound and of originality
[14:42:29 CEST] <zhanshan> :P
[14:42:38 CEST] <JEEBsv> well you already did some filtering to it, right? otherwise it wouldn't be in float
[14:42:48 CEST] <JEEBsv> thus that is now a completely different source
[14:42:54 CEST] <zhanshan> chungy yes that's right for ffmpeg, but in blender I didn't find that export option
[14:43:06 CEST] <theeboat> Hello, im using ffmpeg to capture video from a multicast stream. Does anybody know if its possible to force a black frame when there are no frames from the multicast stream?
[14:43:08 CEST] <zhanshan> that's why I shouldn't export anything from blender I guess
[14:43:16 CEST] <zhanshan> but use Ardour instead
[14:43:22 CEST] <chungy> do you have your original source?
[14:43:33 CEST] <zhanshan> but that workflow didn't establish in my work routine so far
[14:43:46 CEST] <zhanshan> I don't really know how to connect both programs
[14:44:02 CEST] <zhanshan> I can use/show video in Ardour and I can show audio in Blender..
[14:44:43 CEST] <zhanshan> chungy the original source are maybe 15 different files and they've been edited (i.e. cut, fading etc.) in blender
[14:44:54 CEST] <zhanshan> that's why I don't like to start from the beginning right now
[14:45:05 CEST] <zhanshan> chungy yes I have them
[14:47:43 CEST] <zhanshan> JEEBsv:  "well you already did some filtering to it, right? otherwise it wouldn't be in float" what has happened there. is it something bad that happens if I work and save like: signed->float->signed?
[14:47:46 CEST] <zhanshan> more noise?
[14:49:34 CEST] <zhanshan> chungy I like flac a lot but many programs can neither show the length of the flac-file, nor seek them. That's why this time I chose wav_pcm. I have to check the lengths in time of the files easily
[14:50:05 CEST] <chungy> should be irrelevant inside a mkv container
[14:52:49 CEST] <zhanshan> chungy yes, but I have to re-check the files before encoding
[14:55:32 CEST] <c_14> zhanshan: don't work with float if you don't have to
[14:57:06 CEST] <zhanshan> c_14 all this knowledge about the different wav forms really goes over my understanding right now. my head is full of video editing, scenes, composition...
[14:57:50 CEST] <c_14> If you're working with wav, it's pretty simple. Stay away from float, don't decrease the bit depth, and try not to mess around with the samplerate.
[14:58:35 CEST] <zhanshan> so keep inside the same specs for all time..
[14:58:56 CEST] <c_14> If you do that your audio should stay lossless.
[14:59:25 CEST] <zhanshan> technically that sounds resonable
[14:59:59 CEST] <zhanshan> but i.e. I recorded files in 24bit signed, blender can only export 16s or 32f/s
[15:00:01 CEST] <zhanshan> or flac
[15:00:11 CEST] <zhanshan> then I should choose preferrably flac?
[15:00:55 CEST] <c_14> IF you care about it staying lossless 32s or flac.
[15:01:14 CEST] <c_14> You probably won't notice the difference between 24bit and 16bit signed pcm though
[15:07:36 CEST] <zhanshan> thanks, I appreciate you answer!
[15:20:52 CEST] <fiacha> Hi, question, is it possible to process/request only the video of an RTP input stream? I have an IP cam and I want to dump the video but it also sends audio with lots of "Non-monotonous DTS in output stream" errors even though it doesn't have a mic
[15:28:29 CEST] <godofgrunts> I don't have any experience with rtp streams, but can you not just pass -an?
[15:50:37 CEST] <godofgrunts> fiacha, I just tried it with an rtp stream and it worked for me
[17:07:23 CEST] <fiacha> godofgrunts: yes, -an works, thank you
[20:52:42 CEST] <fred1807> why some files I convert from mp4 to h264 result in bigger files , and sometimes result in smaller sizes?
[20:52:47 CEST] <fred1807> from .mp4 to raw .h264
[20:53:25 CEST] <fred1807> and why mediainfo cannot read bitrate information from raw .h264 I create with ffmpeg?
[20:59:51 CEST] <JEEBsv> well, it probably doesn't want to parse all of the pictures to get a picture count, and then try to note if there's any frame rate kind of flag around
[21:00:52 CEST] <JEEBsv> because you would have to a) calculate how many pictures there are all in all b) get all the possible information on how fast the pictures are supposed to be shown c) use the file size of the raw stream on that
[21:01:01 CEST] <JEEBsv> anyways, I wouldn't trust mediainfo too much on various stuff
[21:01:14 CEST] <JEEBsv> you have to know how mediainfo comes to various conclusions to trust it
[21:01:33 CEST] <JEEBsv> because then you know what kind of information or guesses it is giving to you
[21:22:29 CEST] <Bleakwise> can someone tell me where ffdshow's avisynth plugin folder is by default?
[21:22:57 CEST] <JEEBsv> aand that has nothing to do with FFmpeg
[21:33:14 CEST] <anoop_r> hi
[21:34:58 CEST] <anoop_r> ffmpeg -i hd.mp4 -strict experimental -c:v libx265 -c:a -c:a libfdk_aac -b:a 320k output.mkv
[21:35:34 CEST] <anoop_r> if i use like this , does the video and audio is encoded concurrently
[21:35:39 CEST] <DHE> why did you put "-c:a" twice?
[21:35:46 CEST] <anoop_r> sorry
[21:35:54 CEST] <anoop_r> ffmpeg -i hd.mp4 -strict experimental -c:v libx265 -c:a libfdk_aac -b:a 320k output.mkv
[21:37:26 CEST] <anoop_r> does this encode audio and video simultaniously ?
[21:39:49 CEST] <DHE> you mean multi-threaded?
[21:40:09 CEST] <anoop_r> yes
[22:04:05 CEST] <Nolski> Hey, if I'm using the overlay filter and I want the overlaid video to start 30 seconds into the input video do I just do ffmpeg -ss 00:00:30 -i input.mkv -complex_filter "overlay filter goes here" out.mkv?
[22:04:30 CEST] <Nolski> or do filters not really take into account whether you scan to a position or not?
[22:05:16 CEST] <c_14> Do you want the overlay to not have the first 30 seconds of video, or do you want the overlay to show up 30 seconds into the main video?
[22:06:20 CEST] <Nolski> show up 30 seconds into the video
[22:06:26 CEST] <Nolski> c_14: ^
[22:06:42 CEST] <c_14> use the enable argument to the overlay filter
[22:06:55 CEST] <c_14> https://ffmpeg.org/ffmpeg-filters.html#Timeline-editing
[22:07:16 CEST] <Nolski> ah right, I always forget about the enable argument since it's never right under the filter options in the docs
[22:07:21 CEST] <Nolski> thanks :)
[23:35:37 CEST] <RobertNagy> I'm a bit confused by ffmpeg while encoding opus into webm. I have a wav file with exactly 240000 samples @ 48kHz. When i encoding it: ffmpeg -i 0.wav -c:a opus 0.webm. The resulting file gives me the following information: Duration: 5.01, Start 0.007, discard 648/900 samples, 240312 samples decoded.
[23:36:04 CEST] <RobertNagy> If I have several of these files how would I play them seamlessly without gaps?
[23:37:12 CEST] <RobertNagy> i.e. in a browser I've tried: sourceBuffer.timestampOffset = 5 * n - 0.007; sourceBuffer.appendWindowStart = 5 * n; sourceBuffer.appendWindowEnd = 5 * (n+1);
[23:37:22 CEST] <RobertNagy> however, there are audible gaps.
[23:37:52 CEST] <RobertNagy> How many samples am I actually supposed to discard? 0.007 * 48000, 648, or 240312 - 240000?
[23:40:49 CEST] <adamellsworth> can ffmpeg output SVG files?
[23:41:13 CEST] <c_14> no
[00:00:00 CEST] --- Thu Jul  2 2015

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