June 2019 - Ffmpeg-devel-irc - lists.ffmpeg.org

ffmpeg-devel.log.20190606
by burek 07 Jun '19

07 Jun '19

[03:46:47 CEST] <tmatth> trac seems to be unresponsive [06:50:12 CEST] <cone-615> ffmpeg 03Gyan Doshi 07master:3be449001442: avfilter/framesync: fix shortest with eof_action=pass [08:00:29 CEST] <cone-615> ffmpeg 03Xuewei Meng 07master:78e1d7f42110: libavfilter: Add derain filter [10:46:18 CEST] <j-b> 'morning [14:37:38 CEST] <vtorri99978666> hello [14:38:08 CEST] <vtorri99978666> is there a verbose mode for Makefile, so that only something like that : [14:38:19 CEST] <vtorri99978666> CC libavformat/foo.c [14:38:33 CEST] <vtorri99978666> is displayed, instead of the complete command line ? [14:39:14 CEST] <thardin> make V=1 perhaps [14:39:19 CEST] <JEEB> that yes [16:12:12 CEST] <jamrial> was planning on pushing the svt-hevc wrapper patch on the ml [16:12:25 CEST] <jamrial> should i put it after x265 in the priority order? [16:31:12 CEST] <jkqxz> Is the ABI stable yet? They seem to still be making changes with the same SONAME. [16:33:24 CEST] <jamrial> ah, good question [16:33:38 CEST] <jamrial> the configure check doesn't even look for a minimum version [16:33:57 CEST] <jamrial> better wait until 4.2 is tagged, at least [16:35:49 CEST] <vtorri99978666> thardin thank you [16:47:17 CEST] <jkqxz> jamrial: A while ago in response to the AV1 parser change you linked some patch here which simplified it somehow but I don't remember much about it. Any idea what that was? [17:28:05 CEST] <jamrial> jkqxz: that simplified the av1 parser? other than your frame parsing addition to cbs to avoid copying, i don't recall anything [17:28:28 CEST] <jamrial> oh, a patch on top of that cbs addition [17:28:33 CEST] <jamrial> yeah, let me see if i can find it [17:45:21 CEST] <jamrial> jkqxz: https://pastebin.com/gAwHsysR i think it was this [17:46:17 CEST] <jamrial> dunno if we could call it simplification, but it removes the call to cbs_av1_read_unit, tile data references and such [17:46:53 CEST] <jamrial> this was a quick implementation copying a lot of code from cbs_av1_read_unit(). i guess parts could be reused instead [22:46:45 CEST] <_bluez> Hey, I have just started working on a very simple heic muxer, with hevc as default encoding, how can I test this out after I write it? [22:47:11 CEST] <_bluez> Like can I convert a jpg image into hevc encoded bitstream? [22:47:38 CEST] <_bluez> and then maybe mux the bitstream? I am a bit confused... [22:54:30 CEST] <_bluez> can anyone help? [23:00:22 CEST] <JEEB> _bluez: you can use the ffmpeg.c tool for example if built with libx265 to encode HEVC [23:00:45 CEST] <JEEB> ffmpeg -i blah.jpg -c:v libx265 out.heic [23:03:24 CEST] <_bluez> @JEEB: Okay, thanks! [00:00:00 CEST] --- Fri Jun 7 2019

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ffmpeg.log.20190606
by burek 07 Jun '19

07 Jun '19

[00:33:28 CEST] <eschie> Hi all, I'm trying to troubleshoot an issue and am unsure where to start: When seeking past the video buffer of my HLS stream on iOS, the audio cuts out but the video continues. I am unable to sync them back up again unless I start the video over from the beginning. Is this a problem with framerates and segments? How can I compare why some videos do this, when others of my do not? [00:40:42 CEST] <CarlFK> another: mind reviewing my params: https://github.com/CarlFK/voctomix-outcasts/blob/8a70888036f29b34682f010775… [02:42:41 CEST] <CarlFK> Trailing options were found on the commandline. [02:43:07 CEST] <CarlFK> ah, missed a \ [02:57:54 CEST] <Yasuragikara> Hi everyone. I'm trying to build ffmpeg with standard configuration on a win7 host, using the current mingw-get base sets of mingw-dev, mingw32 (+pkg-config) (+libpthreadgc/e) and mingw32-g++ with msys (+yasm) as the make shell. [02:58:01 CEST] <Yasuragikara> Configuring and making ffmpeg in msys works out fine and I got the static .exe to use, but when I try to pass this cmd line (ffmpeg.exe -i a.aac a.wav) to ffmpeg, I receive "[aac @ 04350bd8] Warning: not compiled with thread support, using thread emulation" as output and the software crashes. Could someone please tell how I can fix this? Thanks in advance for all feedback. [03:02:17 CEST] <Yasuragikara> I'm using the source of ffmpeg version 4.1.3. [03:31:18 CEST] <anony11> Hi [03:31:56 CEST] <anony11> It's possible to output a buffer? Like ffmpeg -i continousrawdata filethatkeepsbeing400kb.webm [03:50:54 CEST] <void09> hi. trying to test the SVT-AV1 encoder, and their sample usage line with ffmpeg stream led me to the following: ffmpeg -i /scratchpad/Aferim.2015.1080p.BluRay.REMUX.AVC.DTS.HDMA.5.1-BiTHD.mkv -nostdin -f rawvideo -pix_fmt yuv420p - | ./SvtAv1EncApp -i stdin -n 200 -w 1920 -h 1080 -b out.bin [03:56:48 CEST] <void09> uh nvm i think it works [07:39:47 CEST] <pk08> Hi, [07:39:47 CEST] <pk08> i want to overlay 16 video streams into one video stream and output that video to some udp multicast. [07:39:47 CEST] <pk08> and i am doing this in two steps: 1) scale origianl input to fixed resolution video stream and 2) take input video stream of first step and overlay on nullscr filter and do udp multicast [07:39:49 CEST] <pk08> so in fist stepp i am using this command to scale video stream https://pastebin.com/Tt20J25N [07:39:51 CEST] <pk08> and i am talking above commands output for 2nd step as input for overlaying for final output, and command is https://pastebin.com/36bqQJQb [07:39:54 CEST] <pk08> now in 2nd step i am getting slow encoding speed like between 0.300x to 0.700x and thats why i am getting bad video output and even i am getting some errors like this: https://pastebin.com/v4sTQnXG [07:40:45 CEST] <pk08> can anyone please tell me if my command is wrong or bad input [07:40:56 CEST] <pk08> or what am i doing wrong in here [07:41:01 CEST] <pk08> thanks in advance [07:41:09 CEST] <pk08> thank [08:18:44 CEST] <CarlFK> pk08: I would start by reading and writing files - then you don't have to worry about performance [08:19:53 CEST] <CarlFK> also instead of 16, do 2. [08:20:04 CEST] <CarlFK> get that working, then you can do more [08:20:18 CEST] <pk08> CarlFK: i need real time input and output, so it need to be udp multicast [08:20:35 CEST] <pk08> and yes, i tried to add one by one [08:20:48 CEST] <CarlFK> pk08: you don't need real time while you are testing [08:20:49 CEST] <pk08> and i was able to get good output upto 10 inputs [08:21:23 CEST] <pk08> yes, correct [08:21:33 CEST] <pk08> it dont need to be realtime while testing [08:21:50 CEST] <pk08> but even in file, performance will be same [08:22:14 CEST] <pk08> and when i will do real time, it wont work as expectations [08:23:05 CEST] <pk08> you think ffmpeg filter_complex has bottleneck of input bitrate or something? [08:23:31 CEST] <CarlFK> I have no idea - I was here getting some help with deinterlacing flags [08:23:59 CEST] <CarlFK> it sounds like you have the commands right, but you are running out of some resource [08:26:49 CEST] <pk08> i think its ffmpeg side issue or i need some improvement in command because i have 32 cores and 64 gb of RAM so i dont think i have resources issue [08:27:26 CEST] <pk08> so not sure, what i need to do... [08:29:07 CEST] <CarlFK> could be memory bandwitch [08:29:21 CEST] <CarlFK> again... try files... see if it breaks [08:30:47 CEST] <CarlFK> 16 network streams is a bit of bandwidth, and maybe cores decompressing them [08:38:40 CEST] <pk08> may be you are right about network bandwidth, but need to be sure about it [08:38:56 CEST] <pk08> so let me test everything as file [08:39:08 CEST] <pk08> like all input and output in file format [08:39:19 CEST] <pk08> and check performance [09:38:14 CEST] <th3_v0ice> When is the value of ts_offset set in FFmpeg.c code? [09:57:39 CEST] <th3_v0ice> Found it. [11:20:31 CEST] <keglevich> hello all... [11:22:04 CEST] <keglevich> I have a question regarding ffmpeg concat demuxer... simple scenario: I try to join (concat) two files into single one...both are perfect constant 25fps... after concat the output is variable with minimum framerate (1fps - 20fps) and max framerate still 25fps... same issue is posted here https://trac.ffmpeg.org/ticket/7939 [11:22:28 CEST] <keglevich> why does this happen and what can I do to avoid it and get true constant 25fps end result without the need to re-encode again? [11:23:01 CEST] <keglevich> ffmpeg -f concat -safe 0 -i clip-list.txt -c copy clip-concat.mp4 [11:23:09 CEST] <keglevich> that's the command I'm using [11:38:15 CEST] <th3_v0ice> keglevich, Maybe try to produce those two input files with -stitchable, if you are using x264 for encoding that is. You can also encode input files with -stitchable and output to .h264. Concat those, that would work. [11:38:41 CEST] Last message repeated 1 time(s). [11:38:41 CEST] <th3_v0ice> Sorry for double posting. [11:42:32 CEST] <keglevich> th3_v0ice: hmm, what about if there files are already done and I wouldn't like to reencode them again? [11:43:07 CEST] <keglevich> all the files I have are already mp4, libx264, aac, 25fps ... same resolution, same ar... I'd just like to join them somehow and get 25fps result [11:44:31 CEST] <keglevich> meybe it's worth mentioning, I tried the option with creating two .ts files as described here https://trac.ffmpeg.org/wiki/Concatenate (Using intermediate files) and the result was correct in the end...but I'm not sure, if this method would work for other formats as well (for instance libx264 with mp2 audio?) [12:10:08 CEST] <keglevich> and again... also tried with different .ts files and the final output sometimes is correct and sometimes is variable... so this option also doesn't work as expected [12:33:36 CEST] <anony11> Hi, how to: "ffmpeg -i rawdatastream streamwebmof2seconds.webm". Regards [12:34:45 CEST] <pk08> CarlFK: i have tried to take all inputs as file format and output as file too, and it still gives me 0.7x encoding speed [12:35:14 CEST] <pk08> and to make 5 mins output video, its takes 6:30 mins [12:35:39 CEST] <pk08> so its something related to ffmpeg side issue [12:36:24 CEST] <anony11> Oh, to be clear: the input is ok, only the output(to use as little size as possible) [12:37:31 CEST] <anony11> Like: "ffmpeg -i rawdatastream -fs 200k streamwebmof2seconds.webm". Regards [12:39:20 CEST] <pk08> can anyone please tell me, how can i increase encoding speed in this command https://pastebin.com/36bqQJQb , I am getting around 0.5 to 0.7x speed while encoding [12:39:40 CEST] <anony11> platform? [12:43:15 CEST] <a_c_m> Hi [12:51:13 CEST] <a_c_m> I have a perhaps interesting use case, which i can't quite get to work https://pastebin.com/b03e9URc - TL:DR; i want to start and stop a stream, but with the output not knowning i did. Any one any ideas on how i might do this? I have a basic idea working, but its not right. [12:53:03 CEST] <anony11> lmao [12:53:49 CEST] <a_c_m> @anony11 - :) to who's question ? [12:54:24 CEST] <anony11> Haha yours [12:54:58 CEST] <a_c_m> :) Because it brain dead simple to fix? (i hope :) ) or some other reasons? [12:56:33 CEST] <anony11> Here [12:56:34 CEST] <anony11> https://imgur.com/L7rtCvb [12:57:54 CEST] <a_c_m> lol - someone else asked the same thing? woah. and there was me thinking it was novel :) - So your saying pipe the output to another instance of ffmpeg and all will be well? [12:58:32 CEST] <anony11> Tought of that, but > and after that? You have to continue to pipe [12:59:10 CEST] <a_c_m> will give it a try [12:59:33 CEST] <anony11> People suggested to: make a little program, a while loop, but then, since streaming, the file "changes", and it's not a buffered output [12:59:49 CEST] <a_c_m> i'm actually doing it all from node - so controlling the streaming output is easy to to control [13:00:46 CEST] <a_c_m> its just the timecodes thats the problem - as it confuses the players into showing nothing / the last frame for that "paused" section [13:01:14 CEST] <anony11> I'm above, doing in C haha [13:05:56 CEST] <Yasuragikara> @pk08 Check this out: https://pastebin.com/Bj6TjqSL [13:06:17 CEST] <Yasuragikara> Maybe this can help you. [13:54:51 CEST] <anony11> Regards [14:13:26 CEST] <pk08> @Yasuragikara thank you, let me test it, i will get back with results [14:14:17 CEST] <furq> that's not going to work [14:14:19 CEST] <furq> -vf isn't per-input [14:14:50 CEST] <furq> pk08: keep the filterchain you already had but replace all the overlays with one xstack [14:14:58 CEST] <furq> https://ffmpeg.org/ffmpeg-filters.html#xstack [14:15:37 CEST] <pk08> furq: yes, i know [14:15:50 CEST] <pk08> thank you :) [14:16:46 CEST] <pk08> but this is temporary fix, even if it works [14:17:24 CEST] <pk08> because in future i will need different resolution's video as input [14:17:43 CEST] <pk08> and stack filter work for only same resolution [14:18:03 CEST] <pk08> does anyone know any alternative or fix for that? [14:18:18 CEST] <furq> change fifo,setpts to fifo,setpts,scale [14:19:07 CEST] <furq> also you can put newlines in filterchains (or load the filterchain from a text file) so it's probably worth doing that for readability [14:19:43 CEST] <keglevich> I have a question regarding ffmpeg concat demuxer... simple scenario: I try to join (concat) two files into single one...both are perfect constant 25fps... after concat the output is variable with minimum framerate (1fps - 20fps) and max framerate still 25fps... same issue is posted here https://trac.ffmpeg.org/ticket/7939 [14:20:10 CEST] <keglevich> why does this happen and what can I do to avoid it and get true constant 25fps end result without the need to re-encode again? [14:20:20 CEST] <keglevich> ffmpeg -f concat -safe 0 -i clip-list.txt -c copy clip-concat.mp4 [14:20:30 CEST] <keglevich> this is the command I was using [14:22:33 CEST] <pk08> furq: i have started testing with 2 inputs at a time, command: https://pastebin.com/q4jBUN51 and its working fine [14:23:03 CEST] <pk08> i will add more and more, and bother you guys if i will have any issue ;) [14:36:49 CEST] <keglevich> pk08: I checked your pastebin... what do you actually plan to achieve? multiscreen UDP-ts maybe? [14:38:42 CEST] <pk08> keglevich: i want to scale input video streams to 480x270 and make single multicast of 16 video streams [14:39:37 CEST] <keglevich> that's something I was trying to achieve a while ago... and I gave up in the end [14:39:51 CEST] <keglevich> one multicast stream drops... everything drops [14:40:18 CEST] <JEEB> yea you need to do your own fallback handling in your own ffmpeg api code [14:40:33 CEST] <JEEB> ffmpeg.c won't help you as it's a generic tool [14:40:51 CEST] <JEEB> you can of course add it into ffmpeg.c if you want [14:41:07 CEST] <pk08> yes, but i guess shortest filter will help in that case keglevich [14:41:31 CEST] <JEEB> well that will kill the whole thing :p [14:41:41 CEST] <keglevich> I was also looking for already made alternative, and I was out of luck as well...althought I've seen some pretty nice demo examples on youtube, but never got any real solution to this [14:41:41 CEST] <JEEB> not dynamiclaly flal back to backuo [14:41:49 CEST] <keglevich> so I'm still interested if there's "working" one [14:42:10 CEST] <JEEB> upipe has dynamic source/backup source switching but also requires your own code [14:42:58 CEST] <pk08> keglevich: i will let you know if i make it working... [14:43:27 CEST] <furq> pk08: there's not really any way around that problem [14:43:36 CEST] <furq> i figured all your sources are from the same ip so it should probably work [14:43:37 CEST] <pk08> JEEB: i was trying to make this by coding but i gave up because i wasnt getting output at all... [14:44:00 CEST] <furq> barring issues on the source side [14:44:35 CEST] <keglevich> it's called "mosaic" now I remember... there was some talks about VLC being able to do this...also ffmpeg... but as said, I haven't found anything in the end... [14:44:41 CEST] <keglevich> that's how should it look like: http://mib.pianetalinux.org/blog/images/stories/t/tvmosaic020-1.jpg [14:44:41 CEST] <pk08> furq: its just an example, if this will work i will have different ip, different url format like hls, rtmp or even ts-file [14:44:53 CEST] <furq> yeah that probably won't work well then [14:45:05 CEST] <furq> there's no way of handling reconnection etc [14:46:11 CEST] <keglevich> ^ this [14:46:55 CEST] <keglevich> but as said...there are already made solutions for this kind of mosaic screens...I just haven't found one for free [14:47:27 CEST] <pk08> keglevich: image looks cool, i need something like this [14:47:43 CEST] <pk08> but with some more features [14:47:48 CEST] <keglevich> so I'm just using multiple VLC instances ... each instance one UDP stream [14:48:07 CEST] <keglevich> yeah I know exactly what you need...but I really think you're wasting time with ffmpeg to do this [14:49:08 CEST] <keglevich> more or less all (payable) playout solutions have something like this...but again, it's not free [14:49:24 CEST] <furq> you can do most of this with ffmpeg, just not the network input handling [14:49:33 CEST] <furq> and you could probably do that well enough with the ffmpeg libs [14:50:12 CEST] <keglevich> maybe try to do something with this: https://wiki.videolan.org/VLC_HowTo/Make_a_mosaic/ [14:51:01 CEST] <keglevich> I pretty much believe he needs a solution "out-of-the-box" as soon as possible and can't spend time or doesn't have knowledge to code his own with ffmpeg...or maybe I'm wrong? [14:51:29 CEST] <furq> i suspect most people don't want to write C in general [14:52:07 CEST] <keglevich> with me lack of programming knowledge was always an issue [14:52:32 CEST] <pk08> well, i spent atleast a month in c coding in this issue, but somehow i couldnt get output [14:52:38 CEST] <pk08> so i moved to ffmpeg command line [14:53:41 CEST] <pk08> thank you guys for your feedback, i need to rethink about this... [14:54:21 CEST] <keglevich> try with VLC as I said...it works pretty well...but again, you'll need to restart (stop/play) some stuck UDP streams sometimes as we do [14:55:09 CEST] <keglevich> and you'll also have EPG, teletext, interlacing, etc available out-of-the-box [14:55:17 CEST] <keglevich> to control each UDP stream separately [14:55:30 CEST] <pk08> yeah, i got yo [14:56:00 CEST] <pk08> because of packet loss or audio/video issue or other issue [14:56:07 CEST] <pk08> we might need restart [14:56:23 CEST] <keglevich> yeah, sometimes it happens, without a proper reason [14:57:02 CEST] <pk08> i can understand i have been through this [14:57:59 CEST] <keglevich> I also wasted a lot of time trying to achieve the same as you try now, without any real success in the end...so we just do everything manually now, using multiple VLC instances...it works ok more or less [14:58:04 CEST] <keglevich> and it's free [14:58:36 CEST] <keglevich> if you find a solution, please drop me an email to suljotron(a)gmail.com [14:58:47 CEST] <keglevich> I'd be glad to reply if I can help with something [15:02:39 CEST] <pk08> sure, keglevich [15:03:03 CEST] <pk08> i will mail you, if i find any working solution [15:03:19 CEST] <pk08> and my email is parbatvgec(a)gmail.com [15:05:53 CEST] <EvanCarroll> minor documentation bug: https://unix.stackexchange.com/q/523188/3285 [15:10:19 CEST] <furq> the latest openssl release is still under the old license [15:10:39 CEST] <furq> so i'd say the docs are still correct for now [15:41:28 CEST] <Anill> if i provide different segements of file to ffmpeg, will it write(after transcoding) to same file, or over write the segments or create different files? [16:18:06 CEST] <th3_v0ice> keglevich, I am sorry but -stitichable is the only thing that comes to mind. On a side note I think that the video is perfectly fine but its only reported as variable framerate because last packet's from concatenating may not have proper packet durations set. Mediainfo was giving me variable framerate whenever my packet durations were off even by a little. [16:18:56 CEST] <th3_v0ice> You can use -debug_ts to actually test if the packets are spaced equally. [22:34:19 CEST] <electrotoscope> Does anyone have an example of -vf "drawtext=text=%{metadata:_____}" actually working? I'm absolutely stumped. I'm pretty sure I've escaped the : properly, because I can get it to switch to the "use if metadata not found" second argument, but I can't get it to display anything at all from the first argument [22:35:31 CEST] <electrotoscope> https://stackoverflow.com/questions/24744608/ffmpeg-and-timecode-from-movie… says that it has to be something that shows up in an ffprobe frame under "TAG:____" but I can't find an example file that has that [23:48:05 CEST] <another> hmm.. trac down? [23:52:22 CEST] <steve___> another: something has been funky with the webserver. Just keep trying and the page should load [00:00:00 CEST] --- Fri Jun 7 2019

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ffmpeg-devel.log.20190605
by burek 06 Jun '19

06 Jun '19

[00:06:29 CEST] <BtbN> Ideally I'd like to get rid of cuviddec, but there isn't really a viable replacement for some of its extra features [00:27:00 CEST] <darkapex> BtbN: yeah the code seems redundant at places, but hardware scaling/deint is quite useful ig [00:42:54 CEST] <philipl> BtbN: that wiki page could do with a better description of how nvdec and cuvid differ. I keep thinking of updating it but then I lose the motivation. [11:19:52 CEST] <jdarnley> Is it just my client or has fflogger reported "reopenedticket" for #6813 a few times since last night? [11:20:48 CEST] <nevcairiel> yeah the bot seems to be stuck in a loop again [13:21:19 CEST] <cone-813> ffmpeg 03Michael Niedermayer 07master:0a2b768d3ecc: tools/target_dec_fuzzer: Limit error concealment on pixels instead of just frames [13:21:20 CEST] <cone-813> ffmpeg 03Michael Niedermayer 07master:a99ffb5bb445: avcodec/diracdec: Fix integer overflow in global_mv() [13:21:21 CEST] <cone-813> ffmpeg 03Michael Niedermayer 07master:8e520843dd76: avcodec/dxv: Check op_offset in dxv_decompress_cocg() [13:21:22 CEST] <cone-813> ffmpeg 03Michael Niedermayer 07master:0fef412dffb7: avcodec/mss4: Check input size against skip bits [00:00:23 CEST] --- Thu Jun 6 2019

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ffmpeg.log.20190605
by burek 06 Jun '19

06 Jun '19

[00:05:23 CEST] <electrotoscope> Is there a way to use drawtext=textfile to write one line of the txt file for every frame of the video? Like if my text file was "hello[linereturn]world" it would write "hello" to the first frame and "world" to the second frame [00:10:52 CEST] <Zcom> try to add a subtitle file [00:12:07 CEST] <BtbN> one line of text per frame is either a very low fps video or very fast text [00:16:41 CEST] <electrotoscope> it's from a text document I have with the date/time that the frame was recorded, but it's not written into the source file. It is indeed fast text!! [00:17:06 CEST] <electrotoscope> I was thinking about a .srt but it's weird VFR source [00:17:48 CEST] <electrotoscope> I think I've found a way where I put in a ludicrous amount of line returns and set y=(offsetpx-(n*linereturnspx)) [00:49:27 CEST] <ctcx> Trying to watch DVR rtsp stream with ffplay. I use "ffplay rtsp://user:pass@full_url", [00:49:33 CEST] <ctcx> and get "method DESCRIBE failed: 404 stream not found" [00:50:01 CEST] <ctcx> So the server (the DVR) is indeed found, but stream is not. Any ideas? [02:52:46 CEST] <friendofafriend> Have any of you nice folks made a normal video into a "side-by-side" video for Google Cardboard VR (et al)? [02:58:28 CEST] <mozzarella> why? [03:00:44 CEST] <friendofafriend> mozzarella: The geometry seems kind of confusing. I'm guessing you'd have to crop the sides of a 16:9 to make two side-by-side 8:9 images, or scale it. [03:01:10 CEST] <friendofafriend> (And I didn't see any examples around that I could use as a starting point.) [03:10:45 CEST] <friendofafriend> Ah, this seems to be doing sort of what I'm after. [03:10:51 CEST] <friendofafriend> ffmpeg -i "left.mkv" -i "right.mkv" -filter_complex "pad=in_w*2:in_h, overlay=main_w/2:0, scale=in_w/2:in_h, scale=-1:1080" -b:v 10000k -vcodec libx264 -an "output.mkv" [03:24:30 CEST] <friendofafriend> And this works much better: ffmpeg -i "input.mkv" -i "input.mkv" -filter_complex "pad=in_w*2:in_h, overlay=main_w/2:0, scale=in_w/2:in_h, scale=-1:in_h*2" -b:v 10000k -vcodec libx264 -c:a copy "output.mkv" [03:24:34 CEST] <friendofafriend> Thanks for the help, all. [07:31:06 CEST] <cards> Could someone walk me through the process of demuxing and remuxing audio from one video onto another? [07:31:11 CEST] <cards> example: this video https://www.facebook.com/watch/?v=448098122617310 to this audio https://www.youtube.com/watch?v=DnsDbOi6sTY [09:33:01 CEST] <dongs> [matroska,webm @ 000001cf18d5ad40] Element at 0x1aaaa90d ending at 0x1aaaa96f exceeds containing master element ending at 0x1aaaa908 [09:33:04 CEST] <dongs> ??? [09:33:42 CEST] <dongs> this is on like couple days old ffmpeg build [09:35:09 CEST] <pink_mist> sounds like someone made a shitty webm [09:36:35 CEST] <dongs> its a mkv that came from eac3to from a disk that I own [09:36:40 CEST] <dongs> not even webm [10:13:35 CEST] <dongs> [aac_latm @ 0000027aa6c39900] Multiple programs is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented. [10:36:41 CEST] <ntt> Hello, I'm trying to stream a webcam video to an nginx rtmp server. It works, but I have a huge delay (about 30 sec). How can I solve the issue? Actually I'm using this: ffmpeg -i /dev/video0 -framerate 1 -video_size 720x404 -vcodec libx264 -maxrate 768k -bufsize 8080k -vf "format=yuv420p" -g 60 -f flv rtmp://X.X.X.X:1936/live/camerapc [10:53:37 CEST] <dongs> try using windows [10:54:40 CEST] <ntt> dongs: I don't think this is the solution [10:55:25 CEST] <dongs> i mean [10:55:26 CEST] <dongs> at 1fps [10:55:29 CEST] <dongs> and bufsize of 8megs [10:55:33 CEST] <dongs> and 720x400 [10:55:41 CEST] <dongs> why are you complaining about 30sec delay? [10:55:45 CEST] <dongs> that sounds quite reasonable [10:56:02 CEST] <dongs> 30 sec of 1fps trash in h264 probably is around 8megs. [11:08:44 CEST] <ntt> I can modify all values. The goal is to obtain a reduced delay, even though I will decrease quality [11:09:10 CEST] <ntt> can you give me some idea? The problem actually is delay [11:09:17 CEST] <furq> ntt: how are you watching this video [11:09:25 CEST] <dongs> i just explained you the reason for delay [11:10:00 CEST] <ntt> furq: I'm using vlc [11:10:28 CEST] <ntt> dongs: so, can I use a different framerate and reduce the buffer size? [11:12:20 CEST] <ntt> I'm using vlc with rtmp://X.X.X.X:1936/live/camerapc .... and I'm using nginx-rtmp [11:12:46 CEST] <ntt> dongs: I'm trying with ffmpeg -i /dev/video0 -framerate 15 -video_size 720x404 -vcodec libx264 -preset veryfast -maxrate 768k -bufsize 100k -vf "format=yuv420p" -g 30 -f flv [11:12:51 CEST] <ntt> but nothing changes [11:16:31 CEST] <furq> ntt: -framerate and -video_size are input options, they're not doing anything in that command [11:16:36 CEST] <furq> move them before -i [11:19:11 CEST] <ntt> I have about 8/9 seconds of delay. Is this normal? Can I obtain something better? [11:20:45 CEST] <ntt> Actually I'm using this: ffmpeg -framerate 25 -video_size 640x480 -i /dev/video0 -vcodec libx264 -preset veryfast -maxrate 768k -bufsize 100k -vf "format=yuv420p" -g 30 -f flv [11:22:51 CEST] <ntt> Could I use something different than h264? [12:28:10 CEST] <Anill> Hi [12:28:26 CEST] <rnderr> Hu [12:28:44 CEST] <rnderr> \O/ [12:28:46 CEST] <Anill> is there any way to identify how much time ffmpeg will take to transcode files in a folder without actually transcoding the same. [12:29:01 CEST] <Yuyu0> no [12:29:43 CEST] <Anill> I just started with ffmpeg can you explain what this command will do ffmpeg -i input.avi -vf scale=320:240 output.mp4 [12:29:52 CEST] <Anill> whats -vf option is [12:29:59 CEST] <Anill> and scale=320:240 mean [12:38:53 CEST] <DHE> convert the AVI to MP4 with transcoding, and rescaling the input image to 320x240 [12:41:03 CEST] <Anill> can it be possible to pass the directory to ffmpeg and it transcodes all the files in directory in the directory? [12:43:19 CEST] <DHE> no, have your shell do that [12:44:11 CEST] <Anill> DHE: One more question, can we transcode the video files playlist from ffmpeg [13:17:10 CEST] <Anill> Can we transcode the video files playlist from ffmpeg. [13:17:32 CEST] <TheDude9000> hi, does anyone know if there's a straightforward way to get the desired speaker location for a given audio stream channel? [13:18:01 CEST] <TheDude9000> also does FFMPEG automatically reorder the channel layout? [13:51:28 CEST] <Anill> Has anyone used this https://pypi.org/project/ffmpeg-progress/ on windows? [13:52:07 CEST] <grkblood13> not a ffmpeg specific question but I wasn't sure where else to ask this. When using the HTML5 fetch API to download a MPEGTS stream, is there a way to seperate the download into proper MPEGTS packets? [13:53:11 CEST] <grkblood13> fill a buffer up until it has a completed packet, hand it off to the next process, empty buffer, rinse and repeate [14:14:46 CEST] <grkblood13> repeat* [14:22:38 CEST] <deterenkelt> ffmpeg -v error -dump_attachment:t "" -i input.mkv prints a confusing error At least one output file must be specified. The attachments are all extracted. How to avoid this message? [14:24:02 CEST] <deterenkelt> If you specify -f null - at the end, ffmpeg tries to transcode the video instead of quickly extracting fonts. [14:25:45 CEST] <deterenkelt> FFmpeg is compiled from git, heres -version output: https://pastebin.com/raw/4wpYfsuN [14:47:47 CEST] <th3_v0ice> Why in the examples of the API usage we are shown that packet's DTS and PTS information should be converted from the stream's time base to the codec time base? I cant find a place where FFmpeg.c is doing this particular step. Its using stream time base only. [14:48:31 CEST] <th3_v0ice> I am considering a case where decoding is needed. [16:42:49 CEST] <atbd> hi, i'm trying to use complex filters in C to rescale (1024x576 -> 720x576), change format (yuv420p -> gray8) and deinterlace (yadif) but i have issues with the conversion. Once i add it the output frames are empty. Do I miss something ? [16:43:00 CEST] <atbd> Using ffmpeg api 4.1.3 [16:44:17 CEST] <atbd> My filter description: yadif,scale=720x576,format=pix_fmts=gray8 [17:04:47 CEST] <steve___> I'm trying to concat three videos without transcoding them. The second video is two seconds of black which I generated from two black png's. Here is the stream output from ffprobe for all three videos: [17:04:49 CEST] <steve___> Stream #0:0(eng): Video: h264 (Baseline) (avc1 / 0x31637661), yuvj420p(pc, bt470bg/bt470bg/smpte170m), 1920x1080, 16998 kb/s, SAR 1:1 DAR 16:9, 29.93 fps, 29.92 tbr, 90k tbn, 180k tbc (default) [17:04:51 CEST] <steve___> Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc), 1920x1080, 21 kb/s, 29.93 fps, 29.93 tbr, 11972 tbn, 59.86 tbc (default) [17:04:53 CEST] <steve___> Stream #0:0(eng): Video: h264 (Baseline) (avc1 / 0x31637661), yuvj420p(pc, bt470bg/bt470bg/smpte170m), 1920x1080, 16998 kb/s, SAR 1:1 DAR 16:9, 29.93 fps, 29.92 tbr, 90k tbn, 180k tbc (default) [17:06:15 CEST] <DHE> I would suggest redoing that blank video with -profile:v baseline then [17:08:30 CEST] <steve___> I didn't work. This is the command I used to generate the blank video -- ffmpeg -y -loglevel -8 -framerate 1 -i img/black-%02d.png -c:v libx264 -r 29.93 -pix_fmt yuvj420p -profile:v baseline 03.mp4 [17:08:46 CEST] <steve___> the stream output -- Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuvj420p(pc), 1920x1080, 29 kb/s, 29.93 fps, 29.93 tbr, 11972 tbn, 59.86 tbc (default) [21:03:33 CEST] <CarlFK> can someone show me a current "save mixed to disk" command? [21:04:07 CEST] <CarlFK> I just discovered the one I use has -ilme "Force interlacing support..." [21:04:20 CEST] <furq> mixed what [21:09:38 CEST] <CarlFK> um, mixed streams? [21:09:51 CEST] <CarlFK> https://github.com/CarlFK/voctomix-outcasts/blob/master/record-timestamp.sh… [21:11:16 CEST] <CarlFK> opps, wrong chan - I thought this was #voctomix [21:11:45 CEST] <CarlFK> context: https://github.com/voc/voctomix/blob/master/example-scripts/ffmpeg/record-m… [21:25:08 CEST] <another> CarlFK: still not sure what you want to do [21:25:29 CEST] <CarlFK> another: not force interlacing ;) [21:25:48 CEST] <another> is your content interlaced? [21:27:32 CEST] <CarlFK> no - but don't worry about it, I mostly got what I was looking for: [21:27:41 CEST] <CarlFK> https://github.com/voc/cm/blob/master/ansible/roles/encoder/templates/vocto… [21:28:59 CEST] <another> right. the currently maintained scripts are in ansible [21:29:35 CEST] <CarlFK> another: you know/do vocto? [21:30:02 CEST] <another> i looked at some of the scripts [21:30:52 CEST] <another> and wrote a mini patch once [21:33:14 CEST] <CarlFK> neat [00:00:23 CEST] --- Thu Jun 6 2019

1 0

ffmpeg-devel.log.20190604
by burek 05 Jun '19

05 Jun '19

[00:22:33 CEST] <cone-981> ffmpeg 03James Almer 07master:ff2a638c8316: avcodec/cbs_h264: add support for Alternative Transfer Characteristics SEI message [09:29:19 CEST] <cone-412> ffmpeg 03Paul B Mahol 07master:d3f236b300e3: avilter/avf_showwaves: switch to activate in showwaves filter [11:02:39 CEST] <j-b> 'lo [11:05:33 CEST] <durandal_1707> hi [11:05:41 CEST] <j-b> 'tsup, man? [13:20:32 CEST] <cone-412> ffmpeg 03Michael Niedermayer 07master:3d5863d73915: avcodec/aacdec_fixed: ssign seems always -1 in noise_scale(), simplify [13:20:33 CEST] <cone-412> ffmpeg 03Michael Niedermayer 07master:bc33c99d5679: avcodec/aacdec_template: Merge 3 #ifs related to noise handling [13:20:34 CEST] <cone-412> ffmpeg 03Michael Niedermayer 07master:3d14663f8345: avcodec/aacdec_fixed: Handle more extreem cases in noise_scale() [13:20:35 CEST] <cone-412> ffmpeg 03Michael Niedermayer 07master:cdd886a286fb: avcodec/arbc: Skip tiles in fill_tileX() which are completely outside [13:20:36 CEST] <cone-412> ffmpeg 03Michael Niedermayer 07master:cf3156e762bb: avcodec/aacdec_template: skip apply_tns() if max_sfb is 0 (from previous header decode failure) [13:20:37 CEST] <cone-412> ffmpeg 03Michael Niedermayer 07master:89836ad5a7a8: avcodec/pafvideo: Clear frame buffer later [13:20:38 CEST] <cone-412> ffmpeg 03Michael Niedermayer 07master:279d9a84af37: avcodec/vmnc: Check available space against chunks before reget_buffer() [14:00:39 CEST] <JEEB> )28 [16:22:38 CEST] <kepstin> huh, regarding ticket #7939, I have a half finished patch to improve the framerate/timebase selection in the concat filter, I should look at that again :/ [16:35:17 CEST] <cone-412> ffmpeg 03Andreas Rheinhardt 07master:d81913e68017: bitstream_filters: Correct dump_extradata description [16:43:40 CEST] <Anonffmpeg> Hi fella [16:43:42 CEST] <Anonffmpeg> Hi. Have a particular question: Have a program that generates continously a file, named "file.jpg" every millisecond for 10 seconds. Need that ffmpeg inputs(-i) the file continously and provide a "video.mp4" that is being "streamed"(keeps growing). [16:45:29 CEST] <gnafu> Anonffmpeg: You might be better off asking in #ffmpeg, as this channel is specifically for ffmpeg development. [16:45:34 CEST] <Anonffmpeg> Wrong channel [16:45:52 CEST] <Anonffmpeg> Thanks! [23:25:13 CEST] <darkapex> what's the difference between cuvid and nvdec here https://trac.ffmpeg.org/wiki/HWAccelIntro#NVDECCUVID ? I see that nvdec is a hwaccel module, while cuvid is a full decoder. but what does that mean internally in terms of host-device memory copying? [23:26:11 CEST] <darkapex> also from what i could see, lavc/cuviddec.c does not seem to use the functions in lavc/nvdec* is that correct? [23:26:30 CEST] <BtbN> One uses ffmpeg parser, one nvidia one [23:27:41 CEST] <BtbN> cuvid can also make use of some special features of nvdec like scale/crop/deint, since it's a standalone decoder and can change frame size [23:31:52 CEST] <darkapex> BtbN: oh I see thanks! so CUVID passes bitstream as-is to the NVIDIA driver i believe? but NVDEC being a HWACCEL module uses ffmpeg's parsing and just passes individual frames to the driver right? [23:36:49 CEST] <darkapex> ok yeah this clarifies it https://github.com/FFmpeg/FFmpeg/commit/ae5046e492 thanks! [00:00:00 CEST] --- Wed Jun 5 2019

1 0

ffmpeg.log.20190604
by burek 05 Jun '19

05 Jun '19

[00:00:02 CEST] <Necrosand> pink_mist yes in one file [00:00:12 CEST] <pink_mist> then you want to merge them using a filter [00:00:25 CEST] <pink_mist> find the right filter to use, and figure out how to use it [00:02:37 CEST] <Necrosand> cehoyos -map 0 didn't work [00:02:44 CEST] <Necrosand> ffmpeg -map 0 -f dshow -i audio="desktop audio" -f dshow -i audio="Microphone" output.mka [00:02:48 CEST] <pink_mist> Necrosand: http://ffmpeg.org/ffmpeg-filters.html#amerge-1 [00:02:49 CEST] <cehoyos> That's because I did not suggest it;-) [00:02:51 CEST] <cehoyos> (As said) [00:03:30 CEST] <pink_mist> or possibly amix [00:04:01 CEST] <pink_mist> Necrosand: http://ffmpeg.org/ffmpeg-filters.html#amix [00:05:16 CEST] <Necrosand> cehoyos you said -map 0 is missing [00:06:09 CEST] <Necrosand> pink_mist okay problem is it did not create mka with 2 audio tracks [00:06:41 CEST] <pink_mist> well you probably needed what cehoyos said, but that hardly matters now anyway, since that's not what you wanted [00:06:58 CEST] <cehoyos> No [00:07:12 CEST] <Necrosand> pink_mist true but i know how to merge 2 audio track using audacitry [00:08:17 CEST] <pink_mist> then why did you come here and scream about stuff not working? [00:08:28 CEST] <Necrosand> pink_mist what happens when you run the same command? [00:08:37 CEST] <pink_mist> I don't even have ffmpeg installed here [00:08:44 CEST] <pink_mist> so it tells me that ffmpeg doesn't exist [00:08:56 CEST] <Necrosand> lol [00:09:11 CEST] <Necrosand> you make it sound like ffmpeg is so hard to get [00:10:17 CEST] <pink_mist> it takes a while to compile, especially with all the prerequisites I'd want to make it usefuler [00:15:18 CEST] <Necrosand> cehoyos where do i put -map 0 [00:19:45 CEST] <cehoyos> -map 0 -map 1 missing [00:19:58 CEST] <cehoyos> But only if you want two audio streams in your output file. [00:20:06 CEST] <cehoyos> If you want one stream, use one of the filters mentioned [00:20:50 CEST] <Necrosand> cehoyos i want to see if two audio streams even works [00:21:08 CEST] <Necrosand> even that working is a progress [00:23:55 CEST] <Necrosand> ffmpeg -map 0 -f dshow -i audio="what u hear (Creative SB X-Fi)" -map 1 -f dshow -i audio="Microphone (Creative SB X-Fi)" test.mka [00:24:04 CEST] <Necrosand> that didn't work [00:31:03 CEST] <cehoyos> Doesn' [00:31:09 CEST] <cehoyos> t the console output tell you why? [00:31:23 CEST] <cehoyos> (I ask because I just tested and I get a very descriptive error message) [00:32:24 CEST] <cehoyos> Anyway: ffmpeg -f dshow -i audio="what u hear (Creative SB X-Fi)" -f dshow -i audio="Microphone (Creative SB X-Fi)" -map 0 -map 1 test.mka [00:34:49 CEST] <Necrosand> oh okay, i did it wrong [00:34:50 CEST] <Necrosand> thanks [00:48:58 CEST] <Necrosand> -filter_complex "[0:a][1:a]amerge=inputs=2[a]" -map "[a]" output.mka : why is this creating 4 channel output.mka file [02:35:28 CEST] <kepstin> Necrosand: because that's what the amerge filter does, according to its documentation. you probably wanted amix instead. [02:35:37 CEST] <Necrosand> oh ok [02:35:45 CEST] <Necrosand> amerge is always 4 chnanels? [02:36:47 CEST] <kepstin> no, amerge combines an N channel input and an M channel input into an N+M channel output [02:37:15 CEST] <Necrosand> oh ok [02:37:21 CEST] <Necrosand> thanks [11:52:25 CEST] <Atlenohen> Hey [11:52:37 CEST] <Atlenohen> libx264rgb is not mentioned on the configure [11:53:03 CEST] <Atlenohen> there's no error or anything, it's not reported as included, if I do it specifically or libx264*, same [11:53:19 CEST] <Atlenohen> When trying to build FFmpeg [11:53:36 CEST] <Atlenohen> I'll see later if it's really included or not [11:54:36 CEST] <BtbN> Isn't libx264rgb just a differently named endpoint of just libx264, which feeds RGB to x264, which thinks it's YUV? [11:55:29 CEST] <Atlenohen> That may be the case yes, with some of these phantom encoders/muxers, but I'm no FFmpeg expert. [11:57:26 CEST] <Atlenohen> btw, would lossless mode x264 be of any benefit over FFV1 ? [11:57:50 CEST] <BtbN> You'll need to test what's smaller/faster for your case [11:58:08 CEST] <BtbN> And if speed or efficiency is more important [11:58:24 CEST] <Atlenohen> oh, ok [12:03:35 CEST] <Atlenohen> Slight bug, after building, avconfig.h and ffversion.h don't come out as CRLF [12:12:06 CEST] <dongs> [matroska @ 000001b98c918800] Can't write packet with unknown timestamp [12:12:08 CEST] <dongs> how do i fix this [12:14:36 CEST] <dongs> aight muxing to mp4 works [12:20:26 CEST] <BtbN> Why would anything come out with Windows line endings? [12:20:43 CEST] <BtbN> What even still needs them? Even Windows is slowly abandoning it. [12:20:44 CEST] <dongs> wut [12:21:21 CEST] <dongs> are you one of thos dudes who thinks lunix is succeeding on desktop? [12:21:50 CEST] <BtbN> You are aware they made notepad.exe aware of non-Windows-Lineendings recently? [12:22:02 CEST] <BtbN> They are moving away from CRLF. Very slowly though. [12:22:04 CEST] <zamba> hi! i have an audio recording that has been made with the samsung voice recorder.. when pulling the file from the mobile phone, its extension is .m4a and i'm not able to play it back using mplayer/ffplay.. 'file' says: "ISO Media, MPEG v4 system, 3GPP" [12:22:05 CEST] <dongs> no because i literally have never encountered such a file [12:22:10 CEST] <dongs> in all the decades of usign windows [12:22:28 CEST] <zamba> ffplay barfs with:moov atom not found [12:22:29 CEST] <dongs> zamba, could it be DRM/protected? [12:22:34 CEST] <dongs> oh, its short then [12:22:39 CEST] <zamba> Invalid data found when processing input [12:22:41 CEST] <dongs> did recording abort due to low battery or such? [12:23:14 CEST] <BtbN> Best case copying it from the device was just interrupted and it's missing part of the file. [12:23:24 CEST] <BtbN> Worst case, that recording is busted. [12:23:51 CEST] <zamba> BtbN: what about just parsing/reading what's ok? if the file in fact was cut short? [12:24:12 CEST] <BtbN> Can't, without the moov atom, an mp4 file is unreadable. And it's at the end of the file. [12:25:56 CEST] <zamba> so it's not possible to make sense of the raw binary stream without the moov atom? [12:26:20 CEST] <zamba> i mean.. not even "forensicly"? [12:26:59 CEST] <dongs> you can make another recording [12:27:00 CEST] <dongs> with same settings [12:27:03 CEST] <dongs> and tehre's software/scripts [12:27:16 CEST] <dongs> that can transplant the required mp4 bits from known good fine to bad one [12:27:26 CEST] <dongs> not sure how well that works all the ones I've seen are geared for video [12:27:31 CEST] <dongs> maybe audio works same way? [12:28:43 CEST] <zamba> but i'm quite sure this is not mp4 [12:28:51 CEST] <zamba> i believe it should be 3ga [12:39:39 CEST] <dongs> why dont you just make another recording and see [12:39:42 CEST] <dongs> how did you get the file [12:42:51 CEST] <zamba> dongs: i'm trying to save a file for someone else [12:43:07 CEST] <dongs> ic then not much you can do [12:43:33 CEST] <dongs> -c:v h264_nvenc -preset slow -profile:v high -level 5.2 -b:v 20M why doesnt this work? [12:43:46 CEST] <dongs> oh hm wat no nvenc capable devices found [12:43:52 CEST] <dongs> wtf is going on [13:03:31 CEST] <BtbN> Increase the log level, it should tell you more about what exactly it's unhappy about [13:05:16 CEST] <Atlenohen> I'm starting to dwelve into the FFmpeg API in depth, rewriting the old API implementation [13:05:39 CEST] <Atlenohen> Wondering what does this do right now? https://pastebin.com/biNbErNz [13:06:31 CEST] <Atlenohen> I'm getting rid of support anything below the version I included, I believe it's 4.1.3 [13:07:12 CEST] <Atlenohen> so this whole segment can be deleted? [13:07:14 CEST] <BtbN> Stuff was renamed, and this just renames it via macros [13:07:20 CEST] <BtbN> If the version is too old the have it [13:08:54 CEST] <Atlenohen> oh I guess it doesn't hurt, but I think the stuff I'll be doing later will break old version anyway [13:10:51 CEST] <Atlenohen> Doxygen is the optimal source to look for the version specific API stuff right? [13:10:55 CEST] <BtbN> Might as well just get rid of it if you hard-require something newer [13:11:11 CEST] <Atlenohen> Yeah I was thinking like that too [13:11:16 CEST] <BtbN> I don't think there's anything to look at that [13:11:42 CEST] <BtbN> stuff like that usually comes to by because an old codebase broke when built against a newer ffmpeg version, and then gets patched up. [13:11:43 CEST] <Atlenohen> Oh, I mean in general, when beginning to work on the inital implementation [13:12:02 CEST] <BtbN> I'd just look at the headers [13:12:23 CEST] <Atlenohen> Like what to do first, second, next ... first I need a av_output_context, etc like that, if there's some flow guide, not just alphabetic order [13:12:38 CEST] <Atlenohen> Because it's the first time I'm doing it [13:13:40 CEST] <Atlenohen> This is only encode tho, well first is libavcodec, I'm sure that's not the place to start looking at right [14:00:30 CEST] <Atlenohen> BtbN, are example files of any use? [14:01:11 CEST] <Atlenohen> like encode_video.c ... however this is a C++ program integrating FFmpeg. [14:03:55 CEST] <Atlenohen> Also, seems like version specific shows is exactly that version, so it doesn't include everything, is that trunk? [14:04:12 CEST] <Atlenohen> For example wasn't av_network_init deprecated but it's not showing up in 4.1? [14:04:32 CEST] <Atlenohen> sry: avformat_network_init [14:04:50 CEST] <DHE> it's deprecated. you don't need to do it anymore, but calling it won't break anything [14:06:23 CEST] <Atlenohen> I go to doxygen and search but finds nothing in 3.2 or 2.0 eithe [14:06:40 CEST] <Atlenohen> Maybe it's mentioned only in the version it was deprecated in [14:07:01 CEST] <DHE> oxygen should still mention the function, if only to say it's deprecated and what to do instead now [14:07:04 CEST] <Atlenohen> I think av_register_all(); was too right? [14:08:05 CEST] <DHE> https://www.ffmpeg.org/doxygen/trunk/group__lavf__core.html#ga84542023693d6… # avformat_network_init is deprecated and not needed anymore (on Trunk) [14:09:59 CEST] <Atlenohen> oh ok [15:11:37 CEST] <Gunstick> I already asked yesterday and get reply "channelsplit". Yeah, duh, I don't understand complex filter. So how do I remove the first 2 channels and graph the other 3? Need I 2 ffmpeg commands piped? ffmpeg -y -t 5 -i You_Bore.wav -filter_complex "[0:a]channelsplit=channel_layout=5.0[BL][BR][FL][FR][FC][BL][BR];[0:a]showwaves=s=1280x720:split_channels=y:mode=line,format=yuv420p[v]" -map "[v]" -map 0:a -c:v libx264 You_Bore.mkv # source file is here: [15:11:37 CEST] <Gunstick> http://gkess.homeip.net/~georges/You_Bore.wav [15:12:37 CEST] <Gunstick> the above thing does obviously not work as is. [15:14:34 CEST] <DHE> well you're using 5.0 but have 6 outputs to channelsplit [15:14:46 CEST] <DHE> *7 [15:15:12 CEST] <Gunstick> oops, but removing the duplicate [BL][BR] does not change a thing. [15:16:07 CEST] <Gunstick> because I do not understand complex filter. So "BR has an unconnected output" does not mean anything to me. sorry [15:16:47 CEST] <durandal_1707> learn how to learn how to learn to learn to read documentation [15:17:21 CEST] <DHE> I think channelmap to re-assemble the split waves into a single 3.0 stream for showwaves would do you [15:17:42 CEST] <durandal_1707> no [15:17:49 CEST] <Gunstick> indeed. The documentation just says how to split into different streams, but I want to have one 3 cheannel stream and then put that through showwaves. [15:18:00 CEST] <DHE> there a better way? [15:18:38 CEST] <durandal_1707> plus join/amerge and anullsink [15:19:32 CEST] <DHE> okay amerge makes more sense [15:19:51 CEST] <dongs> yall 5.1 audio plebs, i've got some 22.2 ch audio i can't even play it back [15:26:46 CEST] <Gunstick> @DHE @durandal_1707 anullsink + amerge! ffmpeg -y -t 5 -i You_Bore.wav -filter_complex "[0:a]channelsplit=channel_layout=5.0[BL][BR][FL][FR][FC];[BL]anullsink;[BR]anullsink;[FL][FR][FC]amerge=inputs=3[YM];[YM]showwaves=s=1280x720:split_channels=y:mode=line,format=yuv420p[v]" -map "[v]" -map 0:a -c:v libx264 You_Bore.mkv [15:27:10 CEST] <durandal_1707> dongs: you havent provvided it [15:27:38 CEST] <dongs> durandal_1707: eh i could, what would it do tho [15:27:45 CEST] <Gunstick> DHE: thanks for the help [15:27:47 CEST] <DHE> Gunstick: seems okay. try it [15:27:53 CEST] <dongs> its LATM AAC btw [15:28:19 CEST] <dongs> and im not quite sure of the mapping either. it could be 3 separate streams since aac spec doens't have that many channels configuration [15:28:21 CEST] <Gunstick> DHE: yes, I confirm it does what I want. Now back to rotating the result 90° and I'm done. hoooray. [16:03:50 CEST] <dongs> uh... where exactly do i find out if F302VC has TIM3 or not [16:03:54 CEST] <dongs> im seeing conflicting shit [16:04:01 CEST] <dongs> and i powered it up and all registers are at 0... [16:05:14 CEST] <dongs> hmm my reference manual is for 303 [16:05:17 CEST] <dongs> i bet 302 is more gimped [16:14:54 CEST] <dongs> hm 302xB/C is supposed to have TIm3 [16:14:59 CEST] <dongs> err what [16:15:01 CEST] <dongs> wrong channel [16:15:21 CEST] <durandal_1707> dongs: nope, please continue, its very interesting :) [16:28:53 CEST] <roxlu> hey, when I play a video with `ffplay` can I see what video decoder is used? [16:31:14 CEST] <roxlu> this is what i see https://gist.github.com/roxlu/e9b37b04bd749f45774d7e583bb4f10c [16:36:02 CEST] <tinystoat> roxlu: ffprobe [16:37:02 CEST] <roxlu> tinystoat: what option tells me what decoder is used? [16:37:46 CEST] <tinystoat> roxlu: wow sorry mate i completely misread what you were asking. [16:38:05 CEST] <roxlu> hehe np :) [16:46:56 CEST] <roxlu> ok, I just tried to start a conversion to check if that shows what decoder is used and I got ` Stream #0:0 -> #0:0 (h264 (native) -> theora (libtheora))` [16:47:08 CEST] <roxlu> I guess the "native" is the build in decoder? [16:47:14 CEST] <Anony11> Hi Fellas [16:47:28 CEST] <Anony11> Have a particular question: Have a program that generates continously a file, named "file.jpg" every millisecond for 10 seconds. Need that ffmpeg inputs(-i) the file continously and provide a "video.mp4" that is being "streamed"(keeps growing). [16:55:56 CEST] <dongs> same filename? [16:56:02 CEST] <dongs> is it getting overwritten? [16:57:09 CEST] <kepstin> you'll have a bad time if you're doing something like truncating it then rewriting it, due to race conditions. [16:57:15 CEST] <dongs> yeah every 10ms sounds kinda meh [16:57:17 CEST] <dongs> err every ms [16:57:21 CEST] <dongs> (even worse) [16:57:39 CEST] <kepstin> yeah, 1000fps is kind of a lot :) [16:58:07 CEST] <kepstin> need a hell of a computer to encode that in realtime, unless the images are really small. [16:58:29 CEST] <Anony11> Yeah, but those "were numbers"... lol [16:58:45 CEST] <dongs> anyway so? [16:58:53 CEST] <dongs> either 1) fix your program to number the jpegs [16:58:54 CEST] <Anony11> Yes, it's getting overwritten [16:59:37 CEST] <Anony11> It's to save space [16:59:37 CEST] <dongs> 2) do some filesystem hacking where it takes writen file and somehow sends it to ffmpeg on each open/write/close [16:59:37 CEST] <dongs> maybe lunix + fuse or some otehr garbage could do it [16:59:37 CEST] <Anony11> and resources, i can't keep getting filename_3billions.jpg + the file [16:59:37 CEST] <kepstin> Anony11: truncated and rewrite is *not* going to work, you'll get partial file reads and it'll be corrupt :/ [16:59:46 CEST] <kepstin> Anony11: you probably want to pipe your applications output to ffmpeg rather than write a file. [16:59:49 CEST] <dongs> do you want the output in realtime? [17:00:30 CEST] <Anony11> Oh, yes, even better then [17:00:31 CEST] <dongs> just pipe it as raw YUV/whatever [17:00:40 CEST] <kepstin> yeah, save time by not bothering with the jpeg encoding [17:01:00 CEST] <kepstin> (unless your high speed camera does hardware jpeg or something, i guess) [17:01:56 CEST] <Anony11> No, it's worse. It's a generated computer image, and need to "stream it", that's why i said jpg and possibly can't use raw data, cause who knows how it operates low level [17:02:31 CEST] <dongs> how much control do you ahve over whatever generates it [17:02:32 CEST] <Anony11> it's a square on sdl [17:02:34 CEST] <Anony11> lol [17:02:34 CEST] <kepstin> if it's a generated computer image, at some point it must exist as a bitmap in memory before it's encoded to jpeg... [17:02:44 CEST] <dongs> yeah SDL so uh [17:02:50 CEST] <dongs> you can certainly access to the bbackbuffer [17:03:41 CEST] <Freneticks> hey is it possible to make like a "rule" only rencode sound if it's eac3 ? [17:04:22 CEST] <Anony11> Yeah, but, sdl probably handles it trough opengl, so who knows... it's not really something "intermediate", yeah would be very fast, but, trying to "develop fast" rather than "execute fast" [17:04:54 CEST] <dongs> instead of saving to jpeg just pipe wahtever intermediate shit you get before that to ffmpeg directly. [17:05:22 CEST] <Anony11> R-really? would it work? [17:05:30 CEST] <dongs> why not? [17:05:43 CEST] <dongs> i mean you'll prolly need to edit sores either way [17:06:31 CEST] <Anony11> Cause, even tho of Italian Descent, Mario never tought me how to use pipes, and gave me Windows [17:07:15 CEST] <kepstin> Freneticks: no, you'd have to script it (e.g. by using ffprobe to check codecs before writing your ffmpeg command line) [17:07:31 CEST] <Freneticks> okay [17:11:32 CEST] <dongs> Anony11: pipes work on windows too. [17:11:59 CEST] <dongs> so instead of writing your shit to file everytime. maybe add some mjpeg header (if you really insist on using jpeg) and then feed that into ffmpeg [17:13:39 CEST] <kepstin> no special header needed, ffmpeg can read a stream of jpeg images from a pipe [17:13:45 CEST] <dongs> yea? [17:13:53 CEST] <dongs> just continuous JFIF or wahtever shit works? [17:13:56 CEST] <kepstin> but yeah, skipping the jpeg and streaming raw video would be better here. [17:14:03 CEST] <kepstin> yeah. [18:54:50 CEST] <StyXman> -pattern_type glob -i '*.jpg' is not working for me: Could not open file : *.jpg [19:30:53 CEST] <pink_mist> lol [19:48:18 CEST] <DHE> on the plus side, I doubt they'll make that mistake again any time soon [19:59:58 CEST] <steve___> I'd like to skip the first 1:40 of an mp4 video I took with my android point and add a video fade in and a video fade out. Should this be done in two steps, something like -ss 1:49.0 -c copy and then add the fades, to avoid re-encoding? [20:07:33 CEST] <DHE> it can be done in a single shot, but it will require a transcode step [20:07:54 CEST] <DHE> though you'll need to know how many frames are involved ahead of time, so a dry run may be required first [20:15:03 CEST] <steve___> DHE: So making a 01-fade_in.mp4 and a 03-fade_out.mp4 and then concat those three files would be faster/saner? [20:32:04 CEST] <Toneloc> Can ffmpeg be scripted to create videos such as this?? >> https://www.youtube.com/watch?v=qtdqJQqaajM [20:33:36 CEST] <Toneloc> I would like to import text data into ffmpeg and generate the video using a pre-defined graphocs, format (transistion, fade etc.) [20:34:47 CEST] <kepstin> Toneloc: sort of; one way to do that sort of thing would be to design the text as an ass subtitles file and use ffmpeg to render the subtitles file over the static background image. [20:35:49 CEST] <Toneloc> Would it be easy to embed an audio track that was made seperately also? [20:36:44 CEST] <kepstin> combining separate audio and video tracks into a single output file is probably one of the easiest things you can do with ffmpeg. [20:37:09 CEST] <Toneloc> kepstin - Thanks for that, do you know anywhere there is an example of this setup with ffmpeg? [20:37:48 CEST] <Toneloc> kepstin - I din't really mind if it's not ffmpeg, I'm just looking for an easy and cheap way to achieve this [20:37:59 CEST] <Toneloc> what do people usually use for this? [20:38:29 CEST] <kepstin> this is a kind of unusual use. but there's examples out there for putting subtitles onto a video with ffmpeg and combining audio/video tracks. The hardest part of doing this would be writing the subtitle file to use as input. [20:39:35 CEST] <Toneloc> kepstin - I'm sure I can write a program to write the subtitle file [20:39:52 CEST] <Toneloc> if the format is open, I'm sure its something like .srt ? [20:40:23 CEST] <kepstin> srt doesn't have much in the way of formatting; if you want to do positioning and animations and whatnot, you'd want to use the ass format. [20:41:24 CEST] <Toneloc> kepstin - thanks, I'll look into that [20:47:11 CEST] <DHE> steve___: no, pretty sure you can just do a single encode job with 2 fade commands, one for start and one for end. but you need an exact duration to set up the end fade. [21:32:47 CEST] <Janhouse> I have a problem where using -hwaccel vaapi crashes for me when trying to output to rtmp stream. But it works fine when writing to a file. [21:33:10 CEST] <Janhouse> This works fine: ffmpeg -hwaccel vaapi -hwaccel_device /dev/dri/renderD128 -hwaccel_output_format vaapi -i rtmp://10.10.12.2:1935/stream/YOLO -vf 'fps=30,scale_vaapi=w=1920:h=1080' -c:a copy -c:v h264_vaapi -b:v 6M -maxrate 6M out.mp4 [22:02:25 CEST] <an3k> git on Windows loves to do this and some code fails because of this [22:02:35 CEST] <an3k> nvm [23:12:01 CEST] <upgreydd> JEEB, sleeping? :D [23:13:53 CEST] <upgreydd> Is there an option to set `cpb_*` things in h264 ? What's related to? [00:00:00 CEST] --- Wed Jun 5 2019

1 0

ffmpeg-devel.log.20190603
by burek 04 Jun '19

04 Jun '19

[00:06:02 CEST] <cone-940> ffmpeg 03Mark Thompson 07master:1f8b36329f20: ffmpeg_hw: Mark some strings as const [00:06:02 CEST] <cone-940> ffmpeg 03Mark Thompson 07master:a4448637380d: ffmpeg_hw: Treat empty device string as no device setting [00:06:02 CEST] <cone-940> ffmpeg 03Mark Thompson 07master:d2141a9b652c: hwcontext_vaapi: Add option to specify connection type [00:06:02 CEST] <cone-940> ffmpeg 03Mark Thompson 07master:6b6b8a63713c: hwcontext_vaapi: Make default DRM device selection more helpful [00:06:02 CEST] <cone-940> ffmpeg 03Mark Thompson 07master:7f3f5a24a1c7: hwcontext_vaapi: Add option to set driver name [00:06:03 CEST] <cone-940> ffmpeg 03Mark Thompson 07master:0b4696fbe8db: hwcontext_vaapi: Try to create devices via DRM before X11 [00:06:03 CEST] <cone-940> ffmpeg 03Mark Thompson 07master:468f00384338: hwcontext_qsv: Try to select a matching VAAPI device by default [00:40:57 CEST] <mkver> gmail is running again. My patchset is already on the ML. [03:43:41 CEST] <cone-489> ffmpeg 03Jun Zhao 07master:360bee8ca49d: lavfi/colorlevels: Add slice threading support [03:43:41 CEST] <cone-489> ffmpeg 03Jun Zhao 07master:bbad0bc5ffcf: lavfi/lut: Add slice threading support [03:54:23 CEST] <cone-489> ffmpeg 03Ruiling Song 07master:94ceeba9f991: avfilter/vf_unsharp: enable slice threading [05:25:48 CEST] <nkk> Hi, I am getting "av_interleaved_write_frame(): Broken pipe" and stream stops with "Conversion failed!", what could be the reason for this? [06:56:34 CEST] <cone-592> ffmpeg 03Zhong Li 07master:165eabf19bf2: lavf/qsv: use av_cold for init/uninit [07:14:11 CEST] <cone-592> ffmpeg 03Jun Zhao 07master:51b0e812161d: lavf/sr: Dump input pixel format in error message [07:14:12 CEST] <cone-592> ffmpeg 03Jun Zhao 07master:5c1fbc42397b: lavf/sr: Don't need to check NULL before sws_freeContext [07:14:13 CEST] <cone-592> ffmpeg 03Jun Zhao 07master:0b7bfa8ad79e: lavf/sr: Refine the coding style for init [11:44:51 CEST] <durandal_1707> anybody working on porting rnnoise denoiser to lavfi? [14:21:39 CEST] <cone-592> ffmpeg 03Paul B Mahol 07master:97ddc5d1f741: avfilter/af_anlmdn: add support for commands [22:18:17 CEST] <cone-981> ffmpeg 03Mark Thompson 07master:7056002796ab: vaapi_encode: Remove unused function [22:18:17 CEST] <cone-981> ffmpeg 03Mark Thompson 07master:2880a32c6680: vaapi_encode: Refactor encode misc parameter buffer creation [22:32:07 CEST] <durandal_1707> anybody working on porting rnnoise denoiser to lavfi? [00:00:00 CEST] --- Tue Jun 4 2019

1 0

ffmpeg.log.20190603
by burek 04 Jun '19

04 Jun '19

[06:48:09 CEST] <xochilpili> hello everyone [06:48:53 CEST] <xochilpili> im doing some capture from my screen, first capture works fine, but when i start a new one i got: Input/output error i have check $DISPLAY env that is not changed [06:49:16 CEST] <xochilpili> im using this: ffmpeg -video_size 1920x1080 -framerate 25 -f x11grab -i :0.0 -f pulse -ac 2 -i output.mkv [06:49:38 CEST] <xochilpili> i also used: ffmpeg -video_size 1920x1080 -framerate 25 -f x11grab -i $DISPLAY -f pulse -ac 2 -i output.mkv << same error [11:13:26 CEST] <frappu> I just downloaded ffmpeg and I'm trying to convert a uhd bluray to a 1080p mkv but I don't know to set the resolution, bitrate, select the correct audio stream and convert it to aac or ac3 [11:16:07 CEST] <JEEB> see ffmpeg-all.html under ffmpeg.org and ctrl+f for zscale/tone mapping. for bit rate you usually don't set it unless you need a specific file size (instead some constant rate factor thing is what you'd use). for selection of streams, ctrl+F in that document "-map" [11:16:29 CEST] <frappu> thanks ;) [11:16:30 CEST] <JEEB> you want to (possibly crop), tone map, scale down [11:19:41 CEST] <durandal_1707> so now are all uhd blurays in wrong colorspace? [11:20:13 CEST] <JEEB> I'd say most are HDR [11:20:25 CEST] <JEEB> of course if it's not then tone mapping isn't needed :P [11:20:59 CEST] <JEEB> technically I think the spec lets you do BT.709 but those sweet marketing dollars need to be used on something ;) [11:21:24 CEST] <JEEB> and of course I made the assumption that the person wants to have a normal SDR HD encode [11:34:16 CEST] <an3k> What's the latest compatible version of nVIDIA CUDA? They started to split the npp libs at least with 9.2 so current code doesn't find nppi [11:36:41 CEST] <JEEB> you should attempt to poke the nvidia stuff related maintainers I guess? [11:44:04 CEST] <frappu> is it correct this part to select 4th audio stram and convert to ac3 160k ? -map -codec:a:4 ac3 160k [11:44:14 CEST] <an3k> CUDA SDK 8.0 is the latest that still ships with nppi.lib [11:44:31 CEST] <JEEB> frappu: 4th so beginning from 0 would be 3 [11:44:35 CEST] <an3k> Afaik external headers/libs aren't included anymore so one has to download them manually [11:44:37 CEST] <JEEB> and -map 0:a:3 [11:44:46 CEST] <JEEB> the codec part is on the selected streams [11:45:04 CEST] <JEEB> so if you only have a single audio track selected you can just use -c:a ac3 [11:45:06 CEST] <frappu> sorry, audio stream is the fifth, so 4 is correct [11:45:33 CEST] <JEEB> if you have multiple then -c:a:N where which of the selections of that type it is (if counting the maps) [11:45:44 CEST] <JEEB> because you apply encoding options on the selected streams [11:45:59 CEST] <JEEB> and for the audio bit rate it'd be -b:a or -b:a:N 160k [11:48:02 CEST] <frappu> -map 0:a:3 , 0 is for the video stream? [11:48:23 CEST] <JEEB> input [11:48:37 CEST] <JEEB> so if you have more than one input defined then that'd go upwards from 0 [11:48:45 CEST] <JEEB> in your case you probably have just one [11:48:47 CEST] <JEEB> so 0 [14:45:27 CEST] <lain98> a [14:48:11 CEST] <lain98> is there a way to know at runtime, what container formats are supported by a compression format ? [15:09:59 CEST] <DHE> no. it would be the other way around, and even then there isn't always a good listing. muxers only list their preferred/default codecs which are hence guaranteed to be available [15:37:53 CEST] <an3k> I put everything in mkv. If it contains Dolby Vision only mp4 can handle that correctly [16:54:26 CEST] <Gunstick> hi. I have a file with 5 mono channels. 1st 2 are muted. I want to line-graph the 3 others separately and mix them to a single audio stream. Current command line is: ffmpeg -t 20 -i You_Bore.wav -filter_complex "[0:a]showwaves=s=1280x720:split_channels=y:mode=line,format=yuv420p[v]" -map "[v]" -map 0:a -c:v libx264 You_Bore.mkv # source file is here: http://gkess.homeip.net/~georges/You_Bore.wav [16:58:59 CEST] <durandal_1707> Gunstick: use channelsplit filter [16:59:45 CEST] <lain98> what if i want to know the containers supported by a specific compression method [17:00:05 CEST] <lain98> before runtime [17:00:05 CEST] <furq> then you look it up online [17:00:10 CEST] <lain98> lol uh yeah [17:00:55 CEST] <lain98> ok, i got one more question [17:02:06 CEST] <lain98> i was putting some text in a video file using drawtext filter and when i play the video its there but when i just a third party library to decode the video, the text is missing [17:02:36 CEST] <lain98> is the text added using drawtext really burned into the stream or is it something else ? [17:02:47 CEST] <lain98> it could be a stupid question, im new to video [17:03:08 CEST] <furq> it's burned into the stream [17:03:21 CEST] <lain98> but then why when i decode it,its missing [17:03:42 CEST] <furq> i have no idea [17:03:46 CEST] <furq> are you sure you're decoding the right video [17:03:53 CEST] <lain98> i think so yeah [17:04:14 CEST] <furq> that's the best answer i can offer you [17:04:38 CEST] <lain98> so, its not like audio and video are different streams and so is text ? [17:04:53 CEST] <furq> you can have separate text streams but that's not what drawtext does [17:04:58 CEST] <lain98> okay [17:05:09 CEST] <lain98> must be that im doing something funny [17:05:14 CEST] <lain98> thanks [17:07:51 CEST] <lain98> could it be that the text is on a different video stream with transparent background, and they are overlayed ? [17:52:44 CEST] <kepstin> lain98: no, drawtext draws the text directly into the video frame. it's single-input, single-output. so, unless you have a weird filter chain where you've explicitly done drawtext over an extra transparent video stream then that's not the case. [18:01:38 CEST] <Soni> what's a good compressor if I want to have a 480p file, a 720p-480p file, a 1080p-720p file, etc? [18:01:53 CEST] <Soni> so that you need the smaller file to get to the bigger file [18:03:42 CEST] <DHE> I'm confused. can you rephrase that? [19:18:18 CEST] <kepstin> sounds like Soni wants something like h264 svc [19:19:18 CEST] <kepstin> Soni: it's not worth the effort in general, and it's not well supported by encoders or decoders. [19:19:31 CEST] <Soni> and can I use that in a browser? [19:19:42 CEST] <kepstin> Soni: no, you can't use it in a browser [19:19:44 CEST] <Soni> or in a torrent? [19:19:57 CEST] <Soni> (or in a torrent in a browser?) [19:20:19 CEST] <Soni> (peertube ftw) [19:20:40 CEST] <kepstin> Soni: just encode separate streams for each resolution and download only the one you want. [19:21:48 CEST] <Soni> kepstin: but then the lower bitrates will be poorly seeded [19:25:06 CEST] <kepstin> hmm, it looks like in theory browsers should play back vp9 multi-resolution encodes, but I have no idea how you'd actually make one [19:26:07 CEST] <kepstin> (and to do a torrent-like distribution, you'd have to use variable block size so you can request only the desired packets, and use mse to actually play it) [19:26:48 CEST] <Soni> or you could use the good old client-side javascript [19:27:07 CEST] <Soni> just split it into multiple torrents and add some metadata for recombining [19:28:14 CEST] <Soni> you don't want the bigger resolutions, no problem, just torrent the base torrent. you want the bigger resolutions then torrent multiple torrents. [19:28:34 CEST] <Soni> or maybe use multiple files in the same torrent or something. [19:55:41 CEST] <Necrosand> what is best video/audio container if you want the fastest seeking? [19:59:28 CEST] <DHE> mp4 should do you fine, but especially for network use you should build it properly [20:34:41 CEST] <hashworks> Hi! I have a bunch of files I need to encode, however I only want specific streams. I could map them directly, but there are different quantities and orderings of streams. I know I can drop/map specific languages with -map -m:language:ger, but does that work with formats as well? [20:40:02 CEST] <kepstin> Necrosand: one thing that affects seeking speed is the distance between keyframes - if keyframes are far apart, then you'll have a bit of lag as you seek from having to decode and drop the frames since the previous keyframe. [20:40:03 CEST] <Necrosand> e [20:40:53 CEST] <kepstin> (of course, this is a trade-off - shorter keyframe interval means higher bitrate video for the same quality) [22:42:08 CEST] <Necrosand> i had about 10 different android phones: i cannot do recording on all of them but one phone allowed me to: [22:42:08 CEST] <Necrosand> what did that one phone has that allowed me to do that, when 9 others didn't [22:42:08 CEST] <Necrosand> BIG REWARD [23:48:03 CEST] <Necrosand> ffmpeg -f dshow -i audio="Microphone" output.mp3 worked [23:48:03 CEST] <Necrosand> ffmpeg -f dshow -i audio="desktop audio" output.mp3 also worked [23:48:03 CEST] <Necrosand> ffmpeg -f dshow -i audio="desktop audio" -f dshow -i audio="Microphone" output.mp3 does NOT work [23:49:17 CEST] <pink_mist> I don't believe mp3 supports multiple audio streams [23:50:38 CEST] <Necrosand> then what format does [23:50:57 CEST] <Necrosand> pink_mist does ffmpeg support multiple audio streams though? [23:51:33 CEST] <pink_mist> you perhaps want to add a filter that takes two input audio streams and merges them [23:51:45 CEST] <pink_mist> ffmpeg supports it just fine [23:52:03 CEST] <pink_mist> and .mka should support multiple audio streams ... probably loads of other formats too [23:52:12 CEST] <pink_mist> but is it really multiple audio streams you want? [23:52:20 CEST] <pink_mist> like I said, you probably want a filter that merges them instead [23:52:51 CEST] <Necrosand> mka is just container [23:53:06 CEST] <Necrosand> do you mean aac ? [23:55:00 CEST] <Necrosand> ffmpeg -f dshow -i audio="desktop audio" -f dshow -i audio="Microphone" output.mka does NOT work and it created vorbis [23:55:12 CEST] <pink_mist> it's the container that matters for multiple audio streams [23:55:18 CEST] <pink_mist> in what way did it not work? [23:55:27 CEST] <Necrosand> it just records "desktop audio" [23:55:29 CEST] <Necrosand> not "mic" [23:55:50 CEST] <pink_mist> are you sure? it probably did both, but in two different audio streams [23:55:51 CEST] <pink_mist> like I said [23:56:03 CEST] <Necrosand> output.mka or output.mp3 [23:56:07 CEST] <pink_mist> ... [23:56:19 CEST] <Necrosand> i don't hear my mic [23:56:30 CEST] <pink_mist> switch to the other audio stream in the .mka then [23:57:53 CEST] <Necrosand> pink_mist is that even correctly formatted command ? [23:57:59 CEST] <pink_mist> no clue [23:58:00 CEST] <Necrosand> ffmpeg -f dshow -i audio="desktop audio" -f dshow -i audio="Microphone" output.mkv [23:58:04 CEST] <Necrosand> ffmpeg -f dshow -i audio="desktop audio" -f dshow -i audio="Microphone" output.mka [23:58:06 CEST] <pink_mist> I'm not a ffmpeg guru [23:58:15 CEST] <pink_mist> I just know mp3 can't handle multiple streams [23:58:17 CEST] <pink_mist> but mka can [23:58:18 CEST] <cehoyos> -map 0 -map 1 missing [23:58:18 CEST] <Necrosand> [14:48] <Necrosand> ffmpeg -f dshow -i audio="Microphone" output.mp3 worked [23:58:18 CEST] <Necrosand> [14:48] <Necrosand> ffmpeg -f dshow -i audio="desktop audio" output.mp3 also worked [23:59:02 CEST] <Necrosand> ffmpeg -map 0 -f dshow -i audio="desktop audio" -f dshow -i audio="Microphone" output.mka [23:59:07 CEST] <pink_mist> cehoyos: like I told him before though, he probably wants a filter that merges them into a single stream [23:59:13 CEST] <pink_mist> but he refuses to listen to me [23:59:17 CEST] <cehoyos> He did not answer that he needs such a filter [23:59:30 CEST] <cehoyos> If he wants multiple streams, he has to use the map option [23:59:31 CEST] <pink_mist> he's just flailing around [23:59:37 CEST] <Necrosand> pink_mist sorry i am not understanding [23:59:46 CEST] <Necrosand> i just one one output.mka file [23:59:47 CEST] <cehoyos> It would probably make more sense if he tried the option that I suggested instead of a different one;-) [23:59:48 CEST] <pink_mist> Necrosand: do you want to listen to BOTH STREAMS AT THE SAME TIME? [23:59:51 CEST] <Necrosand> i just want one output.mka file [00:00:00 CEST] --- Tue Jun 4 2019

1 0

ffmpeg-devel.log.20190602
by burek 03 Jun '19

03 Jun '19

[00:34:20 CEST] <keegans> i assume this is the right place to ask ; i am trying to debug what is seemingly heap corruption caused by http.c when reading from HLS streams with large segments [00:34:54 CEST] <keegans> it receives data w/ length header='Content-Length: 351823', then says it's opening for reading and immediately segfaults directly after [00:42:59 CEST] <jdarnley> that does sound bad [00:43:19 CEST] <jdarnley> you need to run it in a debugger [00:43:55 CEST] <jdarnley> run a debug build of course [01:13:36 CEST] <keegans> jdarnley: yeah i have a debug build in a debugger [01:13:48 CEST] <keegans> when i run it in valgrind, it works as intended [01:14:05 CEST] <keegans> which complicates things slightly [01:14:29 CEST] <keegans> where does it allocate memory for the body, can i increase the size and see if that changes anything ? [02:02:08 CEST] <keegans> are there docs on how i can compile ffmpeg w/ asan [02:02:22 CEST] <keegans> " --extra-cflags="-fsanitize=address -fno-omit-frame-pointer" --extra-ldflags="-fsanitize=address -fno-omit-frame-pointer"" is not seeming to work [02:06:55 CEST] <keegans> --toolchain=clang-asan [02:18:52 CEST] <keegans> ==720==ERROR: AddressSanitizer: heap-buffer-overflow on address 0x602000029f79 at pc 0x55aa18c5ae80 bp 0x7fff04569cf0 sp 0x7fff04569ce0 [02:18:59 CEST] <keegans> yeah seems i've found a heap buffer overflow [02:23:14 CEST] <keegans> no something else is wrong [02:23:24 CEST] <keegans> it crashes at different points in and out of asan [02:23:30 CEST] <keegans> asan's stacktrace is wrong [02:39:45 CEST] <iive> keegans, valgrind has an option to avoid reusing same heap memory, to easy catching use-after-free [02:41:10 CEST] <iive> aka, allocate buffer, free buffer, then allocate same memory by another routine. if the first one is still using it, it won't register as use-after-free [02:41:29 CEST] <keegans> do you know what option it is ? [02:44:45 CEST] <iive> -memcheck:freelist-vol=512000000 [02:45:43 CEST] <iive> check also malloc-fill and free-fill [02:46:18 CEST] <keegans> seems that when i do that it has the same effect as asan, it causes the crash to be at a different place [02:46:59 CEST] <iive> that's great, it didn't use to crash :) [02:48:13 CEST] <keegans> well it means that i cannot use it to find the cause of that crash [02:48:21 CEST] <iive> depends [02:48:29 CEST] <iive> valgrind tracks memory too. [02:49:15 CEST] <iive> not only execution. does it try to access previously used memory region? [02:51:10 CEST] <keegans> it doesn't look like it [02:52:58 CEST] <keegans> right [02:53:04 CEST] <keegans> so it doesn't crash at all in valgrind still [02:53:12 CEST] <keegans> i was wrong, it triggered an abort [02:53:21 CEST] <iive> hum? [02:53:32 CEST] <keegans> so it runs as intended inside of valgrind with the option you provided [02:53:41 CEST] <keegans> and intended means no SIGSEGV [02:53:49 CEST] <keegans> whereas when i run it outside of valgrind it crashes [02:54:08 CEST] <keegans> which is the issue i had before iused that option [02:54:40 CEST] <iive> have you tried the -fill options? [02:55:32 CEST] <keegans> no , not yet [02:55:51 CEST] <keegans> which should i be using [02:56:04 CEST] <keegans> or just both malloc and free fill [02:56:08 CEST] <iive> all of them [02:56:44 CEST] <iive> one would fill the memory with pattern when you allocate it, so if you use uninitialized pointer, it would be similar to that pattern. [02:57:23 CEST] <keegans> yeah that crashes it [02:57:29 CEST] <keegans> but in the wrong place, still [02:57:43 CEST] <iive> the other would do the same with the memory when you free it, so use-after-free would show that pattern. [02:57:49 CEST] <iive> what is the wrong place? [02:58:00 CEST] <keegans> let me try and illustrate with logs, one second [02:58:05 CEST] <iive> does abourt come from an assert? [02:58:24 CEST] <keegans> it was my abort [02:58:41 CEST] <keegans> it was there to show that the program crashed before the abort [03:03:12 CEST] <iive> aha [03:05:34 CEST] <keegans> i am just making a writeup [03:05:35 CEST] <keegans> so you can see [03:06:26 CEST] <iive> i'll be leaving in a moment. [03:07:25 CEST] <keegans> iive: https://gist.github.com/meme/30c5cf45dc9b10761ba53fd068c04b30 [03:07:53 CEST] Action: iive looking [03:11:10 CEST] <iive> 0x602000029f79 is located 0 bytes to the right of 9-byte region [0x602000029f70,0x602000029f79) allocated by thread T0 here: #0 0x7f1f7d23b289 in operator new[](unsigned long) [03:12:03 CEST] <iive> ok, this is C++ code, so I assume this is your program, and it is reading right after the buffer, so I guess it is overread. [03:12:30 CEST] <keegans> right but it crashes before it gets to my code [03:12:32 CEST] <iive> all ffmpeg memory allocations are done with extra number of bytes, there is some ugly define for it. [03:12:48 CEST] <keegans> ah yeah if i can find that i'll just bump it up and see if that fixes it [03:12:51 CEST] <keegans> "fixes" [03:13:01 CEST] <keegans> do you know what the define is ? [03:13:15 CEST] <iive> #1 0x55aef37abbf9 in Host::DecryptForEach() [03:13:22 CEST] <iive> i may find it... [03:17:33 CEST] <iive> AV_INPUT_BUFFER_PADDING_SIZE [03:17:53 CEST] <keegans> thank you [03:19:30 CEST] <iive> it is defined in libavcodec/avcodec.h [03:19:39 CEST] <iive> good night and good luck. [11:10:35 CEST] <cone-164> ffmpeg 03Paul B Mahol 07master:cbaa60329a73: avfilter/vf_xmedian: remove limitation of only odd number of inputs [13:03:38 CEST] <cone-164> ffmpeg 03Paul B Mahol 07master:b5355774652c: avfilter/af_anlmdn: add smooth factor option [13:03:39 CEST] <cone-164> ffmpeg 03Paul B Mahol 07master:1a266a1ef91d: avfilter/af_anlmdn: try to recover when cache becomes negative [13:03:40 CEST] <cone-164> ffmpeg 03Paul B Mahol 07master:fcfe85220dbd: avfilter/af_anlmdn: avoid creating frames with zero samples [13:20:26 CEST] <durandal_1707> mplayer is becoming pathetic project [13:41:43 CEST] <JEEB> the wget gnutls code is a bit more verbose http://git.savannah.gnu.org/cgit/wget.git/tree/src/gnutls.c#n476 [13:41:54 CEST] <JEEB> but still seems to have the same basic idea [13:54:54 CEST] <JEEB> and curl also does a similar thing in an eternal loop https://github.com/curl/curl/blob/master/lib/vtls/gtls.c#L332 [14:16:05 CEST] <durandal_1707> JEEB: quick! rewrite every loop in ffmpeg so it does not loop forever [14:16:23 CEST] <JEEB> stop being facetious, thank you [14:18:52 CEST] <durandal_1707> JEEB: michaelni is silent, expect from him only patch if you apply that patch [14:19:39 CEST] <JEEB> I'll just double-check how I/O timeouts are handled. otherwise having that eternally looping is OK [14:19:56 CEST] <JEEB> mostly it could later be improved with a logging patch a la curl [14:20:00 CEST] <JEEB> but that's separate from the fix [14:20:03 CEST] <JEEB> :P [14:27:27 CEST] <durandal_1707> so one can not rescale dvdsub subtitles at all? [14:28:04 CEST] <JEEB> you decode the AVSubtitles, then if you want to use filtering you make AVFrames out of them like with sub2video [14:28:14 CEST] <JEEB> then you see if you can get those into AVSubtitles again [14:28:15 CEST] <JEEB> :P [14:43:29 CEST] <durandal_1707> ubitux: please send patches! [14:44:54 CEST] <JEEB> 32 [14:45:01 CEST] <durandal_1707> 31 [14:53:52 CEST] <ubitux> sorry i'm overhelmed :( [15:55:22 CEST] <cone-164> ffmpeg 03Steven Liu 07master:4ef0bea292d2: doc/muxers: fix typo of the hls var_stream_map example [16:31:12 CEST] <kierank> durandal_1707: why [16:31:12 CEST] <kierank> is mplayer pathetic [17:26:42 CEST] <durandal_1707> kierank: patch with intrinsic code for old filters nobody use [18:16:44 CEST] <JEEB> > MP4RA Scanning specifications for possibly-unregistered 4CCs [18:44:44 CEST] <cone-164> ffmpeg 03Mark Thompson 07master:f1b359aaf5c3: vf_crop: Add support for cropping hardware frames [18:44:45 CEST] <cone-164> ffmpeg 03Mark Thompson 07master:963c4f85fe54: doc/indevs: Add example using cropping to capture part of a plane [18:44:46 CEST] <cone-164> ffmpeg 03Mark Thompson 07master:6ed34a437925: lavfi/vaapi: Factorise out common code for parameter buffer setup [18:44:47 CEST] <cone-164> ffmpeg 03Mark Thompson 07master:5fb9eb9ed256: vf_misc_vaapi: Add missing return value checks [18:44:48 CEST] <cone-164> ffmpeg 03Mark Thompson 07master:5051b7f898ae: lavfi/vaapi: Improve support for colour properties [18:44:49 CEST] <cone-164> ffmpeg 03Mark Thompson 07master:ef2f89bbccc9: vf_scale_vaapi: Add options to configure output colour properties [18:44:50 CEST] <cone-164> ffmpeg 03Mark Thompson 07master:909bcedc581a: vaapi_encode: Warn if input has cropping information [19:19:32 CEST] <cehoyos> JEEB: vlc writes its own internal 4CCs into mov files [20:14:28 CEST] <durandal_1707> cehoyos: why mplayer now have intrinsic code?, that is pathetic! [20:14:45 CEST] <cehoyos> Please elaborate! [20:17:44 CEST] <BradleyS> TIL handbrake is pathetic [20:18:34 CEST] <durandal_1707> cehoyos: http://lists.mplayerhq.hu/pipermail/mplayer-cvslog/2019-May/047359.html [20:19:13 CEST] <durandal_1707> BradleyS: handbrake have intrinsic code but not abandoned filters [20:19:26 CEST] <BradleyS> true [20:19:37 CEST] <BradleyS> what's wrong with vf_pullup? nobody needs ivtc? [20:20:02 CEST] <cehoyos> I didn't know this, thank you for the link. [20:20:21 CEST] <cehoyos> I don't think anything is wrong with vf_pullup, is it? [20:23:02 CEST] <durandal_1707> he wrote intrinsic code to code nobody touched for ages - that is very pathetic, and at same time refuses to update codecs.conf [20:25:02 CEST] <JEEB> how and why does that offend you as much? why the name-calling? :P [20:25:34 CEST] <JEEB> or are you just trying to do the usual thing with trying to get attention? [20:26:18 CEST] <durandal_1707> JEEB: yes, i need your 24h attention to apply my pull request [20:27:07 CEST] <durandal_1707> it offend me very much, they are ruining project [20:35:42 CEST] <BradleyS> can't blame others for how you act ;) [20:43:23 CEST] <durandal_1707> huh, github allows now to report security stuff for each project [20:43:48 CEST] <durandal_1707> BradleyS: i did nothing [20:46:34 CEST] <durandal_1707> what should be name of filter?: _f360_ or _panorama_ or _perigon_ [20:59:24 CEST] <durandal_1707> you get 1k$ if you pick! [21:04:47 CEST] <durandal_1707> nobody wants free money :-( [21:15:12 CEST] <durandal_1707> good, i'm staying rich [21:50:22 CEST] <durandal_1707> gmail is down - i can not work this way [22:08:13 CEST] <BradleyS> _360pano [22:08:25 CEST] <BradleyS> or _pano360 [22:16:20 CEST] <durandal_1707> not cool enough [23:12:34 CEST] <mkver> jkqxz: I completed the cbs-mpeg2 patches, but somehow gmail is making problems. [23:12:48 CEST] <JEEB> yea they're having issues atm [23:12:54 CEST] <mkver> You can nevertheless already take a look: https://github.com/mkver/FFmpeg/commits/mpeg2-cbs-2 [23:14:11 CEST] <mkver> Your warnong wrt __VA_ARGS__ was unfounded in this case: SUBSCRIPTS already expands to "(0 > 0 ? ((int[0 + 1]){ 0, }) : NULL)" if there are no further arguments; this does not change and it is legal C. [23:31:52 CEST] <jkqxz> I was actually thinking of the multiple macro expansions. But yeah, it's fine because foo(a,) and foo(a) end up with the same empty __VA_ARGS__ (unlike if they were functions). [23:33:51 CEST] <jkqxz> Is github display just terrible or are there weird misalignments in that? [23:35:36 CEST] <mkver> Where exactly? [23:36:10 CEST] <_bluez> Hello, I had some questions regarding the implementation of heif support, should the heif (de)muxer be extended as another member in the mov class or should it be a seperate one? [23:36:48 CEST] <JEEB> it generally sounds like it'd utilize quite a bit of the mov/mp4 (de)muxer [23:36:56 CEST] <JEEB> so if not in the same file, but at least very close to mov(enc) [23:37:18 CEST] <jkqxz> mkver: E.g. on <https://github.com/mkver/FFmpeg/commit/b42203510f03edc38060e49840b37309db0f…>. Looks like it's github inserting tabs for no reason and messing up the spacing, it's fine in the actual patch. [23:37:22 CEST] <JEEB> I think we had an initial demuxer patch posted by a person a year or two ago [23:37:27 CEST] <_bluez> yeah i thought so too [23:37:38 CEST] <JEEB> the problme mostly was by the fact that it was preliminary, and then the whole issue with multi-part images [23:38:00 CEST] <_bluez> ohh,where can i find this patch? [23:38:05 CEST] <JEEB> https://ffmpeg.org/pipermail/ffmpeg-devel/2017-August/215003.html [23:39:12 CEST] <_bluez> Thank You! [23:39:18 CEST] <JEEB> but since we effectively cannot handle the stuff nicely on our abstraction level I guess the first step would be to get reading of that stuff in with nice enough code [23:39:36 CEST] <JEEB> I think we already have a flag for image streams [23:40:17 CEST] <JEEB> so in a way muxer would be simpler because you can close your ears from all the thoughts of how to expose multiple part images in lavf+lavc [23:40:46 CEST] <JEEB> and in demuxer the actual demuxer part probably wouldn't be the worst part of it, but rather defining how do we tell the API users that "btw, here's some images which are part of a bigger image" [23:41:08 CEST] <JEEB> stuff like image part ordering etc [23:41:24 CEST] <JEEB> the libraries with higher level APIs have it easier :P [23:41:51 CEST] <JEEB> because f.ex. in imagemagick you just ask for the decoded image and don't really care what it internally does (does it have multiple decoders or what) [23:42:06 CEST] <JEEB> (or how does it stitch the image together etc) [23:42:29 CEST] <JEEB> in our case it'd be a HEIF demuxer returning multiple images, somehow signaled to be a part of a bigger thing... [23:42:44 CEST] <JEEB> so that the API user can then know to stitch them into a single image [23:42:59 CEST] <jamrial> jkqxz: looks fine here. maybe a monitor resolution thing? [23:45:02 CEST] <jkqxz> I see <http://ixia.jkqxz.net/~mrt/alignment.png>. [23:45:12 CEST] <_bluez> isn't it the task of the reader (demuxer) to know how to stich the images? [23:45:12 CEST] <mkver> I don't get tabs and it looks fine in github's preview, but when I copy it, I somehow end up with a newline that is not in actual patch: " uir(8, transfer_characteris\ntics);" [23:45:35 CEST] <JEEB> _bluez: yes, it would still have to be signaled somehow so that the actual stitching can be done after decoding [23:45:48 CEST] <JEEB> _bluez: otherwise you'd just have to guess as you receive the partial images [23:46:29 CEST] <_bluez> There are indeed many little complicated things like tiling and all, but i guess i shoud first focus on a basic implementation first ':D [23:46:38 CEST] <JEEB> sure [23:49:52 CEST] <_bluez> This field is new to me, i only started a month or two before, so i might have some more silly questions i hope you dont mind, for now i'll have a look at that patch.. [23:50:05 CEST] <JEEB> feel free to ask things :) [23:50:25 CEST] <_bluez> Thanx :D [23:50:40 CEST] <jamrial> jkqxz: https://0x0.st/zMxc.jpg [23:50:54 CEST] <jamrial> maybe it's your browser [23:51:42 CEST] <jkqxz> Weird. Oh well, the patch itself is right; it's just rather distracting to read it there. [23:53:41 CEST] <mkver> Why don't you just download them from github and read them with any tool you like? [23:56:47 CEST] <jkqxz> Because github looks (apparently misleadingly) like you can sanely read patches on the site. [00:00:00 CEST] --- Mon Jun 3 2019

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ffmpeg.log.20190602
by burek 03 Jun '19

03 Jun '19

[16:28:10 CEST] <giaco> hello [16:32:04 CEST] <giaco> I want to stream audio from a source over a non-reliabe link. When link is not working the stream should be saved locally and sent later. Now I am using ffmpeg to convert source to mpeg-dash and an rsync loop to cover the transmission part. I wonder if I have other options, so I'm here just for feedbacks and ideas. Thank you [22:30:22 CEST] <Leoneof> hi, i have videos.mp4 files recorded by hikvision dvr, they are not working in any media player & cannot import them to any video editor, so what's the proper command-line to make videos.mp4 readable with any media appication? [22:31:15 CEST] <JEEB> can you get anything useful out of those files with `ffprobe -v verbose -i FILE` [22:31:21 CEST] <Leoneof> ok [22:31:38 CEST] <JEEB> post on a pastebin or so if it seems to output anything useful, and link here [22:50:14 CEST] <cehoyos> Leoneof: Please provide a sample input file [22:53:38 CEST] <Leoneof> ok, i'm suck with windows cmd, just one minute [22:57:22 CEST] <Leoneof> https://pastebin.com/raw/PYQdTESA [22:57:46 CEST] <JEEB> looks like normal mp4, except the audio is mulaw PCM [22:58:31 CEST] <Leoneof> I have other video without audio, it's not working too [23:00:26 CEST] <JEEB> ffmpeg -v verbose -i INPUT.mp4 -c:v copy -c:a aac -b:a 192k -af 'aformat=channel_layouts=stereo' test.mp4 [23:00:29 CEST] <JEEB> try this? [23:00:42 CEST] <Leoneof> k [23:01:56 CEST] <JEEB> maybe you'll also have to set the rate to 44100Hz or whatever [23:02:29 CEST] <JEEB> ffmpeg -v verbose -i INPUT.mp4 -c:v copy -c:a aac -b:a 192k -af 'aformat=sample_rates=44100:channel_layouts=stereo' test.mp4 [23:03:49 CEST] <JEEB> (or 48000 since that's exactly six times 8000) [23:09:46 CEST] <Leoneof> JEEB: [AVFilterGraph @ 0000000002bfef80] No such filter: 'aformat=sample_rates=44100:channel_layouts=stereo' [23:09:50 CEST] <Leoneof> Error reinitializing filters! [23:09:58 CEST] <Leoneof> Failed to inject frame into filter network: Invalid argument [23:10:04 CEST] <Leoneof> Error while processing the decoded data for stream #0:1 [23:11:15 CEST] <JEEB> Leoneof: ok, forgot aresample before [23:11:31 CEST] <JEEB> "aresample,aformat=sample_rates=44100:channel_layouts=stereo" [23:15:01 CEST] <Leoneof> JEEB: it's converting right now [23:18:13 CEST] <Leoneof> JEEB: wow thank you very much, now i can watch videos in any media player, and video editor [23:18:50 CEST] <JEEB> and all we did was remux the video and re-encode the audio to a more common setup of stereo+44.1kHz AAC [23:19:00 CEST] <JEEB> that's why -c:v copy [23:19:14 CEST] <JEEB> (sets the "codec" for video to "copy") [23:19:46 CEST] <Leoneof> :) [00:00:00 CEST] --- Mon Jun 3 2019

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