[Ffmpeg-devel-irc] ffmpeg.log.20180814

burek burek021 at gmail.com
Wed Aug 15 03:05:02 EEST 2018


[00:01:01 CEST] <DHE> in what style? metadata? burned-in subtitle?
[00:04:00 CEST] <ahoo> like the oldscool cameras did
[00:04:16 CEST] <ahoo> a static title, if you will
[00:04:29 CEST] <ahoo> but dynamic enough to display the current system time
[02:40:50 CEST] <Cracki> ahoo, https://einar.slaskete.net/2011/09/05/adding-time-stamp-overlay-to-video-stream-using-ffmpeg/
[02:41:02 CEST] <Cracki> see "addendum", apparently you can use format strings
[02:41:35 CEST] <Cracki> https://ffmpeg.org/ffmpeg-filters.html#drawtext_005fexpansion
[04:33:43 CEST] <ahoo> Cracki: thx! works like a charm.
[04:33:50 CEST] <Cracki> good good
[06:24:58 CEST] <mijofa_> So I don't really know of a better place to ask something like this, I'm looking at doing a web app with streaming media integrated and Chromecast support. Should I be targetting MPEG-DASH or HLS?   As far as I can tell, MPEG-DASH is uncommon but an official standard, however HLS is more commonly used/supported and effectively the unofficial standard
[06:25:40 CEST] <furq> hls is more common for live stuff because iOS refuses to support dash
[06:25:50 CEST] <furq> but yeah, hls
[06:29:44 CEST] <mijofa_> That's the kind of thing I was thinking. Primarily targetting Chrome on desktops, which doesn't natively support either of them
[06:30:24 CEST] <furq> that's why we have hls.js
[06:31:19 CEST] <mijofa_> Yeah, certainly looking like HLS is the way to go, thanks for the 2nd opinion. :)
[09:00:27 CEST] <freedrull> i'm trying to get a movie playable on iphone...this isn't working. any tips ? https://gist.github.com/mcfiredrill/e65c409ce9e8cc71825f23e5da265716
[09:01:06 CEST] <ritsuka> mp3 won't work, you should use aac
[09:01:26 CEST] <freedrull> ah
[09:01:59 CEST] <freedrull> trying swapping libmp3lame for aac .....
[09:05:26 CEST] <ritsuka> you can use high profile too, no need to limit it to baseline, if your iPhone is not 7 years old
[09:09:49 CEST] <freedrull> -profile:v high ?
[09:35:20 CEST] <Nacht> Anyone know a good place where I can find some documentation regarding the PTS/DTS and PCR and their relationship ? I know that PTS/DTS is relative to PCR. But I'm having trouble figuring out how the PCR is generated and what the values mean and how one can deduct them
[10:06:56 CEST] <meepmeep> hello :)
[10:16:07 CEST] <meepmeep> I try to find the possible values for compression_level on hevc_vaapi.
[11:25:24 CEST] <kubast2> Hey ,how can I stream audio that is outputed via an audio jack to my pc ,and how do I receive it? So far I came up with: 'ffmpeg -f pulse -i default -c:a libopus -f mpegts udp://192.168.0.7:10000' ,where 192.168.0.7 is the ip address I want to stream into/it is the receiving end ;and 192.168.0.5 is the device I want to stream from
[11:26:10 CEST] <kubast2> I am not sure if I am doing it right at all
[11:28:28 CEST] <kubast2> I did tried saving it to file first and set the right "input" for the ffmpeg pulseaudio recorder
[11:35:12 CEST] <barhom> Transacoding an input into adaptive HLS, with two different video bitrates and two different audio bitrates works as expected. What I have been unable to do is to transcode the audio once and reuse the encoded audio with both video bitrates.
[11:35:19 CEST] <barhom> Any thoughts on how this can be achieved?
[11:49:32 CEST] <Mavrik> Not easily. Also is it worth it?
[13:48:00 CEST] <Accord> hey, I'm recording on windows from a HDMI recorder over USB with dshow
[13:48:27 CEST] <Accord> I've set x264 to ultrafast with crf 21 and a rtbufsize of 1000M
[13:49:19 CEST] <Accord> it seems to work well, the segments ffmpeg outputs are good quality and seem to be realtime, the speed that ffmpeg reports is 0.998x + and the CPU stays below 90% utilization
[13:49:33 CEST] <Accord> but after a few minutes the 1 gigabyte buffer starts to get full
[13:49:46 CEST] <Accord> how is this possible? if x264 encodes at near 1x speed
[13:52:32 CEST] <DHE> multi-threading isn't perfect. it's possible that 90% utilization is still fully saturated...
[13:54:39 CEST] <Accord> I think the scene changed, it was more dynamic and x264 had a harder time encoding and that's what's caused it
[13:54:47 CEST] <Accord> I'll set a worst crf
[14:18:44 CEST] <Zgrokl> I'm reading the doc but not understand how to use -map ? How can i map all stream except 0:5 --> 0:7
[14:19:05 CEST] <Zgrokl> -map 0:1 -map 0:2 --> 0:4 not working
[14:23:35 CEST] <Zgrokl> -map 0:1 -map 0:2 -map 0:3 -map 0:4 -c copyoutput.ts
[14:23:51 CEST] <DHE> you forgot 0:0
[14:23:52 CEST] <Zgrokl> error : could not find codec parameters for stream 5
[14:24:04 CEST] <Zgrokl> i don't want stream 5
[14:25:12 CEST] <Zgrokl> -map 0:0 -map 0:1 -map 0:2 -map 0:3 -map 0:4 -c copyoutput.ts same error : could not find codec for stream 5
[14:25:29 CEST] <Zgrokl> if i do -map 0:0 it's working
[14:28:58 CEST] <Zgrokl> Ah it's just a warning i get an error for -map 0:4 because it's a subtitle
[14:29:36 CEST] <Zgrokl> Is it possible to stream subtitle ?
[14:30:07 CEST] <DHE> you probably just want to copy them
[14:36:08 CEST] <Zgrokl> DHE, Yeah, i do -map 0:4 -c copy but i got this error : Exactly one WebVTT stream is needed, Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
[14:59:21 CEST] <Zgrokl> Is there a way to extract a dvb teletext stream ?
[15:51:28 CEST] <th3_v0ice> Can anyone explain what RTMP buffers are doing internally? Thanks!
[15:57:49 CEST] <th3_v0ice> When I set the rtmp_buffer to 100s what happens with packets when I send them to the muxer?
[20:19:15 CEST] <barhom> Imagine I want to transcode a video using x264 but want two different outputs, one with aac audio and one with mp3 audio. How would I go about to make sure I only do the transcode process once?
[20:20:50 CEST] <Cracki> transcode each track separately, then mux to taste
[20:21:04 CEST] <furq> the tee muxer can do it
[20:21:13 CEST] <Cracki> there might even be a way to specify multiple output files, each with different streams mapped
[20:21:26 CEST] <furq> i would probably just write the mp3 track separately and then mux the second file
[20:21:31 CEST] <Cracki> https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs#Teepseudo-muxer
[20:23:03 CEST] <furq> -map 0:v -map 0:a -map 0:a -c:v libx264 -c:a:0 aac -c:a:1 libmp3lame -f tee "[select=\'v,a:0\']foo.mp4|[select=\'v,a:1\']bar.mp4"
[20:23:12 CEST] <furq> if you really want to do it all in one process
[21:13:40 CEST] <Bombo> i know ffmpeg keeps the metadata tags, but how can i keep the cover art?
[21:13:43 CEST] <Bombo> Stream #0:0: Audio: flac, 44100 Hz, stereo, s32 (24 bit)
[21:13:43 CEST] <Bombo> Stream #0:1: Video: mjpeg, yuvj444p(pc, bt470bg/unknown/unknown), 600x600 [SAR 72:72 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
[21:14:05 CEST] <Bombo> i tried ffmpeg -y -i ren.flac -sample_fmt s16 -c:v copy recode.flac
[21:14:22 CEST] <Bombo> the tags are there but no cover
[21:15:10 CEST] <Bombo> tried adding -map 0:0 -map 0:1
[21:15:21 CEST] <Bombo> [flac @ 0x5563da2adce0] only one stream is supported
[21:15:24 CEST] <Bombo> hm
[21:30:36 CEST] <poutine> I know with ffmpeg I can adjust subtitles, like ffmpeg -ss 00:01:00 -i in.srt -c:s text out.srt, would offset the subtitles by 1 minute. I'm curious, how could I set the starting timestamp doing this? For video I'd look at setpts filter, but not sure how that'd work with subtitles
[21:33:37 CEST] <durandal_1707> there is no infrastructure yet for subtitle filtering
[21:34:22 CEST] <poutine> is there any roadmap anywhere? I actually think this might be something I could help with
[21:35:01 CEST] <Bombo> hmm 'metaflac --import-picture-from cover.jpg recode.flac' does add a cover to the flac but does that work with ffmpeg too?
[21:35:04 CEST] <Bombo> ffmpeg -y -i ren.flac -sample_fmt s16 -i cover.jpg -c:v copy -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (front)" -map 0:0 -map 1:0 recode.flac
[21:35:13 CEST] <durandal_1707> nope, just some rumors, ask ubitux. he is working on it
[21:35:14 CEST] <Bombo> didn't work
[21:35:24 CEST] <Bombo> [flac @ 0x563454fd7980] only one stream is supported
[21:39:37 CEST] <furq> Bombo: https://trac.ffmpeg.org/ticket/4442
[21:41:46 CEST] <Bombo> furq: ;/
[21:41:48 CEST] <Bombo> but https://trac.ffmpeg.org/ticket/4448#comment:1
[21:42:35 CEST] <furq> just use the flac cli
[21:48:15 CEST] <Bombo> or metaflac
[21:48:16 CEST] <Bombo> $ metaflac ren.flac --export-picture-to -|metaflac recode.flac --import-picture-from -
[21:48:19 CEST] <Bombo> recode.flac: ERROR: while parsing picture specification "-": error opening picture file
[21:48:44 CEST] <Bombo> why does export have stdin and import doesn't... poo
[21:49:06 CEST] <furq> i meant for transcoding but sure
[21:49:13 CEST] <furq> unless you specifically want to use ffmpeg's flac encoder
[21:56:23 CEST] <Bombo> i need that for opus too
[21:56:41 CEST] <Bombo> so would be perfect if ffmpeg could do that
[21:56:58 CEST] <Bombo> there's no metaopus
[21:57:35 CEST] <TD-Linux> opusenc will keep coverart
[21:58:11 CEST] <Bombo> flac -dc ren.flac -|opusenc - ren.opus
[21:58:21 CEST] <Bombo> no metadata
[22:01:03 CEST] <TD-Linux> opusenc reads flac
[22:01:08 CEST] <Bombo> oh
[22:01:09 CEST] <Bombo> opusenc ren.flac ren.opus
[22:01:12 CEST] <Bombo> did work
[22:01:13 CEST] <Bombo> ;)
[22:01:42 CEST] <kepstin> Regarding copying pictures and other tags - I honestly just write a python script using mutagen library when I need to edit audio file tags.
[22:02:54 CEST] <Hello71> TD-Linux: not on Gentoo!
[22:02:58 CEST] <Hello71> (it's optional)
[22:03:48 CEST] <Bombo> but mpv doesn't show the artist/title/etc tags now, just the cover, lol
[22:04:05 CEST] <rjeli> I saved a bunch of libav packets without dts or pts. I assume the dts is monotonically increasing, but is there any way to recover the pts to make it watchable?
[22:04:15 CEST] <Bombo> encoded with ffmpeg, i get artist/title/etc but no cover...
[22:04:48 CEST] <kepstin> rjeli: if you know the framerate of the original video, making up some pts values is easy.
[22:05:05 CEST] <kepstin> rjeli: you're gonna run into problems if there were B frames in the stream, since then the packets might be out of order
[22:05:37 CEST] <kepstin> no real way to recover from that :/
[22:05:47 CEST] <rjeli> libx264 will have b-frames, right?
[22:06:07 CEST] <kepstin> it usually will (it's an optional feature, sometimes turned off for realtime/high speed encoding)
[22:06:31 CEST] <rjeli> okay, that's what i thought :( thanks!
[22:07:17 CEST] <kepstin> I mean, maybe you could do some computer vision analysis on the decoded frames to figure out which order they should be in based on similarities? :)
[22:07:45 CEST] <rjeli> easier just to go out and rerecord my logs :)
[22:07:47 CEST] <kepstin> there might be something you can do that I simply don't know about either.
[22:08:14 CEST] <rjeli> if no b-frames, I would just do dts, pts = 0, 1, 2, 3 ... right?
[22:08:42 CEST] <kepstin> yeah, and set the timebase to the inverse framerate.
[22:46:37 CEST] <sakrecoer> greetings! i have a video of pixel art at 128x72 pixels. I'm trying to figure out how to scake it to HD without anti-aliasing and at this point my websearcg-fu has failed me miserably. Any hint would be greatly appreciated.
[22:47:16 CEST] <sakrecoer> basically scale it up from 128x72 to 1280x720 but keeping the pixels as sharp as possible
[22:48:17 CEST] <kepstin> sakrecoer: -vf scale=w=1280:h=720:flags=neighbor
[22:48:38 CEST] <sakrecoer> \o/ thank you kepstin!!!!!
[22:49:14 CEST] <kepstin> (there's a few ways to do that, but this is the most explicit)
[22:49:19 CEST] <ariyasu> kepstin if you don't specify a scaler, whats the default?
[22:49:28 CEST] <kepstin> bicubic, i think
[22:50:04 CEST] <kepstin> see the 'ffmpeg-scaler' docs for details
[22:50:13 CEST] <ariyasu> ok, ty
[22:50:47 CEST] <sakrecoer> thank you very much! :)
[22:52:57 CEST] <kepstin> for some types of pixel art, you might consider using the 'super2xsai' filter instead, which tries to smooth diagonals and whatnot.
[22:53:27 CEST] <kepstin> depends on content whether you'd want that or not.
[23:56:35 CEST] <Maverick|MSG> anyone run into an issue with the windows build helpers cross compiler throwing a "ERROR: libmfx not found" when the compile is almost done?
[23:57:05 CEST] <Maverick|MSG> I keep getting it but can't find a way around it
[00:00:00 CEST] --- Wed Aug 15 2018


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