[Ffmpeg-devel-irc] ffmpeg.log.20180513

[burek]

[00:01:16 CEST] <faLUCE> hello. Given myfile.mp4,  how can I CUT video (while preserving audio)  from 00:01:03  to 00:01:10   ?
[00:31:47 CEST] <klaxa> faLUCE: use -ss and -t or -ss and -to ?
[00:33:39 CEST] <faLUCE> klaxa: sorry, I wrote a wrong question. I want to DELETE video from 00:01:03 to 00:01:10  (without re-encoding)
[00:34:07 CEST] <klaxa> use -ss and -t and concat?
[00:35:41 CEST] <faLUCE> klaxa: I tried  ffmpeg -i "concat:la-vestale-video-part1.mp4|la-vestale-video-part2.mp4" -c copy lavestalesolovideo.mp4   but the result is only the first part
[00:36:47 CEST] <klaxa> hmm... can you try using .ts as intermediate instead of .mp4?
[00:37:07 CEST] <faLUCE> ok, let's try
[00:40:54 CEST] <c_14> you want to delete video while preserving audio?
[00:40:57 CEST] <c_14> so you want it to be blank?
[00:41:07 CEST] <c_14> just overlay with black?
[00:44:24 CEST] <faLUCE> thanks klaxa, it worked with ts.   c_14 I want to add blank video from 00:01:03 to 00:01:10
[00:44:31 CEST] <faLUCE> black video, sorry
[00:45:04 CEST] <faLUCE> well, the problem is that the video, during this time, has some problem, so it looses sync with audio
[00:45:16 CEST] <c_14> ah
[00:45:21 CEST] <c_14> then klaxa's version is probably better
[00:45:23 CEST] <faLUCE> so I don't know what to do exactly, I'm trying to delete this part
[00:53:19 CEST] <faLUCE> what if I want to REPLACE audio from mymovie.flv   with myaudio.mp3  ?
[00:55:19 CEST] <c_14> all of it?
[00:55:27 CEST] <faLUCE> c_14: yes
[00:55:37 CEST] <faLUCE> both audio have same duration
[00:55:42 CEST] <c_14> just have 2 inputs and use -map
[00:56:14 CEST] <faLUCE> so, ffmpeg -i mymovie.flv -i myaudio.mp3 .....  output.flv ?
[00:56:47 CEST] <c_14> yeah, but with -map 0:v and -map 1:a
[01:05:21 CEST] <faLUCE> thanks c_14
[09:08:51 CEST] <th3_v0ice> Hi guys, I have been using ffmpeg c API for some time, but recently noticed that last frame is not written to the output. I tried sending the NULL packet to the av_interleaved_write_frame() but that doesnt flush the packet, even tho I have saved all of the frames as pictures and they are all accounted for. What could be the problem? Thanks!
[09:11:40 CEST] <rindolf> th3_v0ice: hi
[09:12:13 CEST] <rindolf> th3_v0ice: did it happen in previous ffmpeg versions?
[09:13:01 CEST] <th3_v0ice> rindolf: I am currently using 3.4.2 version. My last resort is to use the 4.0 but that requires updating a lot of code.
[09:13:31 CEST] <th3_v0ice> I am not sure if it was happening in the previous versions. I started with 3.4.2
[09:16:52 CEST] <rindolf> th3_v0ice: ah
[09:16:52 CEST] <th3_v0ice> rindolf: It shows as that the packet was written. But decoding the output mp4 file into the YUV shows 1 frame less.
[09:17:12 CEST] <rindolf> th3_v0ice: ok
[09:33:12 CEST] <th3_v0ice> rindolf: Any hints as to what may be the problem?
[09:33:26 CEST] <rindolf> th3_v0ice: no idea
[09:34:13 CEST] <th3_v0ice> rindolf: Ok, thanks anyway :)
[09:34:25 CEST] <rindolf> th3_v0ice: you're welcome
[12:35:28 CEST] <spood> Allo
[12:36:20 CEST] <spood> Hi
[14:41:10 CEST] <faLUCE> Hello. I'm trying to replace the audio track in a audio+video file, with this command:   "  ffmpeg -i La_Vestale-704x400-832k.flv -i lavestaleuff.ac3 -vcodec copy -acodec copy -map 0:v -map 1:a la-vestale-sync.flv    "      ----->  but get an error:    http://paste.ubuntu.com/p/3kGk6CvWsd/    . What's wrong?
[14:42:26 CEST] <klaxa> >Audio codec ac3 not compatible with flv
[14:42:31 CEST] <klaxa> can't use ac3 in flv
[14:42:54 CEST] <faLUCE> klaxa: which codec can I use?
[14:43:02 CEST] <klaxa> mp3 i think
[14:43:14 CEST] <c_14> probably also aac
[14:43:27 CEST] <faLUCE> I tried with mp3 but got the same error
[14:43:38 CEST] <klaxa> also only 44.1 khz iirc
[14:43:52 CEST] <klaxa> flv is weird like that
[14:44:31 CEST] <faLUCE> in fact the problem is in the sample rate
[14:45:00 CEST] <klaxa> yeah
[14:45:12 CEST] <klaxa> even says so >[flv @ 0x55f823606ea0] FLV does not support sample rate 48000, choose from (44100, 22050, 11025)
[14:45:49 CEST] <JEEB> it supports misc sample rates with AAC
[14:45:53 CEST] <JEEB> but not with other formats
[14:46:02 CEST] <faLUCE> JEEB: so I can use AAC ?
[14:46:19 CEST] <JEEB> yes, the latest FFmpeg should let you put in AAC at 48kHz just fine
[14:46:23 CEST] <JEEB> into FLV
[14:46:28 CEST] <faLUCE> let's try
[14:46:31 CEST] <klaxa> oh? that's new
[14:46:39 CEST] <JEEB> not really new, it's been in the spec for quite a while
[14:46:51 CEST] <JEEB> for AAC they just say "write stereo, 44.1kHz" here for shits and giggles
[14:47:06 CEST] <JEEB> and tell the implementation to actually parse the data from the AAC bit stream headers
[14:47:20 CEST] <faLUCE> otherwise, how can I reasmple myfile.wav from 48kHz to 44Khz ?
[14:47:28 CEST] <JEEB> you don't need to resample with AAC
[14:47:31 CEST] <klaxa> ok that's new since the last time i did something with flv like 4 years or so ago
[14:47:44 CEST] <faLUCE> JEEB: my ffmpeg version is of some months ago
[14:47:58 CEST] <furq> 48k aac in flv has worked for ages
[14:48:07 CEST] <faLUCE> ok
[14:48:40 CEST] <JEEB> yea, actually I think it's been specified proper in FLV for quite longer than when FFmpeg improved the check
[14:48:49 CEST] <JEEB> we just noticed at some point that the check was bullshit
[14:48:52 CEST] <furq> i meant in ffmpeg but yeah
[14:49:05 CEST] <JEEB> yea even in FFmpeg it was quite a few years ago IIRC
[14:50:30 CEST] <furq> faLUCE: i take it you actually need to use flv
[14:50:35 CEST] <furq> presumably for rtmp
[14:51:54 CEST] <JEEB> and yea, it seems like this might have gotten fixed in flvenc back in 2008
[14:52:03 CEST] <JEEB> which is from before git
[14:52:06 CEST] <furq> nice
[14:52:26 CEST] <furq> i knew it was at least 2012 because i was using it pretty regularly then
[14:53:19 CEST] <JEEB> but yes, everything else than MP3 and AAC seems to get the boot with 48kHz
[14:53:36 CEST] <furq> it doesn't support any other worthwhile audio codecs anyway
[14:53:45 CEST] <JEEB> yes
[14:53:58 CEST] <JEEB> and adobe making no additions to it make it even sadder than companies have semi-standardized to it as ingest
[14:54:25 CEST] <JEEB> well, I am not really 100% sure if smooth streaming is much better, but at least MS specified HEVC in it, har har har
[14:54:29 CEST] <JEEB> and netflix VP9
[14:54:55 CEST] <JEEB> although it seems like there's interest for MPEG-TS in UDP and proper fragmented isobmff for ingest
[14:55:04 CEST] <JEEB> latter through HTTP
[14:55:12 CEST] <furq> yeah it's annoying flv doesn't support ac3
[14:55:13 CEST] <JEEB> like, long-going eternal POST kind of things
[14:55:24 CEST] <furq> but i guess there's not even any procedure by which you'd get that added
[14:55:34 CEST] <furq> other than "just start doing it and hope everyone else notices"
[14:55:37 CEST] <furq> the apple technique
[14:55:42 CEST] <JEEB> you could prod Adobe but they're most likely not interested
[14:55:45 CEST] <furq> right
[14:55:51 CEST] <JEEB> and so far we said nope at some random codec id being added for HEVC by the chinese
[14:56:02 CEST] <JEEB> because seriously, if you want to still use FLV - go to adobe
[14:56:07 CEST] <JEEB> and make them specify it
[14:56:18 CEST] <JEEB> because otherwise in theory since it's just an integer ID
[14:56:23 CEST] <JEEB> they could use that for something else
[14:56:30 CEST] <JEEB> it's unlikely but more likely than with strings
[14:57:28 CEST] <JEEB> also looking at the history of flvenc it seems like it briefly had a feature to mux whatever, but then that was very quickly reverted. and thank goodness for that
[14:57:36 CEST] <JEEB> I don't even want to think about how that worked :P
[15:52:56 CEST] <acresearch> people i am trying to understand how to record and export video for youtube. instead of just doing a recording and uploading, i want to understand what audio recording then compression parameters first (then when i understand it move on to video)    i am not sure where to go so i thought i ask here and someone might advise me which channel to ask? or an online resource that can educate me
[16:44:54 CEST] <klaxa> acresearch: https://trac.ffmpeg.org/wiki/Encode/AAC and https://trac.ffmpeg.org/wiki/Encode/H.264 may be good starting points
[16:45:03 CEST] <klaxa> also read stuff they link to you don't understand
[17:06:53 CEST] <acresearch> klaxa: thanks
[17:20:32 CEST] <faLUCE> I don't know how to solve that. In myvideo.flv, at 00:10:03 , video track freezes for one second and from then, audio and video go out of sync. How can I correct the sync problem?
[17:21:15 CEST] <faLUCE> should I use a video editor for that, or I can obtain that result with ffmpeg ?
[18:49:53 CEST] <djk> Any hints on why an rtmp t facebook live would get " Input/output error"
[18:50:24 CEST] <JEEB> -v verbose and the error usually comes before that. the end result is an IO error
[18:53:39 CEST] <djk> RTMP_ReadPacket, failed to read RTMP packet header
[18:53:59 CEST] <fooman2011> Hello. I'm tyring to debugging a problem and i'm stuck in the code. In the file hls.c there is a line: ret = s->io_open(s, pb, url, AVIO_FLAG_READ, &tmp);     And I don't find what is this io_open  function.   Could you please help me to understand ? Maybe you're more familar with this code.
[18:54:47 CEST] <fooman2011> it should be an http get request
[18:59:26 CEST] <djk> I guess the static builds don't include support of -v? I get verbose info with a local compile of 2.8 but the 4.0 that I downloaded doesn't show any verbose.
[19:00:15 CEST] <JEEB> uhh, that would require a very special way of doing stuff :P
[19:00:23 CEST] <JEEB> the thing is, though, that -v requires a parameter
[19:00:27 CEST] <akk> Hi -- I'm trying to make an animation from a series of images named yyyy-mm-dd.png, e.g. 2012-09-24.png
[19:00:28 CEST] <JEEB> so -v PARAMETER
[19:00:32 CEST] <JEEB> -v verbose for example
[19:00:42 CEST] <akk> Is there any way to do that short of renaming them all to img0001.png, img0002.png etc.?
[19:01:38 CEST] <JEEB> fooman2011: without looking at the internal code the s is probably an internal avformat context because the hls thing is a "meta" muxer, and io_open is a set function pointer for whatever is being called
[19:01:46 CEST] <JEEB> or it is the AVIO context
[19:01:50 CEST] <JEEB> one or the other
[19:02:04 CEST] <JEEB> you should figure out what structure "s" is if you look upwards in the code of that function
[19:06:00 CEST] <djk> right I have -v verbose but the static build doesn't show the verbose details. odd probably me
[19:08:26 CEST] <JEEB> they could have been later moved to -v debug, or you have a different RTMP implementation going
[19:08:29 CEST] <fooman2011> JEEB: thanks
[19:08:31 CEST] <JEEB> (there's two available)
[19:10:08 CEST] <djk> two rtmp?
[19:10:45 CEST] <JEEB> the librtmp wrapper (older), and the internal rtmp implementation by wbs
[19:10:49 CEST] <djk> frustrating thing is I had a process working about a year and yesterday the process stopped working
[19:12:30 CEST] <djk> ok -v debug works now to figure out things
[19:12:32 CEST] <djk> thank you
[20:02:45 CEST] <fooman2011> Ok I think that there is a bug about cookies when you try to open a hls
[20:03:07 CEST] <fooman2011> the cookies are not correctly transmitted for all http request
[20:03:31 CEST] <fooman2011> I have identified one location where there are lost
[20:03:40 CEST] <fooman2011> but it seems to have at least another
[20:03:47 CEST] <fooman2011> any idea about this king of problem ?
[20:05:30 CEST] <fooman2011> king => kind ..
[21:41:58 CEST] <Asterisk> I have a weird issue with chaining filters together -- I'm getting different results doing two scale filters in sequence from what I get if I write to an intermediate file (losslessly) and run each filter in a separate ffmpeg command.
[21:45:53 CEST] <Asterisk> The input is RGB at 320x200, and my goal was to first upscale to 1600x1200 with a nearest-neigbor scalar, then downscale to 960x720 using bilinear.
[21:47:14 CEST] <Asterisk> See https://i.imgur.com/SIfpOXg.png -- the first column is from doing scale=1600:1200:sws_flags=neigbor, outputing with libx264rgb losslessly, then running scale=960:720:sws_flags=bilinear.
[21:48:10 CEST] <Asterisk> The second column is from a single instance of "scale=1600:1200:sws_flags=neighbor,scale=960:720:sws_flags=bilinear" -- if you zoom in, you'll see that it's full of artifacts.
[21:48:56 CEST] <Asterisk> The third column is just upscaling the source to 960x720 with bilinear without doing 1600x1200 first, so you can see why I want to do the two filters in sequence.
[21:50:24 CEST] <Asterisk> The first column is the output I want, so why do I get artifacting when I do both scale filters in sequence?  Is ffmpeg converting the pixel format differently in between the scale filters or something?
[22:35:40 CEST] <faLUCE> I don't know how to solve that. In myvideo.flv, at 00:10:03 , video track freezes for one second and from then, audio and video go out of sync. How can I correct the sync problem?
[22:35:42 CEST] <faLUCE> should I use a video editor for that, or I can obtain that result with ffmpeg ?
[23:04:18 CEST] <kepstin> Asterisk: when you're doing the command with both filters, what's the pixel format of the final output?
[23:04:56 CEST] <Asterisk> RGB
[23:04:57 CEST] <kepstin> Asterisk: if your input is rgb and output is yuv420p or something, the problem might be that it's doing the conversion too early.
[23:05:27 CEST] <Asterisk> I'm keeping everything in RGB and using libx264rgb for everything.
[23:05:53 CEST] <Asterisk> It happens even with -pix_fmt rgb0
[23:06:44 CEST] <Asterisk> But the input source (from DOSBox) is a capture of a 16-color EGA game, and it's actually using an 8-bit palette.
[23:07:28 CEST] <Asterisk> Still, the problem doesn't occur if I do the two scale filters sequentially in separate ffmpeg commands.
[23:08:23 CEST] <kepstin> Asterisk: can you try running the command with both filters with "-v verbose" and pasting the output here? That verbosity will make it log the format conversions in the filter chain
[23:08:48 CEST] <Asterisk> Sure, let me run it with verbose output.
[23:09:14 CEST] <kepstin> Asterisk: if the input is paletted, adding something like a "format=rgb24," at the start of the filter chain might fix it.
[23:10:35 CEST] <Asterisk> I'll try it with and without that using verbose output.
[23:14:17 CEST] <Asterisk> Here's the original command output: https://pastebin.com/raw/ZKqzgbUn
[23:14:44 CEST] <Asterisk> It looks like the RGB conversion *is* taking place in the middle of the filter chain.
[23:15:00 CEST] <kepstin> yeah, it's running the bilinear scale while doing the pal8 to rgb24 conversion, i could see that messing up
[23:15:11 CEST] <kepstin> my suggestion with adding a format to the filter chain should fix it
[23:15:24 CEST] <Asterisk> Let me try specifying the format as you suggested.
[23:16:22 CEST] <Asterisk> That did it.  Thanks!
[23:49:11 CEST] <kerio> can the palette be yuv instead of rgb?
[23:51:58 CEST] <kepstin> i don't think ffmpeg supports any paletted yuv formats.
[23:55:52 CEST] <kepstin> i'm not really sure it would be useful - the main benefit of yuv is that it makes compressing image planes separately more efficient
[23:56:37 CEST] <kepstin> but with paletted images, you don't have separate image planes, and the palette is typically not compressed, so... might as well just store the rgb and save a conversion.
[23:56:37 CEST] <JEEB> yea, and the RGB paletted things in FFmpeg seem to be quite far just an afterthought
[23:57:36 CEST] <JEEB> basically I wouldn't care too much about the palettes in stuff like DVD/DVB/PGS subpictures
[23:57:53 CEST] <JEEB> if you wanted raw YCbCr output I'd probably just give you 4:4:4 YCbCr instead of RGB
[23:58:18 CEST] <JEEB> RGB out of those decoders isn't paletted currently anyways
[23:58:42 CEST] <kerio> kepstin: i like having the luminance as a channel tho :<
[00:00:00 CEST] --- Mon May 14 2018