January 2020 - Ffmpeg-devel-irc

ffmpeg-devel.log.20200110
by burek 11 Jan '20

11 Jan '20

[00:00:00 CET] --- Fri Jan 10 2020

1 0

ffmpeg.log.20200110
by burek 11 Jan '20

11 Jan '20

[00:27:04 CET] <tytan> hello everyone, I kinda need help with a task I am unable to perform. I have a video with two stereo audio tracks and one of those is the standard audio track. can I reencode the file with the second track to be standard instead? [00:29:01 CET] <pink_mist> shouldn't need to reencode for that, remux or perhaps just changing some metadata should be all you need ... I'm not the guy to ask for how though, sorry [00:33:13 CET] <tytan> ok, what if I do a -cv copy and shuffle the audio tracks? [00:48:14 CET] <Beam_Inn> would using ffmpeg to double the speed of a video also automatically adjust its pitch? [00:48:54 CET] <Beam_Inn> Lypheo thank you so much. Do you have more docs about making animated gifs? Someone at imagemagik told me ffmpeg would be a better option for making animated gifs. [00:50:11 CET] <tytan> have a nice day everyone [00:58:03 CET] <kepstin[m]> Beam_Inn: if you use a filter that modifies only the video speed, then the audio speed will be untouched. Depending on your goals, the "rubberband" filter is good for adjusting audio speed. [01:25:35 CET] <hseg> Hi. Am having trouble figuring out the correct format to use for my Samsung TV. Have one file that is accepted, and one that isn't, but can't tell what the relevant difference is. [01:27:16 CET] <hseg> Working: http://ix.io/26YY, http://ix.io/26YZ Not working: http://ix.io/26Z2, http://ix.io/26Z4 [01:28:31 CET] <hseg> The Samsung manual is here: http://downloadcenter.samsung.com/content/UM/201408/20140804154219440/[UH42… [01:28:50 CET] <nicolas17> I'd say "has to be baseline" [01:28:59 CET] <nicolas17> but then it's really weird that "constrained baseline" doesn't also work [01:30:09 CET] <nicolas17> wait [01:30:12 CET] <nicolas17> hseg: could it be the fps? [01:32:59 CET] <hseg> checking... [01:33:53 CET] <hseg> Hrm, am seeing sub-30 fps in supported videos, ~60 in unsupported [01:35:22 CET] <furq> it does specify 6-30fps for h264 in the manual [01:35:28 CET] <furq> that would be really weird though [01:35:37 CET] <hseg> Why? [01:38:11 CET] <Beam_Inn> well..., kepstin, do you have a command for slowing a video and its audio without causing it to sound like chipmonks? [01:44:01 CET] <kepstin[m]> Yes, use the rubberband filter to adjust the audio speed. [01:47:06 CET] <hseg> hrm, ran through files, getting some files with lower fps, lemme check whether those work [01:48:13 CET] <nicolas17> hseg: baseline vs high could *still* be a problem, but I can't tell from your previous testcases [01:48:19 CET] <nicolas17> see if 30fps high works :P [01:48:48 CET] <hseg> lemme check [01:49:43 CET] <furq> also make sure the files are level 4.1 or lower [01:52:33 CET] <Beam_Inn> how long have rubberband filters existed? [01:53:42 CET] <Beam_Inn> according to https://ffmpeg.org/ffmpeg-filters.html#rubberband (secion 8.81) the rubberband filter needs to be flagged before compilation. [01:56:17 CET] <pink_mist> I mean, that's true of almost every external library ffmpeg supports [01:56:33 CET] <pink_mist> why does this need to be noted? [01:56:46 CET] <Beam_Inn> i want to add it to my log file. [01:57:00 CET] <Beam_Inn> I haven't compiled ffmpeg yet. [01:57:08 CET] <hseg> OK, so the low-framerate videos work [01:57:17 CET] <hseg> Side question - is m4v=mp4? [01:57:25 CET] <Beam_Inn> mp4 [01:57:56 CET] <Beam_Inn> They're just very similar. [01:59:09 CET] <hseg> So a simple rename should be fine [01:59:51 CET] <nicolas17> note that iTunes uses m4v extension [02:00:01 CET] <Beam_Inn> no. [02:00:34 CET] <Beam_Inn> you can't just rename an m4v file to *.mp4 and have a working output. [02:00:37 CET] <nicolas17> which means it *might* be encrypted with FairPlay-DRM... but if your m4v didn't come from iTunes it should be just a matter of renaming [02:01:08 CET] <hseg> It's my own video, and I don't recall encrypting it [02:01:20 CET] <Beam_Inn> Yes, I agree with nicolas17. [02:01:21 CET] <hseg> Eh, renaming is non-destructive [02:03:46 CET] <hseg> OK, have more working cases, all of which are ~30fps [02:04:03 CET] <hseg> or 25fps [02:04:24 CET] <hseg> how do I reencode a 60fps video to 30fps? [02:07:29 CET] <kepstin[m]> m4v and mp4 may indicate different brands of isombff files, but that's detected when they're opened and read. [02:07:50 CET] <kepstin[m]> So no real effect to renaming them [02:08:04 CET] <nicolas17> kepstin[m]: it's a Smart TV [02:08:11 CET] <nicolas17> it could be dumb and only look for .mp4 extensions [02:08:21 CET] <nicolas17> and refuse to load anything else [02:08:37 CET] <hseg> I [02:08:50 CET] <hseg> That's exactly what happened [02:55:38 CET] <hseg> Success! [02:55:43 CET] <hseg> Thanks [03:15:37 CET] <wondiws> hi furq [10:09:14 CET] <yang> Is this the right line to convert Matroska -> AVI and flac -> mp3 ? [10:09:16 CET] <yang> ffmpeg -i out0021.mkv -c:v libx264 -acodec libmp3lame -b:a 128k -channels 2 -ar 48000 -crf 23 -c:a copy -threads 1 outmp3.avi [10:09:50 CET] <Mavrik> Huh, wat. [10:13:05 CET] <yang> converting the file [10:13:19 CET] <yang> from Matroska/Flac to AVI/MP3 [10:15:47 CET] <Mavrik> Creating an AVI with H.264 in it is not a reasonable thing to do in 2019 [10:15:50 CET] <Mavrik> or 2020 [10:15:56 CET] <Mavrik> What are you trying to actually do? [10:16:13 CET] <yang> I am trying to make a file playable by hardwarwe DVD player [10:16:24 CET] <yang> best would be DVD [10:16:35 CET] <Mavrik> And does your DVD player support H.264 video in AVI? That doesn't seem very likely. [10:16:42 CET] <yang> but maybe4 the DVD player can aswell play AVI [10:16:49 CET] <yang> I don't know [10:16:52 CET] <Mavrik> AVI is just a container [10:16:56 CET] <Mavrik> It's also important what is inside it [10:17:03 CET] <Mavrik> Can you look at the spec sheet? [10:17:24 CET] <yang> no, I don't have the DVD player nearby...it's for anotehr location [10:17:42 CET] <Mavrik> The older era DVD players supported only MPEG-2 and DivX/Xvid content [10:17:53 CET] <Mavrik> So it's more likely you'll have to put divx in it [10:17:57 CET] <yang> Best would be to make DVD out of Matroska/FLAC, that would be playable by the DVD [10:17:57 CET] <Mavrik> with -c:v libxvid [10:18:14 CET] <yang> ok [10:18:27 CET] <yang> i assumed that AVI = DivX [10:20:21 CET] <yang> ffmpeg -i out0021.mkv -c:v libxvid -acodec libmp3lame -b:a 128k -channels 2 -ar 48000 -crf 23 -threads 1 outmp3.avi [10:20:24 CET] <yang> like this ? [10:21:12 CET] <yang> I get this warning [10:21:14 CET] <yang> Codec AVOption crf (Select the quality for constant quality mode) specified for output file #0 (Berlinmp3.avi) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream. [10:23:20 CET] <vlt> yang: If you assumed "AVI = DivX", why did you specify "-codec:v libx264"? And why "-codec:a libmp3lame" AND "-codec:a copy"? [10:23:43 CET] <yang> vlt: it was a command line from previous conversion [10:23:54 CET] <Mavrik> yang, yeah, libxvid doesn't listent to crf, you probably need to specify bitrate with -b:v [10:27:14 CET] <yang> what is a bitrate, with previous conversion I had high bitrate 14000, now its averaging at 2000 [10:35:56 CET] <th3_v0ice> Why would i get segmentation fault here avfilter_graph_config(), while using CUDA? I created a HW device, used av_buffer_ref to reference it with AVCodecContext, used av_hwframe_ctx_alloc() and av_hwframe_init() to init HWFrameCtx for encoder and decoder. And when I try to open scale_cuda filter I get segmentation fault. Line printed before the error happens w:1920 h:1080 pixfmt:cuda tb:1/25 fr:25/1 sar:1/1 sws_param:flags=2 [11:19:38 CET] <CounterPillow> yang: a bitrate is the rate of information per time interval, so 2000 kbit/s is 2000 kbit (250 kByte) per second. With a higher bitrate, you (generally speaking) get better quality, because the encoder can use more information to compress the input data into. [11:23:57 CET] <CounterPillow> a good codec and encoder allows you to achieve the same quality at a lower bitrate compared to other codecs and encoders. [11:23:57 CET] <CounterPillow> not all scenes are equally difficult to compress, so various mechanisms of adjusting the bitrate dynamically have been invented, for example ABR (reaching an average bitrate of a certain value over a certain span of time, but letting the actual bitrate within that fluctuate as is needed), VBR (variable bitrate, aiming for a target bitrate but having a maximum bitrate to cap it by, letting the rate fluctuate as it's needed otherwise) and [11:23:57 CET] <CounterPillow> CRF (constant rate factor, trying to keep the quality consistent throughout and adjusting the bitrate as is necessary) [11:29:31 CET] <yang> thank you CounterPillow [11:59:14 CET] <yang> ffmpeg -i out0021.mkv -c:v libxvid -acodec libmp3lame -b:a 128k -channels 2 -ar 48000 -crf 23 -threads 1 outmp3.avi [12:00:03 CET] <yang> this came out ok, except video isn't very clean, so how do I apply a higher bitrate options with -b:v ? [12:01:01 CET] <yang> Duration: 01:03:34.16, start: 0.000000, bitrate: 1088 kb/s [12:01:05 CET] <yang> this is default bitrate [12:01:18 CET] <yang> probably I need to do something like [12:01:35 CET] <yang> ffmpeg -i out0021.mkv -c:v libxvid -acodec libmp3lame -b:a 128k -channels 2 -ar 48000 -b:v 3000 -threads 1 outmp3.avi [12:02:41 CET] <CounterPillow> probably 3000k, not 3000, though I'm not sure if ffmpeg accepts the latter too [12:02:46 CET] <yang> ok yes [12:05:12 CET] <CounterPillow> Instead of setting a bitrate with -b:v, you can try setting a video quality level with -qscale:v with a value ranging from 1 to 31 with 1 being the highest quality [12:06:31 CET] <CounterPillow> (note that a value of 1 is not recommended, as this will be unnecessarily large) [12:07:40 CET] <CounterPillow> I'd start with -qscale:v 10 and adjust it up if you notice no video degradation, or down if you do notice it [12:31:02 CET] <yang> ok CounterPillow [13:30:32 CET] <th3_v0ice> Why would i get segmentation fault here avfilter_graph_config(), while using CUDA? I created a HW device, used av_buffer_ref to reference it with AVCodecContext, used av_hwframe_ctx_alloc() and av_hwframe_init() to init HWFrameCtx for encoder and decoder. And when I try to open scale_cuda filter I get segmentation fault. Line printed before the error happens w:1920 h:1080 pixfmt:cuda [13:30:38 CET] <th3_v0ice> tb:/25 fr:25/c1 sar:1/1 sws_param:flags=2 [13:30:47 CET] <th3_v0ice> Sorry for double posting. [15:09:36 CET] <plumbe0> Hi, i'm trying to create a video with a base solid color and an image overlay that needs to be resized and stay in the center. the video should also have a song playing and should be as long as the audio file. This is my command: ffmpeg -f lavfi -i color=c=0xc03a08:s=1920x1080 -i .\a.png -i '.\audio.flac' -filter_complex [15:09:36 CET] <plumbe0> "[1:v]scale=1000:1000[v1];[0:v][v1]overlay=(W-w)/2:(H-h)/2[out]" -map "[out]" output.mp4 ; the problem is that i don't have audio and the encoding doesn't stop until i press Ctrl+C [15:11:49 CET] <BtbN> Well, you didn't map the audio. [15:12:05 CET] <BtbN> And never told it to use the shorter of the two inputs. [15:14:28 CET] <plumbe0> How can i do both? i tried adding [2:a] in the filter chain but i don't know where to put it. also i tried to add "shortest=1" after the overlay but i got an error [15:14:55 CET] <BtbN> You map the audio from input 3 [15:15:06 CET] <BtbN> shortest is not a filter [15:15:46 CET] <plumbe0> input 3 or input 2? [15:17:02 CET] <BtbN> The one that has your audio [15:17:11 CET] <plumbe0> i edited the map to be -map [out][2:a] but it is still silent [15:17:17 CET] <plumbe0> i edited the map to be -map "[out][2:a]" but it is still silent [15:17:29 CET] <plumbe0> also tried with 3 [15:19:08 CET] <BtbN> You map one source at a time. [15:19:48 CET] <plumbe0> I'm not that expert using ffmpeg, sorry. Can you be a bit more explicit? [15:20:16 CET] <BtbN> Not sure how to be more explicit than that. One source per -map. [15:21:29 CET] <plumbe0> So where do i tell ffmpeg to also pick up the audio and limit the video length to match the length of the audio? [15:21:48 CET] <pomaranc> plumbe0: you add another -map followed with the audio encoding params [15:21:57 CET] <pomaranc> before the output file params [15:23:13 CET] <plumbe0> ffmpeg -f lavfi -i color=c=0xc03a08:s=1920x1080 -i .\a.png -i '.\audio.flac' -filter_complex "[1:v]scale=1000:1000[v1];[0:v][v1]overlay=(W-w)/2:(H-h)/2[out][2:a]" -map "[out]" -map "[2:a]" output.mp4 -> No output pad can be associated to link label '2:a'. [15:24:11 CET] <pomaranc> remove that [2:a] from the complex filter [15:24:14 CET] <pomaranc> it makes no sense there [15:31:21 CET] <plumbe0> removed -> "Output with label '2:a' does not exist in any defined filter graph, or was already used elsewhere." [15:40:11 CET] <vlt> pomaranc: And don't map it. [15:41:44 CET] <plumbe0> so how do i tell ffmpeg to use the audio? with or without the map doesn't seem to work and i'm not expert enough to figure it out myself what i'm doing wrong [15:42:53 CET] <vlt> plumbe0: Ah, sorry, I didn't see the "-i" that corresponds to "[2:a]". [15:48:40 CET] <pomaranc> plumbe0: 2:a, not [2:a] [15:48:57 CET] <vlt> plumbe0: Ot just 2 [15:49:32 CET] <pomaranc> or 2:a:0 to be cool [15:52:52 CET] <plumbe0> thanks that worked! now i only need to tell ffmpeg to match video length to audio length. i tried adding -shortest option but it adds 2 seconds of silence at the end [18:18:36 CET] <plumbe0> Hello, I'm back with another call for help on options order: my command is "ffmpeg -i color=c=0xc03a08:s=1920x1080 -i ./a.png -i ./audio.wav -filter_complex "[1:v]scale=1000:1000[v1];[0:v][v1]overlay=x=(W-w)/2:y=(H-h)/2:shortest=1[out]" -map "[out]" -map "2:a" -c:v libx264 -preset slow -profile:v high -crf 18 -coder 1 -pix_fmt yuv420p -movflags [18:18:36 CET] <plumbe0> +faststart -g 30 -bf 2 -c:a aac -b:a 384k -profile:a aac_low output.mp4" and when i run it i get "Trailing options were found on the commandline. [...] Filter overlay has an unconnected output". From what i read on the internet it should just be a matter of reordering the options, but i'm at a loss. Can you help me spot the offending options? [19:26:18 CET] <vlt> plumbe0: It might just be the missing "-f lavfi" [20:38:52 CET] <saber1> hello! trying my luck here again. I have an mp4 video in Variable frame rate, and I'd like add the PTS to each frames, the problem I have right now is multiple frames in the output video have the same timestamp [21:32:03 CET] <ddubya> saber1, by "add PTS" do you mean as a text overlay? [21:32:34 CET] <saber1> ddubya: yeah [21:33:08 CET] <ddubya> if you are transcoding, may as well make it cfr then [21:34:02 CET] <saber1> ddubya: my knowledge is very limited and I don't know whether I am transcoding or not...can't I keep it same as VFR? [21:34:39 CET] <ddubya> I'm not sure about that... cfr is more compatible [21:34:57 CET] <ddubya> if you change the frame rate in the filter chain you should be good [21:35:12 CET] <ddubya> could be before or after text overlay [21:36:27 CET] <saber1> ddubya: do you mean I do a `-r 30` in the text overlay command? [21:37:52 CET] <ddubya> you could do for example -vf framerate=30,text=... [21:38:30 CET] <kepstin> I don't normally recommend the framerate filter, it does frame blending and produced bad/unnatural result on many types of video [21:39:00 CET] <kepstin> what makes you think you have an issue with "multiple frames in the output video have the same timestamp"? [21:40:56 CET] <Palkesonoh> Hello im using this command to remove the black bars and make the video fullscreen , ffmpeg -i 1.mkv -vf "crop=1920:800:0:140,scale=1920:1080,setsar=1" og1.mkv It works good , But the problem is the video is a stretched because of the new crop settings. [21:41:05 CET] <kepstin> Palkesonoh: please don't pm me for no reason [21:41:26 CET] <Palkesonoh> i need help [21:41:28 CET] <saber1> ddubya kepstin : thanks. So I can see it by playing the video in the video player, as frame changes but the timestamp of the video doesn't change. [21:41:55 CET] <kepstin> sefeng211: what video player with what options? are you using a deinterlacing filter? [21:42:17 CET] <Palkesonoh> how i can fix it ? [21:43:23 CET] <kepstin> Palkesonoh: you can't remove bars to make a video fullscreen without stretching it, that doesn't make sense [21:43:51 CET] Action: saber1 sent a long message: < https://matrix.org/_matrix/media/r0/download/matrix.org/MjfnLbswYvLcaSrrFXu… > [21:44:17 CET] <kepstin> sefeng211: please don't paste long messages in the channel, use a pastebin site [21:44:34 CET] <saber1> sorry! [21:44:36 CET] <Palkesonoh> kepstin when scaling, how to keep shapes of people's heads , thats what i mean is like is stretched because of the height when u remove the black bars [21:44:37 CET] <kepstin> sefeng211: please try running "ffprobe -show_frames" on the file, and pastebin the first 10 or so frames of output. [21:45:12 CET] <kepstin> Palkesonoh: what you're asking makes no sense. you can't stretch a video without stretching the video. [21:45:51 CET] <Palkesonoh> So the issue/problem that I'm trying to solve is how to prevent any stretching in either direction. In other words, a round soccer ball in the input file must yield a round ball in the output file ! (NOT oval-shaped in either axis). [21:47:18 CET] <saber1> kepstin: https://termbin.com/yqix [21:47:24 CET] <ddubya> saber1, what is your command line? Do other variables increment like frame_num etc [21:48:08 CET] <kepstin> Palkesonoh: remove the ",scale=1920:1080,setsar=1" from your filter [21:48:09 CET] <saber1> ddubya: this was the command ffmpeg -nostdin -i tmp.mp4 -c:v libx264 -profile:v baseline -vf "drawtext=x=(w-(max_glyph_w*13))/2: y=H-h/10:fontcolor=white:fontsize=h/14:box=1:boxcolor=0x000000AA:text='%{pts\\:hms}'" -y -vsync 0 tmp_timestamp.mp4 [21:48:32 CET] <kepstin> sefeng211: there are no repeated pts, it's just a rounding error in the player output log [21:48:44 CET] <kepstin> sefeng211: there are however some really close pts values [21:50:33 CET] <saber1> kepstin: er, so show_frames gave different result than show_packets, and I used show_packets previously [21:50:59 CET] <kepstin> oh, hmm, show_frames might have some fixups by the decoder [21:51:12 CET] <kepstin> they are the same values with show_packets? that's annoying [21:51:25 CET] <ddubya> saber1, works for me, ... maybe "-fflags genpts" ? [21:52:03 CET] <kepstin> sefeng211: so anyways, you have this file with this issue, what do you want to do with it? [21:52:04 CET] <saber1> could it be I have some negative frames in the input video? [21:52:39 CET] <kepstin> or is the file something that you're generating, and you need to fix it? [21:53:26 CET] <saber1> kepstin: all I want to do is to get timestamps overlay for doing some analysis, because I need to know what the frame looks at at each timestamp [21:54:06 CET] <kepstin> hmm. the filterchain only gets the "fixed up" timestamps, like the -show_frames option to ffprobe [21:54:39 CET] <saber1> and since this issue, the generated video isn't very trustworthy [21:54:58 CET] <kepstin> there might be some things you can do with e.g. the -copyts option to disable parts of ffmpeg's timestamp processing [21:55:38 CET] <kepstin> also -vsync 0 [21:58:06 CET] <saber1> hmm, okay, let me check with -copyts, and don't I already using -vsync 0? or I should use it somewhere else [21:58:51 CET] <Palkesonoh> kepstin current code ffmpeg -i 1.mkv -vf "crop=1920:800:0:140" -c:a copy out.mp4 , when i remove setsar and scale nothing happens (still with black bars) [21:59:35 CET] <kepstin> the video no longer has the black bars encoded in it [21:59:46 CET] <kepstin> but it doesn't match your screen shape [21:59:53 CET] <kepstin> so there's space left over [22:01:29 CET] <kepstin> Palkesonoh: if you want the video to fill the screen, you have two options: either stretch it, or chop off the sides [22:01:46 CET] <kepstin> stretching makes it look stretched, chopping off the sides removes part of the video [22:04:13 CET] <Palkesonoh> So because there are movies that are full screen and look good [22:04:32 CET] <saber1> copyts doesn't seem to work, and I have to use -vsync 0, otherwise some frames will be missing :( [22:08:07 CET] <plumbe0> Hello, I'm back with another call for help on options order: my command is "ffmpeg -i color=c=0xc03a08:s=1920x1080 -i ./a.png -i ./audio.wav -filter_complex "[1:v]scale=1000:1000[v1];[0:v][v1]overlay=x=(W-w)/2:y=(H-h)/2:shortest=1[out]" -map "[out]" -map "2:a" -c:v libx264 -preset slow -profile:v high -crf 18 -coder 1 -pix_fmt yuv420p -movflags [22:08:07 CET] <plumbe0> +faststart -g 30 -bf 2 -c:a aac -b:a 384k -profile:a aac_low output.mp4" and when i run it i get "Trailing options were found on the commandline. [...] Filter overlay has an unconnected output". From what i read on the internet it should just be a matter of reordering the options, but i'm at a loss. Can you help me spot the offending options? [22:08:53 CET] <kepstin> Palkesonoh: movies that fill your screen were either filmed in 16:9 ratio, or were adjusted to that ratio before being sent to you. [22:09:24 CET] <kepstin> plumbe0: your error message does not match the command line you pasted. [22:11:22 CET] <kepstin> plumbe0: also, you left before anyone responded last time, vlt helpfully mentioned that you need "-f lavfi" before the input with the color filter in it [22:15:40 CET] <Palkesonoh> kepstin then is not possible remove top & bottom black bars without look stretch? [22:15:59 CET] <kepstin> Palkesonoh: the other option is to remove some video from each side [22:16:46 CET] <Palkesonoh> Yes, but it won't look so good, I'll just download the open matte version, thanks anyways. [22:17:14 CET] <kepstin> i'm not sure what you're hoping for, it's not like ffmpeg can "make up" something to fill the missing space [22:17:56 CET] <plumbe0> Looking into these comments, thanks [22:18:21 CET] <Palkesonoh> kepstin https://i.ytimg.com/vi/4q-kFrnXHL0/maxresdefault.jpg [22:18:30 CET] <Palkesonoh> Look the image , i wanted like that but you said is not possible [22:18:35 CET] <Palkesonoh> So i will try to find the open matte version [22:18:44 CET] <kepstin> no, they did the thing i said you could do [22:18:48 CET] <kepstin> they removed some video from the sides [22:19:08 CET] <kepstin> notice how the person's head is closer to the left, and there's less black space on the right [22:19:18 CET] <kepstin> you can do that with the crop filter in ffmpeg [22:19:40 CET] <Palkesonoh> true [22:19:54 CET] <Palkesonoh> ffmpeg -i 1.mkv -vf "crop=1920:800:0:140" -c:a copy out.mkv [22:19:59 CET] <Palkesonoh> so what i have to change here? [22:20:35 CET] <kepstin> change the width so that it's a ratio of 16:9 with the height (and also a multiple of 2) and adjust the x value to center it. [22:21:12 CET] <kepstin> if you do that, you can also add the ",scale=1920:1080,setsar=1" back to the end if you need a 1080p video [22:21:39 CET] <Palkesonoh> okey i will try and let u know thanks you [22:25:04 CET] <kepstin> I'd use crop=1422:800:250:140 but note that those values all have to be rounded a bit, and you can shift it left or right if you prefer. [22:26:27 CET] <Palkesonoh> i will try [22:28:27 CET] <Palkesonoh> kepstin [22:29:48 CET] <Palkesonoh> why is that [22:30:06 CET] <kepstin> why is what? [22:32:01 CET] <Palkesonoh> kepstin [22:32:40 CET] <kepstin> ... do you have a question? [22:35:02 CET] <plumbe0> thanks everybody, i got it working [23:01:27 CET] <saltarob> I got following error while compiling ffmpeg-4.2.2 on macOS 10.14.6 with the Haivision-SRT 1.4.1 package:ERROR: srt >= 1.3.0 not found using pkg-configFirst I tried it with the srt package from Homebrew and then I compiled it from source. But both times I got the same error. I compiled the same ffmpeg 4.2.2 package on Ubuntu with SRT 1.4.1 without [23:01:28 CET] <saltarob> problems.From the ffbuild/config.log:require_pkg_config libsrt srt >= 1.3.0 srt/srt.h srt_socket check_pkg_config libsrt srt >= 1.3.0 srt/srt.h srt_socket test_pkg_config libsrt srt >= 1.3.0 srt/srt.h srt_socket false --exists --print-errors srt >= 1.3.0 ERROR: srt >= 1.3.0 not found using pkg-configWhy is it asking for such an old libsrt version? [23:01:28 CET] <saltarob> From where does it get this requirement?Thank you. [23:10:23 CET] <void09> i found a video where if i select 2 consecutive scenes with -ss -to, I get a duplicated frame (tried with both ss/to before and after -i). anyone here with some knowledge who can take a look at what happens ? [23:17:15 CET] <vlt> void09: Can you show the command you used? [23:20:27 CET] <void09> vlt: let me check if it happens with local file too.. i used it over http [23:20:31 CET] <void09> i mean http input [23:24:03 CET] <void09> ./ffmpeg -ss 00:02:16.845 -to 00:02:23.643 -i file.mkv -crf 0 1.mkv [23:25:15 CET] <void09> sorry that's: [23:25:41 CET] <void09> ./ffmpeg -ss 00:01:59.619 -to 00:02:16.845 -i file.mkv -crf 0 1.mkv [23:26:11 CET] <void09> ./ffmpeg -ss 00:02:16.845 -to 00:02:23.643 -i file.mkv -crf 0 2.mkv [23:26:30 CET] <void09> vlt: last frame of 1.mkv is the first frame of 2.mkv [23:26:37 CET] <void09> i checked the source and there is no duplicate frame there [23:27:39 CET] <vlt> void09: You tell it to do exactly that. [23:27:52 CET] <void09> vlt: nope, that's not how it works [23:28:11 CET] Action: vlt doublechecks [23:28:14 CET] <void09> not sure exactly how it works, but i think the end timestamp is exclusive, not inclusive [23:28:25 CET] <void09> [ss to) [23:29:02 CET] <void09> because I am using exactly the same pattern to select scenes from the input video (generated with pyscenedtect), and this happens in only 2 plaes i found so far [23:29:05 CET] <void09> places* [23:30:39 CET] <void09> could be because of some rounding errors i think. the input video is 23.976 fps [23:31:44 CET] <void09> sometimes the frame length is rounded to 41ms sometimes to 42ms (as it appears in mpv) [23:33:25 CET] <void09> for 25 fps video a frame is always 40ms, no rounding errors [00:00:00 CET] --- Sat Jan 11 2020

1 0

ffmpeg-devel.log.20200109
by burek 10 Jan '20

10 Jan '20

[00:00:00 CET] --- Thu Jan 9 2020

1 0

ffmpeg.log.20200109
by burek 10 Jan '20

10 Jan '20

[00:35:23 CET] <ddubya> ok it seems the threads are per-frame and execute/execute2 are not used at all [07:56:10 CET] <th3_v0ice> Can anyone shed some light on this issue? https://pastebin.com/P0Z6RQkB I am using API and I get the same message with scale_cuda. [08:29:09 CET] <th3_v0ice> Or how to disable autoscale filter from API? [09:17:11 CET] <yang> To (re)encode from Matroska to .avi (or a format visible on DVD players) do I just type "ffmpeg video.mkv video.avi" ? [09:21:48 CET] <yang> seems not to work [09:43:54 CET] <yang> I did it with "-i" not sure if it's optimal [10:02:05 CET] <yang> but its actually a not well re-encoded stream [10:02:09 CET] <yang> it has small cubes [10:12:14 CET] <th3_v0ice> yang: You can try something like this ffmpeg -i video.mkv -c:v libx264 -crf 23 -c:a copy video.avi [10:12:25 CET] <yang> thanks I will try [10:13:23 CET] <yang> what is crf 23 ? [10:13:25 CET] <th3_v0ice> adjust -crf value to one that suits your needs. Lower number is better but gives bigger file size. Min: 0 [best] -> Max: 51 [worst] [10:13:56 CET] <th3_v0ice> basically that number is controlling quality [10:15:49 CET] <yang> Does it matter if I "do work" on computer and lag the machine, does it has any effect on the re-encoding process ? [10:26:29 CET] <th3_v0ice> It will simply take more time if you "do work" :) [11:02:02 CET] <lofo> i'm trying to live-stream my Mac's screen over RTP. what am i doing wrong ? ffmpeg -re -f avfoundation -i "0" -c:v mpeg2video -q:v 20 -pix_fmt yuv420p -f rtp rtp://224.0.0.1:8081 [11:26:35 CET] <yang> th3_v0ice: with that command the bitrate goes to 14000kbit/s, while original video capture was around 3000kbit/s...and my load goes to 8 when re-encoding...hmmm [11:33:17 CET] <BtbN> DVD players don't typically play h264 [11:34:43 CET] <yang> is there a way to re-encode with "slow" options on a very slow computer aswell ? [11:34:55 CET] <yang> What would I need to apply to teh settings ? [11:35:20 CET] <BtbN> It would just be slow. [11:35:57 CET] <BtbN> You want mpeg2 for a DVD player. And mpeg2 is quite terrible. [11:38:18 CET] <yang> I mean is there a way to re-encode with load not being over 1 with some kind of a ffmpeg "slow" setting ? [11:44:02 CET] <pink_mist> yang: you don't want "slow" settings ... the "slow" ones all take *more* time. you want "fastest" or something [11:44:19 CET] <yang> more time is fine, if the machine is slow [11:44:39 CET] <pink_mist> ..... you just said .... uh, what [11:44:47 CET] <BtbN> It will always use all available resources. And just finish faster or slower. [11:44:58 CET] <yang> I mean takng more time to process encoding is fine, if I have a slow machine/CPU [11:44:59 CET] <BtbN> You can limit the amount of threads it will use. [11:59:10 CET] <yang> BtbN: Yes, it's using 7 threads seen in "htop"... [11:59:20 CET] <yang> How can I limit that to 1-2 threads ? [12:02:31 CET] <yang> ok it seems it's "threads 1" [12:02:50 CET] <yang> but I still see 4 threads being in use [12:06:23 CET] <yang> OK I see load going down [13:31:10 CET] <th3_v0ice> Are there any examples of hardware transcoding using API? [16:17:10 CET] <ddubya> th3_v0ice: i don't know of any examples. maybe do the equivalent at the command line using "-loglevel trace" will give clues how the filter chain is built up [17:17:35 CET] <th3_v0ice> ddubya: Thanks, I will try that. [17:38:14 CET] <KodiakIT> Not strictly an ffmpeg question, got no response asking ##linux and #archlinux and thought you guys might know. If VCE is properly enabled for an AMD card it'd show up if I were to run 'lsmod | grep vce" right? [17:41:31 CET] <kepstin[m]> what driver? with the regular open-source drivers i don't think there's a special log, but you can use the va-api tools (e.g. vainfo) to check if it's set up right. [17:44:00 CET] <KodiakIT> Yeah, the FOSS one; xf86-video-amdgpu. I'll check with vainfo [17:44:28 CET] <KodiakIT> Or rather I will, as soon as I find which package provides it. [17:45:13 CET] <kepstin[m]> usually has 'libva' in the name, some distros call it 'libva-utils' [17:46:10 CET] <KodiakIT> Was the latter. currently getting " [17:46:18 CET] <KodiakIT> "vaInitialize failed with error code -1 (unknown libva error),exit" when I try to run it though [17:47:30 CET] <kepstin[m]> hmm. the video decoding/encoding driver is provided by mesa, iirc. [17:47:49 CET] <KodiakIT> Yep. I checked that mesa-vdpau is there, and pacman shows it is [17:48:12 CET] <kepstin[m]> vdpau is something different, you want vaapi [17:48:46 CET] <kepstin[m]> you're on arch? https://wiki.archlinux.org/index.php/Hardware_video_acceleration#ATI/AMD is a helpful page for you :) [17:49:00 CET] <KodiakIT> Already had it open lol [17:50:02 CET] <KodiakIT> I'm not exactly well versed on the peculiarities and differences between VA-API and VDPAU though, I was under the impression that the latter was AMD's alternative to Intel's VAAPI [17:50:10 CET] <kepstin[m]> hopefully it's as simple as installing libva-mesa-driver to get it to work. [17:51:03 CET] <KodiakIT> Fingers crossed! vainfo shows proper output now at least. [17:51:40 CET] <kepstin[m]> right then, should be good. https://trac.ffmpeg.org/wiki/Hardware/VAAPI is the docs for how to use ffmpeg with vaapi, btw :) [17:51:43 CET] <KodiakIT> Now to see if handbrake uses it or if I'll have to switch to Windows to get it to play nice. [17:52:10 CET] <kepstin[m]> i dunno about handbrake specifically. [17:52:18 CET] <KodiakIT> (I know it partly uses ffmpeg on the back end, just dunno about config wise if it's set up) [17:52:45 CET] <kepstin[m]> (note also that amd's hardware video encoder, especially on pre-vega cards, isn't particularly good - if you're not doing e.g. realtime streaming, a software encoder is usually a better option) [17:53:06 CET] <KodiakIT> Nah, digitizing some old optical media [17:53:19 CET] <KodiakIT> Or rather attempting to. [17:53:39 CET] <kepstin[m]> for that, i'd recommend using a software encoder. you'll get smaller files that look better. [17:53:59 CET] <KodiakIT> If x265 weren't taking ~2h per episode, I would [17:54:41 CET] <kepstin[m]> for <4k content, better to use x264 probably [17:55:12 CET] <KodiakIT> Eh, I'm more concerned with space above all [17:55:47 CET] <kepstin[m]> if you're mostly concerned about space, you should not use the hardware encoder [17:56:34 CET] <KodiakIT> I mean, I've done tests, the VCE-x265 is smaller than the software-x264 of the same episodes... [17:56:51 CET] <kepstin[m]> well, you can make anything smaller by setting the bitrate to whatever you like [17:57:35 CET] <kepstin[m]> how does the visual quality compare? [17:58:21 CET] <KodiakIT> I'm using RF18 as it stands right now, might re-encode later on with an ABR preset instead [17:58:44 CET] <bencoh> I kinda doubt a hardware encoder would beat x264 as well, but ... [17:58:57 CET] <kepstin[m]> note that the hardware encoder doesn't have a constant-quality mode like x264's crf [17:59:14 CET] <kepstin[m]> (it has a constant-qp mode, iirc, but that's not the same thing) [17:59:50 CET] <KodiakIT> Ah, in handbrake it's still refered to as RF not CQ like it is on VP8/9 [18:00:39 CET] <KodiakIT> Though AFAICT, this is still using the software encoder seeing as it's still got ~1h40m left and it's only ~7% done. [18:01:56 CET] <kepstin[m]> i haven't personally done much with amd's video encoder, but in general i think the hardware hevc encoders have tended to get close to software h264 encoders in terms of efficiency? [18:04:23 CET] <KodiakIT> Ah screw it, I'll just up the RF and do x264 as suggested, and then maybe re-run them with 10-bit x265 to squish em down further later. [18:05:13 CET] <kepstin[m]> i'd recommend avoiding doing multi generations of lossy encodes :/ [18:06:39 CET] <KodiakIT> Well I don't have anywhere near that kinda space so [18:07:17 CET] <KodiakIT> I literally just this past year finally got more than 1TB inside my tower (because my buddy gave me his old 2TB) [18:08:08 CET] <KodiakIT> Hell I'm not sure I've got 4TB total between every device I have in the whole house [18:09:54 CET] <kepstin[m]> well, your best option, assuming you don't want to keep the original files, is probably to encode each file from the original directly to the best available software encoding method at the quality/size you want. [18:09:56 CET] <kepstin[m]> It'll be slow, but each time a video finishes you just delete the original then go on and do the next. [18:12:44 CET] <kepstin[m]> (setting the quality/size tradeoff appropriately is the tricky bit; if you're using x264 or x265 you should start by doing some short sample encodes with different crf to see how small you can go before quality is an issue) [18:13:49 CET] <KodiakIT> The time/space trade off isn't worth it for me. I need to free up the space from the DVD raws, if that means that my end-result RF18-x265 is slighly lower quality since it was converted from an RF15-x264 instead of the bloated MPEG-2, so be it [21:49:03 CET] <matto312> Hello! I'm wondering if someone can point me in the right direction to debug a flood of "DTS 4294969402, next:2147485753967 st:1 invalid dropping [21:49:03 CET] <matto312> PTS 4294969402, next:2147485753967 invalid dropping st:1" messages [21:50:13 CET] <matto312> Input is live RTMP & output is multi-variant HLS ... I just can't figure out why sometimes the screen gets flooded with the "invalid dropping" messages ... any help that could push me in the right direction would be greatly appreciated [21:50:32 CET] <matto312> I'm not sure where to begin to debug this [21:57:45 CET] <matto312> anything thoughts at all on what may be causing this, I can't find anything on google [21:58:03 CET] <matto312> https://pastebin.com/ZC27A6vY [22:02:21 CET] <AppAraat> hi, I want to record my screen losslessly using ffmpeg but I can't seem to find the right command. Here it says "Lossless recording" but when I view the video the colors look smudged: https://www.bogotobogo.com/FFMpeg/ffmpeg_video_screencasting_screen_recordi… [22:04:29 CET] <kepstin> AppAraat: if you want to do actual lossless screen recording, you need to leave it in rgb [22:04:57 CET] <AppAraat> so "-pix_fmt rgb" ? [22:05:07 CET] <kepstin> remove the -pix_fmt option, and switch the encoder to -c:v libx264rgb (or another encoder that supports lossless rgb, like ffv1) [22:06:26 CET] <kepstin> the "smudged" colours are caused by the conversion to yuv420p, which uses subsampled (quarter-resolution) colour data. This is fine for normal video, but causes issues with sharp edges in computer graphics. [22:06:58 CET] <AppAraat> yeah I mainly want to record terminal text and stuff [22:07:37 CET] <AppAraat> so now I have: ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s 1366x768 -i :0.0 -c:v libx264rgb -vcodec libx264 -preset ultrafast -crf 0 -threads 0 -acodec pcm_s16le -y output.mkv [22:07:44 CET] <AppAraat> does that look all right? [22:07:55 CET] <kepstin> no, you didn't do what i told you to do [22:08:03 CET] <kepstin> er, wait, [22:08:08 CET] <kepstin> you have redundant video encoders [22:08:13 CET] <AppAraat> sorry, I'm very noob with ffmpeg [22:08:14 CET] <kepstin> -c:v and -vcodec are the same option [22:08:17 CET] <AppAraat> oh [22:08:20 CET] <kepstin> and the second is overwriting the first [22:09:21 CET] <AppAraat> so like this then? ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s 1366x768 -i :0.0 -vcodec libx264rgb -preset ultrafast -crf 0 -threads 0 -acodec pcm_s16le -y output.mkv [22:09:30 CET] <kepstin> yeah, that should be fine. [22:10:25 CET] <kepstin> note that the "-threads 0" option is redundant, that's the default [22:10:35 CET] <kepstin> (0 means "auto") [22:10:49 CET] <AppAraat> oh ok, I'll remove that then, thanks. [22:11:49 CET] <AppAraat> seems to work, thanks :) [22:19:26 CET] <matto312> Anything at all that could help me figure out what is causing the DTS PTS flood of "invalid dropping st:1" ... I'll even take guesses, just trying to figure out where to look [22:49:27 CET] <feettrip> Hi [00:00:00 CET] --- Fri Jan 10 2020

1 0

ffmpeg-devel.log.20200108
by burek 09 Jan '20

09 Jan '20

[00:00:00 CET] --- Wed Jan 8 2020

1 0

ffmpeg.log.20200108
by burek 09 Jan '20

09 Jan '20

[00:22:16 CET] <kepstin> cpplearner: do you mean, you want a list of all the possible keys ffprobe can output in the [FORMAT] section, regardless of file type? [04:06:24 CET] <unixabg> lastlog unixabg [04:06:36 CET] <unixabg> oops sorry [04:08:48 CET] <unixabg> kepstin: first ty for the response. Is there an example page or is that information in the docs somewhere? [04:16:25 CET] <kepstin[m]> for the zmq filter commands stuff? There's docs with examples at https://ffmpeg.org/ffmpeg-filters.html#zmq_002c-azmq [09:01:52 CET] <Beam_Inn> is it possible to make animated gifs with ffmpeg from image files? [09:02:43 CET] <Beam_Inn> Iwanted to follow up on something I read the other day. Basically, someone was talking about videos using the same image over multiple frames rather than multiple instances of the same photo over multiple frames. Is that really possible? [09:02:57 CET] <Beam_Inn> I don't know anything about how videos are coded. [10:06:41 CET] <Lypheo> Beam_Inn: yes. something like ffmpeg -f image2 -framerate 1 -i image%d.jpg animated.gif should do it [10:07:33 CET] <Lypheo> also what you read might have been referring to interframe compression which is indeed possible [10:13:17 CET] <lofo> I'd like to stream video from a device to another on my local network. I want my receiver to be able to connect/disconnect multiple times without having the emitter to re-create a stream every time. is that possible ? [14:54:00 CET] <AlexanderBrock> identify AlexanderBrock Nsc6mwUoWoHuQMzPrp94UjKGXRh7iFUxOB11CRlKWRpOcYMk4FZJ7SQoUwYlI2my [14:56:19 CET] <AlexanderBrock> I want to compile FFmpeg on Debian testing with mp3 encoder support. It complains that libmp3lame is too old or missing ("ERROR: libmp3lame >= 3.98.3 not found"). [14:56:21 CET] <DHE> good password but I think you need a new one [14:56:30 CET] <AlexanderBrock> DHE: already changed it :-) [14:56:53 CET] <BtbN> Well, is libmp3lame too old? [14:57:17 CET] <DHE> or is often the case, you need some kind of libmp3lame-dev package installed as well [14:57:26 CET] <AlexanderBrock> "apt-cache madison libmp3lame0" shows "3.100-3" [14:58:21 CET] <AlexanderBrock> ah, -dev package was missing [15:42:19 CET] <tMH> hello ffmpeg gurus, I want to ask a question. I'm user of videoredo software that has very nice feature 'quickstream fix', what this feature does I did paste here: https://justpaste.it/77nef [15:43:19 CET] <tMH> the question is - how to do the same with ffmpeg codec ? I really need to make this quickstream fix on many files while VideoRedo software is not easy to use in batch mode... that is why I came to this channel to ask you about ffmpeg ability to do the same as videoredo does. [15:43:28 CET] <tMH> ffmpeg program I mean [15:45:07 CET] <tMH> videoredo itself tells me that it uses ffmpeg libraries for video operations... [15:46:34 CET] <DHE> sounds like it's just a remxer. ffmpeg can do that [15:46:42 CET] <tMH> dhe: how to ?:) [15:47:10 CET] <DHE> ffmpeg -i INPUTFILE -c copy OUTPUTFILE [15:47:16 CET] <tMH> just it ? [15:47:19 CET] <tMH> hm. [15:47:32 CET] <DHE> maybe. having not used this software I can only go by what this brief text blub says [15:47:36 CET] <DHE> blurb [15:48:02 CET] <tMH> and ffmpeg cannot operate files in batch mode, I suppose.. [15:48:10 CET] <DHE> that's what your shell is for [15:48:37 CET] <tMH> btw [15:48:53 CET] <tMH> dhe: no difference encountered between INPUTFILE and OUTPUTFILE [15:49:10 CET] <DHE> ... good? [15:49:15 CET] <tMH> no work proceeded. [15:49:19 CET] <BtbN> It cannot do it in place. [15:49:21 CET] <tMH> nothing has changed. [15:49:24 CET] <BtbN> They need to be distinct files. [15:49:38 CET] <tMH> so what about real remuxing with ffmpeg ? [15:49:44 CET] <BtbN> ? [15:49:56 CET] <tMH> btbn: https://justpaste.it/77nef [15:50:03 CET] <tMH> I need to do the same with ffmpeg. [15:50:18 CET] <BtbN> Yes, and that command is how you remux. [15:50:22 CET] <tMH> doing ffmpeg -i input -c copy output gives no difference between two files [15:50:32 CET] <BtbN> Like I said, they need to be distinct. [15:50:36 CET] <BtbN> It cannot modify a file in-place. [15:50:57 CET] <tMH> don't get what you mean, sorry. [15:51:22 CET] <DHE> you are providing a file with a bad initial timestamp I assume? [15:52:21 CET] <tMH> dhe: no, youtube downloaded video. I'm trying to play it on some oldie media player and it stops video while audio is continue playing. after I do quickstream fix in videoredo - mediaplayer plays that video ok [15:52:24 CET] <tMH> video and audio. [15:53:15 CET] <tMH> e65fb32da7af19e6e3e283877e312e31 *ANCIENT BARDS-2018-Impious Dystopia (Official Video 4K).fixed.mp4 [15:53:15 CET] <tMH> e65fb32da7af19e6e3e283877e312e31 *ANCIENT BARDS-2018-Impious Dystopia (Official Video 4K).mp4 [15:53:16 CET] <DHE> how long does it take to fix it in videredo? [15:53:18 CET] <tMH> no difference. [15:53:25 CET] <tMH> dhe: pretty fast, no recoding at all [15:53:36 CET] <tMH> it does 'remux' as I understand. [15:54:16 CET] <th3_v0ice> How can I init hardware scaler? I am currently getting this error https://pastebin.com/XHsrNLAk [15:55:20 CET] <BtbN> Use scale_cuda instead of scale. And build ffmpeg with support for it, of course. [15:58:43 CET] <th3_v0ice> BtbN: Let me test if that will work. [16:01:43 CET] <lofo> I'd like to stream video from a device to another on my local network. I want my receiver to be able to connect/disconnect multiple times without having the emitter to re-create a stream every time. is that possible ? [16:08:28 CET] <th3_v0ice> BtbN: This is what I get now https://pastebin.com/CgAsfwyB [16:09:58 CET] <DHE> lofo: a live/realtime stream? [16:10:37 CET] <lofo> yup [16:15:14 CET] <DHE> what's the device? have you considered multicast? [16:15:41 CET] <lofo> a smartphone [16:16:00 CET] <lofo> what do you mean by multicast ? [16:18:00 CET] <DHE> UDP to IP addresses from 224.0.0.0. through 239.255.255.255 are multicast. users on the local lan can "join" a stream and watch a live feed, and "leave" at any time and the sender is largely unaware [16:18:18 CET] <DHE> this is how I watch over-the-air TV with this little coax-to-ethernet box I have [16:21:04 CET] <lofo> UDP wasn't performing well and delay is an acceptable thing on my setup. Would that work with TCP as well ? [16:21:27 CET] <DHE> no. the nature of multicast doesn't work with TCP [16:22:05 CET] <DHE> so if you want something TCP based, this isn't really something ffmpeg handles directly. you probably want to get something like nginx-rtmp involved, or HLS or DASH with a static HTTP content server (apache is fine) [18:12:49 CET] <zerodefect> I'm trying to build ffmpeg in Ubuntu 18.04.3 with --enable-gnutls but when I do ./configure, I get error 'ERROR: gnutls not found using pkg-config' [18:13:02 CET] <zerodefect> Now I've done an install of gnutls-dev but to no avail [18:13:19 CET] <zerodefect> I've also tried setting PKG_CONFIG_PATH to the location of .pc file [18:15:42 CET] <DHE> I'd check the config.log (possibly in the ffbuild directory) to see what went wrong. could be a problem with gnutls itself, version mismatch, etc... [18:20:21 CET] <zerodefect> Brilliant. Thanks. I wasn't aware of that file. It tells me that I was missing libunistring.so. Installed and good to go. [20:43:52 CET] <garoto> does anyone knows why these configure options fail to generate a static ffmpeg binary? -> http://sprunge.us/L0VUkb [20:45:09 CET] <garoto> don't mind the extra spaces, it was inserted by my console app) [20:56:12 CET] <DHE> --enable-static is for the libraries, not the binary itself. if you want a static link of ffmpeg, ffprobe you need --extra-ldflags=-static [20:56:50 CET] <DHE> though you may find that hard to do on many distros... [20:58:15 CET] <garoto> ah ok, gonna give it a try with --extra-ldflags=-static [21:15:44 CET] <garoto> yep, doesn't work. `ERROR: libass not found using pkg-config` [21:15:48 CET] <Stryker> Hi, I need help using FFMPEG. I want to create a clip from a video at a smaller resolution, then append the same clip but at half the playback speed to it. For some reason I can't get the latter to work, ie. I only get the clip but without the replay. https://pastebin.com/HT3kMrYk Can anyone tell me what I'm doing wrong? Thanks in advance! [21:16:36 CET] <garoto> and once you remove that configure option, another `pkg-config` error pops-out [22:53:42 CET] <ManDay> How do I concatenate a large set of ts files into one? Specifying them all by a sequence of "-i ..."s fails with an error about "too many open files" [22:54:21 CET] <pink_mist> make a script that does a few at a time [22:54:28 CET] <ManDay> I tried putting their names into an m3u file, but "-i list.m3u" gives "Invalid data" as an error [22:54:41 CET] <ManDay> pink_mist: thanks, but how awful [22:55:22 CET] <nicolas17> ManDay: using multiple -i isn't the way to concatenate anyway [22:55:39 CET] <ManDay> nicolas17: What would be correct instead? [22:56:17 CET] <nicolas17> -i video1.ts -i video2.ts would create a file with multiple video streams (if the output even supports that), not concatenate them in time [22:56:24 CET] <ManDay> I mean the synopsis in ffmpeg(1) seems to kind of suggest -i ... times X is the way to go [22:57:22 CET] <ManDay> hm I see, there is a wiki entry [22:57:24 CET] <nicolas17> afaik .ts files can be concatenated byte-wise due to how the file format works, so you can use 'cat' :) [22:57:27 CET] <ManDay> thanks for pointing that out [22:57:38 CET] <nicolas17> or you can use the concat protocol as the wiki says [22:57:39 CET] <ManDay> nicolas17: heh, okay - didn't know that :D [22:58:05 CET] <nicolas17> "Certain files (MPEG-2 transport streams, possibly others) can be concatenated" [23:05:24 CET] <ManDay> fwiw, trying the "concat:" protocol, the error is "Filename too long" [23:06:38 CET] <BtbN> You can concat .ts files by just using cat. [23:06:56 CET] <BtbN> And maybe follow that with a remux via ffmpeg for some fixups [23:07:13 CET] <ManDay> yes, thanks. I just wanted to mention it [23:07:24 CET] <ManDay> using the concat demuxer also works [23:23:38 CET] <AiNA_TE> https://trac.ffmpeg.org/wiki/Concatenate [23:23:53 CET] <AiNA_TE> echo the fils to a txt file and then use that for concat [23:24:02 CET] <AiNA_TE> oh bleh, i was scrolled up [23:24:20 CET] <saber1> hey! I have a video which has negative frames, because I cut it from the original video. And then I used a command like this to add timestamps to the video `ffmpeg -nostdin -i after-remove-orange.mp4 -c:v libx264 -vf "drawtext=x=(w-(max_glyph_w*13))/2: y=H-h/10:fontcolor=white:fontsize=h/14:box=1:boxcolor=0x000000AA:text='%{pts\\:hms}'" -y ./after-remove-timestamp.mp4`, the mystery to me is the output video isn't the same as [23:24:20 CET] <saber1> the input video [23:25:02 CET] <saber1> Looks like a few frames got removed, and I don't know why. I wonder if it could be related to the negative frames [23:48:29 CET] <Soni> so uh, how do you even encode DSD? [23:48:46 CET] <Soni> this is for Sega 32X development btw [23:56:51 CET] <ddubya> I'm supplying my own execute() and execute2() in AVCodecContext, and it seems to work for some codecs, but h264 codec will not use it. It seems to use execute(): https://github.com/FFmpeg/FFmpeg/blob/95e5396919b13a00264466b5d766f80f1a4f7… [00:00:00 CET] --- Thu Jan 9 2020

1 0

ffmpeg-devel.log.20200107
by burek 08 Jan '20

08 Jan '20

[00:00:00 CET] --- Tue Jan 7 2020

1 0

ffmpeg.log.20200107
by burek 08 Jan '20

08 Jan '20

[01:47:19 CET] <dostoyevsky> Could I use ffmpeg to detect if an mp3 is empty? Meaning: mostly contains silence? [01:50:10 CET] <Diag> Just saw intel bragging about svt-av1. thats a good one [02:09:14 CET] <nicolas17> dostoyevsky: there's a silencedetect filter [02:29:26 CET] <Mellowlink> hello [02:30:41 CET] <Mellowlink> I'd like to ask. Normally when I encode video into another format I have to do just the 'ffmpeg -i INPUT OUPUT' [02:30:53 CET] <Mellowlink> I am currently trying to encode videos into ascii art [02:31:03 CET] <Mellowlink> with this command: ffmpeg -i input.webm -pix_fmt rgb24 -window_size 160x38 -f caca output.webm [02:31:46 CET] <Mellowlink> ffmpeg starts playing the video but when it ends the output is same like input so it doesn't get converted into ascii [02:36:06 CET] <furq> Mellowlink: caca isn't an encoder [02:38:05 CET] <Mellowlink> furq: so are there any other ways how to convert videos into ascii and same them into file please? [02:38:33 CET] <Mellowlink> it should be just one line of bash command right? or it's actually more difficult? [02:38:36 CET] <furq> no good way that i know of [02:38:51 CET] <furq> the libcaca support in ffmpeg just draws decoded frames to a caca window [02:38:55 CET] <furq> so you'd have to capture that window separately [02:39:07 CET] <furq> there's probably a less dumb way but you can't do it directly with ffmpeg [02:44:30 CET] <DHE> Well what format would you use? a text file with newlines at the end of rows and ^L as the frame boundary ? [02:45:36 CET] <Mellowlink> DHE: .webm format just with convered frames into ascii [02:46:37 CET] <Mellowlink> probably best way would be mpv it with caca and then capture the mpv window with ffmpeg tho [03:02:29 CET] <nicolas17> Mellowlink: for that you'd need text rendering, which would surely need an option for font to use, etc [03:02:37 CET] <nicolas17> it could be done but ffmpeg doesn't do it yet [03:05:46 CET] <Mellowlink> I figured it out already by capturing the caca window spawned by mpv in real time. [03:06:33 CET] <Mellowlink> it's very time and performance consuming, not an optimal solution but at least it works [05:58:04 CET] <ryzenda> `ffplay -filter:v:crop "crop=1920:1080:0:140" video.webm` How do I use the crop filter to play a video file? This command doesn't work [06:00:20 CET] <ryzenda> Ah, I figured it out! `ffplay -vf crop=1920:1080:0:140 video.webm` [06:37:15 CET] <grosso> can ffmpeg know the duration of a mpeg-ts file without seeking to the end? [06:38:12 CET] <nicolas17> I think ts doesn't have such metadata [06:38:43 CET] <nicolas17> it may be able to approximate it from bitrate in the first part of the file × file size [06:41:20 CET] <grosso> thank you nicolas [06:42:17 CET] <grosso> I'm wondering if mpeg-ts support any kind of metadata, and if ffmpeg do look at it to extract duration [06:45:06 CET] <nicolas17> transport streams are designed to be streamed over digital TV and stuff [06:45:17 CET] <nicolas17> afaik there isn't even a "file header" [06:46:17 CET] <nicolas17> you can chop them at any 188-byte-aligned point, which would change the duration [06:57:01 CET] <grosso> I'm pretty sure that you can put any information other than audio and video on a ts stream. The fact is, if ffmpeg actually look at something to get duration, just like it does with flv, for example, where it looks for a "on_metadata" packet, even if that packet can not exist in practice [06:59:36 CET] <grosso> I mean, you can have a flv stream without such packet and it's a valid flv file... so you can have a ts without any kind of metadata, but you can actually put metadata on a ts, but it seems that ffmpeg don't care about it [17:49:45 CET] <Arnob> Hi, when I convert input audio to a video output, representing the audio frequency spectrum using this command: ffmpeg -i 1.wav -filter_complex \ [17:49:47 CET] <Arnob> "[0:a]showspectrum=s=1280x720,format=yuv420p[v]" \ [17:49:49 CET] <Arnob> -map "[v]" -map 0:a output3.mp4 [17:50:30 CET] <Arnob> I obtain a video but the spectrum doesn't fill the whole video length. Am I wrong somewhere? Thank you. [17:52:48 CET] <Arnob> To better understand I uploaded a short video of a bird recording on my peertube account: https://peertube.co.uk/videos/watch/bc1dda45-cc29-40d8-9b17-b63f8e8c0e0e [17:58:24 CET] <relaxed> Arnob: You may need to use "-shortest" or a specific time "-t $time" [18:09:42 CET] <kepstin> Arnob: the way the showspectrum filter works, a larger image size means more samples used in the fft, so the speed going across the video is slower. You should probably generate a smaller video size. [18:11:53 CET] <kepstin> Arnob: in general, the video height should be a power of 2, and it corresponds to the fft size. The default is 512. [18:15:43 CET] <lofo> What streaming protocol should i use if i intend my receiver to connect multiple times over time [18:17:35 CET] <kepstin> one that you have a streamer server for (ffmpeg is not a multi-client streaming server) and that your client supports [18:18:10 CET] <kepstin> you'll need to be more specific if you want any specific recommendations :) [18:36:24 CET] <lofo> i just need to stream to a single device, on my local network [18:45:44 CET] <kepstin> what device/software is receiving? [18:47:04 CET] <lofo> a phone [18:47:55 CET] <lofo> i would like to start a stream of my computer screen from my desktop. and connect to it anytime with my phone [18:48:34 CET] <lofo> i know how to do that with tcp and the `listen` option. but whenever my phone disconnects i have to relaunch the stream from my desktop [18:49:41 CET] <lofo> i use libVLC running on a custom app to receive the stream [18:51:18 CET] <unixabg> Greetings, is it possible to send sighup or the like to change params of running ffmpeg process? [18:56:36 CET] <kepstin> unixabg: certain filter can take "commands" which can be injected e.g. through the zmq filter from an external process [18:56:38 CET] <DHE> there's some very limited support for stdin to take some commands [18:56:39 CET] <kepstin> but otherwise, no [18:56:49 CET] <kepstin> oh, yeah, and the stdin stuff [18:59:40 CET] <kepstin> the underlying libraries the ffmpeg tool is built on let you make something a lot more flexible or suited to your needs, of course. [19:10:09 CET] <Phoenix|> I'm getting the warning "Only '-vf scale=-1:720' read, ignoring remaining -vf options: Use ',' to separate filters", but I'm not using any other -vf filters [19:12:26 CET] <durandal_1707> Phoenix|: ignore it [19:24:33 CET] <Arnob> kepstin: well, thank you !! It's too difficult for me now, I'll have to read more the ffmpeg docs as I mainly use it to transcode video and the waveforms/spectrum generation is new to me...I managed to get sthg not too bad though, but not perfect. I just changed the video size. [19:25:21 CET] <Arnob> relaxed: Thank you, I will read more the ffmpef docs as weveform/spectrum generation is new to me... [19:26:07 CET] <relaxed> Arnob: you're welcome, but kepstin's advice was probably better :) [19:32:09 CET] <Arnob> relaxed: ok, thanks for the hint, I'll follow kepstin path then :-) ... [19:35:07 CET] <kepstin> lofo left, so i guess they didn't want to head my follow-up advice on their problem. [20:57:42 CET] <ladders> is it possible to write a filter chain to selectively deinterlace? for example, i have a star trek dvd where the live action is progressive but the cgi cut scenes are interlaced. [20:59:51 CET] <kepstin> many of the interlacing filters in ffmpeg optionally support deinterlacing only frames flagged as interlaced. [21:00:18 CET] <CoreX> i was gonna say try the bwdif filter [21:00:20 CET] <CoreX> https://ffmpeg.org/ffmpeg-filters.html#bwdif [21:01:32 CET] <kepstin> note that on an ntsc format dvd with mixed content, i'd recommend using the "repeatfields" filter to undo the detelecine done by the encoder, followed by a detelecine filter, and then a deinterlacing filter on any remaining interlaced frame [21:03:15 CET] <kepstin> (probably use fieldmatch as the detelecine filter, it's the easiest to use for this type of thing) [21:04:48 CET] <kepstin> fieldmatch docs have an example of how to use the yadif deinterlacing filter on remaining interlaced frames; it's similar for other deinterlacing filters that have a deint option including bwdif. [21:05:57 CET] <ladders> 95% of my input files are progressive, would it be detrimental to blindly always use that process for all input or should i special case it? [21:06:17 CET] <ladders> i am writing a backup script for my discs and have gotten it to mostly hands-free operation [21:06:44 CET] <furq> yeah deinterlacing will negatively affect picture quality [21:06:56 CET] <furq> also bear in mind all frames on dvds are flagged as interlaced [21:07:03 CET] <kepstin> hmm, if you have inputs that you know are fully progressive (most recent hollywood dvds are) then you might want to skip that, it's possible the detelecine will mismatch, or it'll detect non-interlaced frames as needing deinterlacing [21:07:12 CET] <furq> if you're not also doing an ivtc you'll want something like the idet filter to try and autodetect [21:07:24 CET] <ladders> currently i am using idet to count frames and applying yadif if it meets some arbitrary threshold. this is the first case of mixed progressive/interlaced input. [21:07:38 CET] <kepstin> fieldmatch filter sets the interlaced flag based on its own combing determination during the detelecine step, fwiw [21:08:02 CET] <kepstin> it's not perfect, but it's ok for an automated solution [21:08:16 CET] <furq> ladders: -vf idet,yadif=deint=interlaced will only deinterlace frames idet marks as interlaced [21:08:22 CET] <furq> autodetection is never perfect though [21:08:39 CET] <kepstin> note that any autodetection can be tricked by progressive frames with high-frequency vertical detail [21:08:44 CET] <furq> it's better to go through manually where possible [21:10:04 CET] <furq> if you have film content then you can pretty safely assume it's always telecined (ntsc) or progressive (pal) [21:10:26 CET] <furq> film meaning released in theatres, not shot on film [21:11:14 CET] <kepstin> star trek... voyager, for example, all the live action was done on film (24fps), usually encoded on the dvd as progressive, but cg effects were different. [21:11:18 CET] <furq> right [21:11:43 CET] <kepstin> i'd expect the cgi there to be 30fps progressive, but i haven't actually looked at the dvds for that closely :) [21:12:46 CET] <furq> my expert advice for that sort of content would be cross your fingers that there's a blu-ray and someone else has dealt with it for you [21:13:06 CET] <kepstin> in the really bad cases tho, they do an interlaced upscale for the blu-ray :) [21:13:29 CET] <kepstin> since blu-ray still doesn't support in-stream framerate switching :/ [21:13:37 CET] <kepstin> (or if it does, it's unreliable in players) [21:13:56 CET] <ladders> i have dvds only, i am just attempting to convert to x264 mp4 that doesn't look completely terrible [21:14:40 CET] <kepstin> ntsc dvds of tv shows are such a pain that it's often easier to see if someone else has already done the work :/ [21:25:52 CET] <ladders> ok, thank you everyone for the advice [21:46:33 CET] <cehoyos> Note that there is also soft-telecine which is actually progressive and ignored by FFmpeg [21:46:48 CET] <cehoyos> So you can ignore it as well, only hard-telecined needs de-telecine filters [22:13:25 CET] <kepstin> my recommendation was to use the 'repeatfields' filter to apply the soft-telecine, so it could be re-detelecined, but then I've been burnt by badly encoded dvds where the "progressive" encoded parts had field mismatches. [23:20:50 CET] <bray90820> So I am getting an error while trying to convert a WMA file to an mp4 [23:20:52 CET] <bray90820> https://pastebin.com/raw/R9xetjgk [23:21:06 CET] <bray90820> Anyone know what that is? [23:24:50 CET] <furq> bray90820: you can't mux wma into mp4 [23:24:58 CET] <furq> either use mkv or reencode the audio to aac [23:27:20 CET] <bray90820> How would I reencode it? [23:27:38 CET] <furq> remove -c:a copy [23:28:03 CET] <bray90820> Thanks [23:28:15 CET] <bray90820> I feel kinda dumb that it was that easy [23:35:23 CET] <BtbN> uhm, that does explicitly NOT reencode it [23:38:51 CET] <cpplearner> Guys, how can I list the predefined(=allowed) entries of FORMAT on ffprobe? [23:40:42 CET] <pink_mist> ffprobe -formats? [23:42:34 CET] <cpplearner> Oh, I mean for -show_formats, like -show_formats format=duration. [23:51:06 CET] <furq> show_format lists all the entries [23:51:22 CET] <furq> for that particular file, idk how you'd list every possible entry [00:00:00 CET] --- Wed Jan 8 2020

1 0

ffmpeg-devel.log.20200106
by burek 07 Jan '20

07 Jan '20

[00:00:02 CET] --- Mon Jan 6 2020

1 0

ffmpeg.log.20200106
by burek 07 Jan '20

07 Jan '20

[03:48:21 CET] <wondiws> hi furq [04:02:09 CET] <CounterPillow> hi furq [05:38:18 CET] <Roden> the -to parameter decides the length of the video, is there a way to decide the ending point instead? [05:54:02 CET] <furq> that is what -to does [05:54:10 CET] <furq> it needs to be on the same side as -i [06:04:29 CET] <Roden> Can you check my command for your off-side error? `> ffmpeg -ss 00:00:05 -i i.mp4 -to 00:02:02 -c copy o.mp4` [06:05:22 CET] <Roden> You mean it must come before -i if -ss is before -i and after -i if -ss is after -i? [06:06:39 CET] <furq> on the same side as -ss i mean [06:06:42 CET] <furq> of -i [06:07:31 CET] <Roden> `> ffmpeg.exe -ss 00:00:24 -to 00:01:10 -i .\1xmas.mp4 -c copy 0xmas.mp4` works. [06:10:24 CET] <Roden> I don't know if you feel like talking about ffmpeg stuff, but if you do, I'm also wondering if it's possible to use ffmpeg to quickly make an animated gif from 2 or more image files (png/jpg). [06:10:53 CET] <Roden> maybe it's an imagemagik question [08:32:11 CET] <jaggz> I have a 10mb mp3, and a 600k jpg, and I'm making a video out of them with the image looped.. [08:33:10 CET] <jaggz> it ends up being a bit over 20mb in size though (it's actually more like 15mb mp3, 700k jpg, and ends up as a 33mb mp4, but probably no matter) [08:33:27 CET] <jaggz> ffmpeg -loop 1 -r 18 -i "$img" -i "$snd" -c:v libx264 -tune stillimage -c:a aac -b:a 160k -pix_fmt yuv420p -shortest -q:a .1 "$out" [08:34:01 CET] <jaggz> I was testing with that possibly horrible -q:a (I haven't even listened yet), but am wondering what options are available to make the final mp4 much smaller [08:50:15 CET] <jaggz> ok.. setting video quality helps.. now I'm down at 900k and it still sounds and looks nice :) [08:50:23 CET] <jaggz> out-ba160k-qa1.8-qv31-r18.mp4 [08:50:33 CET] <jaggz> I wonder if -loop 1 is a problem [09:25:15 CET] <Roden> jaggz that sounds really strange to me, man, but I don't have a deeper explanation of why I expect things to go differently other than an instinct: I feel like the same jpg file should be referenced at multiple times in an optimized video rather than having the file redundantly loaded throughout the video. [09:27:24 CET] <Roden> Like it should only load frame 1 and frame 2 and activate like this: (t=1 fr=1), (t=2 fr=2), (t=3 fr=1), (t=4 f=2) ... (t=last_odd_time f=1), (t=last_even_time f=2). [09:28:16 CET] <Roden> I've also been very curious about how to composite images into video format without having them load redundantly. If you see me around and happen to come upon an answer, or if you have a URL linking to your question online, please let me know. [10:01:45 CET] <helloimdumb> hello [10:01:57 CET] <helloimdumb> I'm wondering if someone can help me [10:02:39 CET] <helloimdumb> I'm trying to get ffmpeg running on a new pc [10:03:01 CET] <helloimdumb> but unlike prior versions, this one doesn't come with ff-prompt.bat [10:04:16 CET] <helloimdumb> how am I supposed to open ffmpeg with the new version? [11:09:45 CET] <wondiws> hi furq [11:51:32 CET] <pagios> hi all, is it possible to create an rtmp redirect like an http redirect? Idea is to completely send a broadcaster to a new server instead ofrelaying netween servers and consuming server to server bandwidth [11:52:20 CET] <BtbN> no [11:52:44 CET] <BtbN> rtmp is an entirely different protocol [11:56:35 CET] <nimaje> how does being a diffrent protocol imply it can't have a similar feature, BtbN? [11:57:13 CET] <BtbN> rtmp is pretty damn primitive [11:57:27 CET] <BtbN> and somehow over-complex at the same time, but not in a useful way [11:58:55 CET] <nimaje> so the anwser is just that it doesn't have such a feature [12:01:10 CET] <TheAMM> pagios: see nginx-rtmp [12:01:20 CET] <furq> nginx-rtmp doesn't do it either [12:01:25 CET] <furq> there's an ancient open issue for it [12:01:47 CET] <furq> FMS has some kind of redirect functionality but it's not clear if anything actually supports it [12:01:51 CET] <TheAMM> nevermind then [12:01:57 CET] <furq> lavc/librtmp don't [12:02:03 CET] <furq> or lavf rather [12:30:53 CET] <pagios> so what options do i have [13:11:33 CET] <BtbN> nginx-rtmp can re-stream it [14:21:06 CET] <kepstin> pagios: the best option is probably to use dns to load balance, possibly in combination with running the traffic from one ip to multiple severs with a frontend like haproxy. [14:27:14 CET] <wondiws> hi furq [14:29:33 CET] <CounterPillow> ^- what is the "hi furq" all about? [14:34:18 CET] <wondiws> CounterPillow, I wanna talk with him, a followup on an earlier conversation [14:34:41 CET] <CounterPillow> maybe you should just say what you want to say. [14:51:04 CET] <wondiws> Then I will ask another question that I can't find a more appropriate active channel for: I took an old VCD from the shelf. I wanted to copy it to the harddrive, in linux I needed to use cdrdao in raw mode. Windows gives me a filesystem an I can copy the (virtual) DAT file to C:\ but the resulting file is in RIFF format. [14:51:15 CET] <wondiws> This is not the format on the disc itself I imagine [14:51:29 CET] <wondiws> what kind of format is this? It isn't AVI either I suppose [14:52:35 CET] <DHE> I've burned VCDs in the past. it's just .mpg format video on a track containing mpeg1video (!) [14:53:07 CET] <DHE> though AVI is a variant of RIFF I think... [14:53:17 CET] <wondiws> that is true [14:54:46 CET] <wondiws> In linux you can't copy this *.dat file [14:55:38 CET] <DHE> linux doesn't work that way. there's not really a filesystem. I think it might literally by a .mpg file burned directly as a track on the CD [14:57:05 CET] <wondiws> DHE, yes on track 2. Track 1 is a filesystem though [14:57:28 CET] <wondiws> but the problem is, it is in CDXA mode [14:58:26 CET] <wondiws> I think the mpeg does use 2 byte CRC per sector though, I'm afraid that if I readraw track 2 I will get the CRC intertwined in the resulting file, right? [14:59:33 CET] <DHE> this is going beyond my level of knowledge. but since unrepaired corruption would be highly visible on playback I would guess VCD would use the full ECC of data CD modes [15:01:52 CET] <wondiws> DHE, no, it doesn't use ECC for the actual video [15:05:46 CET] <kepstin> yeah, vcd did a tradeoff where they accepted video corruption in favour of getting more video on disk [15:06:13 CET] <DHE> interesting.. [15:14:26 CET] <kepstin> wondiws: fwiw, if your cdrdao toc file says the track is MODE2/2352 then it should be just the data, no crcs [15:14:44 CET] <kepstin> er, no cd-rom format junk specifically [15:15:26 CET] <wondiws> kepstin, it says MODE2_FORM_MIX [15:15:56 CET] <kepstin> oh, i'm getting the cue and toc format mixed up, whoops [15:17:16 CET] <kepstin> hmm, that includes the subchannel data (cdrom stuff), it's not just the video :/ [15:23:39 CET] <wondiws> kepstin, yes, I don't understand why they would want the to save those few bytes on the tiny track 1 [15:32:00 CET] <devinheitmueller> Is there still a ffmpeg meeting at 11:00 EST? Has anyone posted meeting info? [16:08:33 CET] <durandal_1707> devinheitmueller: nope, j-b is skiing [16:08:53 CET] <devinheitmueller> @durandal_1707 Ok, thanks. [16:11:33 CET] <durandal_1707> devinheitmueller: it will probably be in next few weeks, follow -devel mailing list [16:14:42 CET] <devinheitmueller> ok [16:31:02 CET] <unixabg> Greetings, I launch a process to grab images from a rtsp stream and it works great. I would like to leave the process running [16:31:29 CET] <unixabg> and send a sighup or something to send it new params and save the overhead of launch with new params. Is this possible? [17:47:22 CET] <andymaier> I am using ffprobe to get the title from an AVI file, outputting it in JSON format. When the title in the file contains non-ASCII characters, ffprobe represents these with the Unicode replacement character (U+FFFD). Is there any way to tell ffprobe to either assume a particular codepage for the metadata, or to output it in bytes as read from the file (i..e. without converting it to Unicode)? [19:15:27 CET] <relaxed> andymaier: I don't think so, look at string_validation_replacement in the man page [19:26:03 CET] <dostoyevsky> > [mp4 @ 0x72211c0] AV1 muxing is currently not supported. [19:26:49 CET] <andymaier> @relaxed Thanks much, this helped! Using -of json=sv=ignore made the difference for me, then i can process the result myself. [19:27:01 CET] <dostoyevsky> Does ffmpeg support av1 muxing with mp4 these days? (I think the error comes from an older version of ffmpeg) [19:41:25 CET] <Hello71> did you search for that string in the source code [20:04:49 CET] <cehoyos> av1 muxing in mp4 is supported [22:31:27 CET] <^Neo> Hello, in ff_spdif_probe in spdifdec.c, why does it look to see if the data type (buf[1]) is less than 0x37/55? [22:46:50 CET] <BtbN> The data type? [22:46:54 CET] <BtbN> You mean... the value? [22:47:01 CET] <BtbN> Probably because the format looks like that. [22:50:28 CET] <^Neo> yeah, I'm reading through the 61937 standard and it's not obvious [22:55:02 CET] <^Neo> the datatype field is supposed to be 7 bits, but is limited to 4 bits (0x1F) [22:55:12 CET] <^Neo> there was an ammendment in 2011 to add sub data types [22:55:16 CET] <^Neo> and the code is from 2010 [23:02:35 CET] <cehoyos> Do you see a sample for which probing fails? [23:18:27 CET] <^Neo> No, not really. I'm just trying to figure out why it's 37... [23:18:30 CET] <^Neo> I think I figured it out though [23:18:52 CET] <^Neo> 0x37 = 10111 for the datatype, and in the 2010 version of the standard, it only went up to datatype 22 (10110) [00:00:00 CET] --- Tue Jan 7 2020

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