January 2020 - Ffmpeg-devel-irc

ffmpeg-devel.log.20200125
by burek 26 Jan '20

26 Jan '20

[00:00:00 CET] --- Sat Jan 25 2020

1 0

ffmpeg.log.20200125
by burek 26 Jan '20

26 Jan '20

[00:19:17 CET] <emr> Hello i'm trying to use hw accel, however i got different results here is the story https://privatebin.net/?fb7414da57fedadd#9aUe5Q2UBCgTfX4RJzeGuw8dAmn6tQS9HQ… [00:22:44 CET] <BtbN> The md5 sum of an mp4 file is not very useful as comparison [00:23:25 CET] <BtbN> Also, you aren't seriously expecting two lossy encoders to produce bitexactly the same result? [00:24:57 CET] <emr> but the sizes are also different like test.mp4 -> 258M and test2.mp4 -> 185M [00:25:07 CET] <emr> so i was little worried about it [00:25:39 CET] <BtbN> It's two different encoders after all [00:25:55 CET] <BtbN> I'm not sure if either of them even care about -qp [00:26:37 CET] <emr> BtbN, but its still same for viewers right, can i use safely [00:26:38 CET] <cehoyos> For x264 (and x265), you cannot expect to get identical output on two runs for identical input (if you use multi-threading) [00:26:59 CET] <BtbN> You will have to look at it to find out how well each encoder did [00:27:29 CET] <BtbN> it's a lossy codec, quality of encoders varies wildly, specially hardware encoders are pretty much always worse than libx264 [00:27:59 CET] <emr> i got it, thanks BtbN, going to check it [05:23:54 CET] <Media_Thor> Greetings! Is there any filter/tools to help identify channels for 6X mono >> 5.1 please? https://trac.ffmpeg.org/wiki/AudioChannelManipulation#a6mono5.1 [07:44:05 CET] <JustLandedOnMars> hi [07:47:23 CET] <JustLandedOnMars> so I used this command: [07:47:27 CET] <JustLandedOnMars> ffprobe -loglevel quiet -print_format flat -show_streams file.mp4 [07:47:40 CET] <JustLandedOnMars> and it prints: [07:47:41 CET] <JustLandedOnMars> streams.stream.0.r_frame_rate="353/12" [07:48:16 CET] <JustLandedOnMars> so is "353" frame rate correct ? [07:48:22 CET] <JustLandedOnMars> because if I just use: [07:48:26 CET] <JustLandedOnMars> ffprobe file.mp4 [07:48:38 CET] <JustLandedOnMars> it prints: [07:48:40 CET] <JustLandedOnMars> Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], 689 kb/s, 29.42 fps, 29.42 tbr, 90k tbn, 58.82 tbc (default) [07:48:48 CET] <JustLandedOnMars> here it is: 29.42 fps [07:48:57 CET] <JustLandedOnMars> so where those 353 came from ? [07:49:56 CET] <JustLandedOnMars> I tryed some other videos, and it prints: [07:50:04 CET] <JustLandedOnMars> streams.stream.0.r_frame_rate="2997/100" [07:50:05 CET] <JustLandedOnMars> streams.stream.0.r_frame_rate="30000/1001" [07:50:19 CET] <JustLandedOnMars> so how to get actual numbers like 30 fps ? [08:37:21 CET] <emr> Hello, https://bpaste.net/4BLA i'm losing video when try to encode via h264_vaapi [11:29:39 CET] <Aelius> is there some disadvantage to the +faststart flag? [11:30:52 CET] <Aelius> Or is it perhaps just unnecessary? I did not know about it until now, and firefox definitely plays my converted videos before downloading all. [13:36:38 CET] <DHE> Aelius: you'll want it for streaming, but it requires a rebuild of the file once the job finishes which could take a while for huge files [13:44:33 CET] <another> JustLandedOnMars: 353/12 = 29.4166666 [13:45:53 CET] <another> Aelius: streaming might work without it if the client uses range-requests [14:06:05 CET] <mss> i am using the alsa hw:0,0 on my machine to record audio but can't have the audio play on anything except for mpd, anyone can help me with that? [14:07:09 CET] <mss> i tried aac, mp3 and wav codecs to test, the file size is far more larger than 0k but i get no audio on mpv, mpg123 or the browser. [14:08:05 CET] <mss> and i can clearly hear the audio recording if i add it to my mpd playlist, but inspecting the file through mpd clients, they say that the audio bitrate is 0khz even though i can clearly hear it. [14:34:26 CET] <another> meh [14:34:53 CET] <another> mss: please post your full commandline + ouput to some pastebin site [14:45:47 CET] <mss> another: `ffmpeg -f x11grab -r 25 -s 1280x800 -i :0 -f alsa -i hw:0,0 test.mkv` this is both audio+video recording, i used just `ffmpeg -f alsa -i hw:0,0 test.wav` for audio only. [14:45:56 CET] <mss> will upload sample in a second. [14:52:28 CET] <mss> another: http://0x0.st/iry7.mkv, here's a demonstartion. if you play that through mpv, it won't output any audio but adding it to mpd and playing it will give you audio. [15:02:33 CET] <debianuser> mss: `mplayer iry7.mkv` plays the audio just fine for me... I can hear someone (you?) clicking on the keyboard and some noise next to a microphone... [15:03:26 CET] <mss> debianuser: i'm probably having some issues with alsa itself i guess. [15:03:34 CET] <mss> something is grabbing and not letting alsa go. [15:04:15 CET] <mss> i'll deal with it with the folks over alsa, i'll come back if i can't figure it out. [15:04:19 CET] <another> sound works [15:05:01 CET] <another> what's surprising me, is that ffmpeg chooses ac3 [16:21:42 CET] <cehoyos> another: If you didn't compile FFmpeg with --enable-libvorbis, ac3 is the default for mkv output (the native vorbis encoder is too bad) [16:22:22 CET] <furq> is there some reason it's not aac other than nobody got around to changing it [16:27:44 CET] <another> huh [16:28:32 CET] <another> goes i never had an ffmpeg w/o libvorbis [17:31:46 CET] <cehoyos> furq: For multi-channel, I believe the ac3 encoder is superior but that doesn't mean you are wrong:-) [17:32:16 CET] <cehoyos> another: I have to compile it with libvorbis when I see a related bug report, I normally never do it [17:43:06 CET] <furq> is libvorbis superior for multichannel [17:44:11 CET] <pink_mist> I thought he just literally said ac3 was superior for multichannel [17:45:44 CET] <Hello71> well if you're going to change the default, why not just change it to opus (notwithstanding that that's not native either) [17:47:48 CET] <furq> pink_mist: superior to aac [17:59:46 CET] <relaxed> Hello71: aac is a native codec and better supported [18:48:52 CET] <cehoyos> furq: I don't know, I thought the ac3 encoder produces good quality [22:53:54 CET] <LFSVeteran> modified transcode_aac.c to use PCM_S32LE: https://pastebin.com/q3TZKaCD [22:54:00 CET] <LFSVeteran> now I get "Could not allocate output frame samples (error 'Invalid argument')" [22:57:03 CET] <BtbN> Why do you want an encoder for a raw format? [22:58:49 CET] <LFSVeteran> Maybe some code can be removed but I'm not familiair with it [22:59:25 CET] <LFSVeteran> what it has to do is "ffmpeg -f s32le -ac 2 -ar <samplerate> -i http://ip:port/stream -acodec pcm_s32le output.wav" in code [23:00:06 CET] <BtbN> And what part of it fails? [23:01:08 CET] <LFSVeteran> it seems that it fails at init_output_frame [23:01:19 CET] <LFSVeteran> row 619 [23:03:28 CET] <BtbN> That part of the code is even fully independend of the "codec" in use [23:05:14 CET] <LFSVeteran> what I have changed: https://pastebin.com/3mSn7B7J [23:05:16 CET] <BtbN> I'd print all the parameters it sets and see if one of them is nonsensical [23:05:56 CET] <LFSVeteran> with AAC it looked like it worked [23:06:16 CET] <BtbN> so one of the few parameters must be wrong [23:06:19 CET] <BtbN> easy to check [23:16:59 CET] <LFSVeteran> nb_samples = 0? [23:21:54 CET] <CustosLimen> hi [23:22:06 CET] <CustosLimen> how do I concatenate many mp3s and add silence between them? [23:27:27 CET] <BtbN> create an mp3 with silence, and put it in between [23:28:03 CET] <BtbN> LFSVeteran, sounds about wrong [23:29:52 CET] <DHE> CustosLimen: you probably want to concatenate silence mp3s into the list [23:30:13 CET] <CustosLimen> ah [23:30:14 CET] <CustosLimen> good idea [23:30:28 CET] <LFSVeteran> (*frame)->nb_samples = frame_size; [23:30:28 CET] <LFSVeteran> (*frame)->channel_layout = output_codec_context->channel_layout; [23:30:28 CET] <LFSVeteran> (*frame)->format = output_codec_context->sample_fmt; [23:30:28 CET] <LFSVeteran> (*frame)->sample_rate = output_codec_context->sample_rate; [23:30:32 CET] <CustosLimen> now to make silence mp4 [23:30:35 CET] <CustosLimen> mp3* [23:30:44 CET] <LFSVeteran> are the only one set, and only nb_samples is zero [23:30:52 CET] <LFSVeteran> guess I have to check that part [23:31:52 CET] <CustosLimen> got it [23:33:43 CET] <BtbN> LFSVeteran, well, you are thus asking it to allocate memory for 0 samples. Which makes no sense. Hence the error. [23:33:52 CET] <LFSVeteran> y [23:34:07 CET] <LFSVeteran> alloc (0) doesn't make sense [00:00:00 CET] --- Sun Jan 26 2020

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ffmpeg-devel.log.20200124
by burek 25 Jan '20

25 Jan '20

[00:00:00 CET] --- Fri Jan 24 2020

1 0

ffmpeg.log.20200124
by burek 25 Jan '20

25 Jan '20

[01:01:08 CET] <kingsley> furq: My benchmarks suggest libvpx renders fastest with about 8 threads. Does 8 seem plausible? [01:01:27 CET] <kingsley> I used... [01:01:36 CET] <kingsley> $ ffmpeg -y -f lavfi -i mandelbrot=s=1920x1080 -threads $CPU_threads -row-mt 1 -frames 50 -c:v libvpx /tmp/bench.webm [01:20:48 CET] <cehoyos> It should not get slower with more threads... [01:20:54 CET] <cehoyos> (If you have more cores) [01:21:17 CET] <cehoyos> But it does get less efficient with every thread, there is no free lunch;-( [01:21:40 CET] <furq> you should probably use a real sample clip rather than mandelbrot [04:34:21 CET] <mys_721tx> Hello all, I was trying to build ffmpeg on Arch Linux. yay could not find the pubkey for the source code signature. Which key server can I use to import it? [05:35:37 CET] <void09> mys_721tx: which version from aur ? i do not see signature checks in ffmpeg-git [05:35:52 CET] <void09> or you mean install ffmpeg, from the main repos ? [06:09:53 CET] <void09> JEEB: [06:18:27 CET] <void09> did anyone get rounding errors when working with fractional frame rates like 23.976... ? [08:07:43 CET] <JEEB> void09: i have no idea wtf ypu're talking about but containers like matroska are not exact with such rates, see "ffprobe -show_packets -of json -select_streams v -i INPUT" for example (pipe it to less or something) [08:08:23 CET] <void09> JEEB: about my issue with ffmpeg selection ending up on a different frame for the same -to time but different -ss time [08:08:32 CET] <void09> with a 23.976.. fps video [08:08:49 CET] <JEEB> also the exact value is 24000/1001 [08:08:54 CET] <void09> I know [08:09:36 CET] <void09> what does it have to do with matroska, if I use -ss -to after -i ? [08:09:42 CET] <void09> and no copy [08:10:27 CET] <JEEB> anyways still on a train and still have no idea what your issue is. i gave you debug_ts to getinfo on which packets get passed withypur commands, and now gave you a way to dump the timestamps of packets wothout anything extra bw done to them [08:10:54 CET] <void09> yes you gave me the debug_ts command but I have no idea to do with that data :P [08:11:05 CET] <JEEB> void09: i have no.fucking idea but you mentioned rounding errors and matroska is not effing exavt [08:11:28 CET] <JEEB> it technically cannot hold exact 24000/1001 tomestamps [08:12:12 CET] <void09> it would need to store that data as an numerator/denominator to be exact, right ? [08:12:43 CET] <furq> it already does [08:12:54 CET] <furq> but the timebase is always 10^n [08:13:23 CET] <furq> other containers support it being some multiple of 1001 [08:13:49 CET] <JEEB> so if you see that the packet timestamps are indeed off compared to your math, which you can check with f.ex. that ffprobe command or debug_ts then you can start making notes about whether or not it is rounding errors [08:14:43 CET] <JEEB> also i don't think you ever actually noted your assumptions and wherethings don't match them [08:15:01 CET] <JEEB> yes you posted two -ss values sure [08:15:34 CET] <JEEB> but without context i don't think anyone can say jack shit wither about your asssumptions or otherwise [08:16:13 CET] <JEEB> i just gave you the tools to dump the packet timestamps from.whatever ypu are reading [08:16:37 CET] <JEEB> so now you can actually check what timestamps there are in your input [08:16:54 CET] <JEEB> and how theycompare to what you thought you were doing [08:17:00 CET] <JEEB> hopefully that helps [08:18:36 CET] <void09> I have no assumptions, don't think I am doing anything :P [08:18:49 CET] <void09> the timecodes I get are from pyscenedetect [08:20:04 CET] <JEEB> and then you match the demuxer/demuxer+ffmpeg things with your command line against the packets in ffprobe [08:20:29 CET] <JEEB> pretty sure that will give you some answers on wtf your command is foing [08:21:41 CET] <JEEB> and prettt sure you have assumptions because you thinl something is being done wrong [08:21:50 CET] <JEEB> that is what i mean with assu'ptions [08:22:01 CET] <JEEB> you can validate or break them with actual data [08:22:37 CET] <JEEB> also thta's enough from me. i have $dayjob to care about and cannot hand-hold [08:24:35 CET] <void09> well obviously something is wrong. a timestamp should hold the exact same frame/data each time you use it [08:58:34 CET] <kode54> I'm trying to split a single 14 channel WAV file into two 7 channel WAV files [08:58:53 CET] <kode54> the input file has a specific layout that needs to be reordered to correspond to the 7.0 output I desire [08:59:07 CET] <kode54> the command line I'm using is [08:59:08 CET] <kode54> ffmpeg -i gsx.wav -filter_complex "channelsplit=channel_layout=14[FLl][FLr][SLl][SLr][BLl][BLr][FCl][FRr][FRl][SRr][SRl][BRr][BRl][FCr],[FLl][FRl][FCl][BLl][BRl][SLl][SRl]join=inputs=7:channel_layout=7.0[a],[FLr][FRr][FCr][BLr][BRr][SLr][SRr]join=inputs=7:channel_layout=7.0[b]" -map "[a]" gsx_L.wav -map "[b]" gsx_R.wav [08:59:22 CET] <kode54> this results in: [08:59:23 CET] <kode54> [AVFilterGraph @ 0x563f474f5500] No output pad can be associated to link label 'SLr'. [09:06:30 CET] <kode54> ah, I can use map_channel [09:06:39 CET] <kode54> but how do I force it to map 7.0 instead of 6.1? [09:08:28 CET] <kode54> ah, it's ignoring my mapping directives [10:18:28 CET] <hendry> when I update ffmpeg, I have to run `sudo setcap cap_sys_admin+ep /usr/bin/ffmpeg` otherwise my subsequent "kmsgrab" invocations will fail. Is there anyway to ease my pain on a distro like Archlinux? [10:37:42 CET] <c_14> hendry: you can use a pacman hook on install/upgrades of the ffmpeg package to run that command for you https://jlk.fjfi.cvut.cz/arch/manpages/man/alpm-hooks.5 [10:46:41 CET] <hendry> c_14: whoa, I didn't know this! [12:18:50 CET] <aleek> hi! is it possible to use map before filter-complex, to select audio streams with particular language? sth like: ffmpeg -i <input> -map 0:a:m:language:eng -map 0:a:m:language:ger -filter_complex "[a:0]filters_for_english[eng_out];[a:1]filters_for_german[ger_out]" -map [eng_out] -map [ger_out] -c:a:0 codec_for_eng -c:a:1 codec_for_ger ... [14:46:36 CET] <ponyrider> aleek: from what i can tell, you are meant to map within the complex filter, as in: -complex-filter "[0:a:0]something[label]" -map '[label [14:49:25 CET] <ponyrider> no need to map [14:50:14 CET] <ponyrider> i guess you are trying to automate it's selection huh [14:55:33 CET] <aleek> i've managed to do it [14:56:08 CET] <aleek> -filter_complex "[0:a:m:language:eng]aresample...[a0out]" and so on [15:00:07 CET] <ponyrider> gggg [15:00:29 CET] <aleek> ? [15:00:41 CET] <aleek> :D [15:01:08 CET] <ponyrider> it g good job lol [15:01:35 CET] <aleek> actually, when using this pattern to resample two input audio streams, it seems to be not possible to switch between them, when outputting to DASH [15:01:51 CET] <aleek> but that might be just my mistake in other place of cmd line [15:04:01 CET] <ponyrider> what do mean ? [15:06:03 CET] <ponyrider> can you post? [15:33:20 CET] <danilo82> hello, its possible to use vaapi in ffplay? [15:59:06 CET] <yepla> hello [15:59:55 CET] <yepla> does it possible to get a h265 from a h264 without losing quality ? [16:02:35 CET] <BtbN> no [16:07:42 CET] <danilo82> -crf 0 ? [16:09:57 CET] <furq> that won't reduce quality but it will also be about 50x bigger than the source [16:15:01 CET] <yepla> in fact i just want reencode my h264 in h265 with same quality [16:16:06 CET] <yepla> danilo82: i think i will try -crf 0 thanks [16:17:17 CET] <danilo82> mah man, just keep it h264 [16:17:42 CET] <danilo82> its a waste of time to re-encode it [16:18:29 CET] <yepla> want a little size [16:18:59 CET] <BtbN> -crf 0 will do the exact opposite of that [16:19:09 CET] <danilo82> ye [16:19:19 CET] <BtbN> And every non-lossless reencode will lose you quality. So best to just keep it as it is. [16:20:00 CET] <yepla> i want reduce size [16:20:10 CET] <yepla> use h265 to reduce size [16:20:18 CET] <danilo82> so you will lose quality [16:21:01 CET] <yepla> i dont get it h265 not reduce size for same quality ? [16:21:42 CET] <danilo82> you can mess with the -bv option [16:23:05 CET] <furq> it would be smaller for the same quality if you were encoding from the same source the h264 copy was encoded from [16:23:22 CET] <furq> you can't reencode something lossy and not lose quality [16:23:55 CET] <danilo82> Original > h265 [16:24:09 CET] <danilo82> not Original > h264 > h265 [16:24:55 CET] <yepla> what you call Original which format is it ? [16:25:14 CET] <danilo82> hmm [16:25:19 CET] <danilo82> the raw file? [16:25:21 CET] <danilo82> idk [16:25:30 CET] <danilo82> you can encode in AV1 [16:25:39 CET] <danilo82> so you will not lose quality [16:25:51 CET] <danilo82> ;D [16:26:05 CET] <BtbN> Even AV1 is a loss encoder, so same rules apply. [16:26:07 CET] <BtbN> +y [16:26:16 CET] <danilo82> shush [16:26:17 CET] <danilo82> XD [16:26:34 CET] <BtbN> Every time you encode with a loss encoder, you lose quality. No way around that. [16:28:57 CET] <danilo82> aey man, i want to use svt but it needs yuv files, how to convert mp4 to yuv? [16:32:03 CET] <furq> ffmpeg -i foo.mp4 bar.yuv [16:32:20 CET] <BtbN> Though there's A TON of stuff that can be in a .yuv file [16:32:24 CET] <BtbN> there isn't really any standard [16:33:10 CET] <yepla> ok guys thanks for your answer it s what i was thinking [16:33:13 CET] <Hello71> the same applies for "a .mp4 file" [16:33:50 CET] <yepla> directly in h265 not in h264 then h265 [16:49:35 CET] <danilo82> are you sure this is the correct way to convert to yuv? [16:50:04 CET] <danilo82> i get "[IMGUTILS @ 0x7ffe226d29d0] Picture size 0x0 is invalid" [16:51:04 CET] <cehoyos> danilo82: You have to specify the resolution, raw are "raw! [16:51:06 CET] <cehoyos> danilo82: You have to specify the resolution, raw are "raw" [16:52:11 CET] <danilo82> how i do that? [16:55:10 CET] <furq> you don't need to do that for raw output [16:55:51 CET] <danilo82> looks like i do need to do that [16:55:53 CET] <furq> and if you're using the ffmpeg svt stuff then you don't need to convert to yuv [17:04:41 CET] <aleek> ffmpeg [...] -af "somefiltering" -map "0:a:m:language:eng" -map "0:a:m:language:ger" -c:a:0 aac -c:a:1 aac [...] does this cmdline results in both audio stream to be filtered independently? [17:05:55 CET] <aleek> afaik I cannot do -filter:a:0 or -filter:a:m:language:eng [17:07:28 CET] <durandal_1707> that is design limitation of ffmpeg cli [17:08:06 CET] <aleek> durandal_1707: what do you mean by that? [17:08:11 CET] <durandal_1707> via API you can create unlimited number of filtergraphs [17:08:46 CET] <aleek> so with this approach, I cannot filter multiple audio streams? [17:08:50 CET] <durandal_1707> aleek: you need to filter all audio streams with single graph [17:09:25 CET] <aleek> it is ok for me to use the same filtergraph configuration [17:09:58 CET] <aleek> so this cmdline will create two filtergraphs with the same configuration, rigth? [17:10:00 CET] <aleek> rigth? [17:10:03 CET] <aleek> grrrr [17:10:06 CET] <aleek> right! [17:10:33 CET] <durandal_1707> no, only single of multiple -vf/af will ever be used [17:10:56 CET] <aleek> so only one of two audio streams will be filtered? [17:11:23 CET] <aleek> even when using map to map two audio streams [17:11:23 CET] <durandal_1707> use -lavfi or -filter-complex to filter more than 1 -> 1 streams [17:12:47 CET] <aleek> yah, I've got some issue with filter_complex. I'm using external packager and creating DASH stream, and player is crashing when switching between audio streams created with filter_complex [17:12:58 CET] <aleek> most likely something with PTS [17:13:24 CET] <aleek> my audio filtergraph: aresample=async=1:min_hard_comp=0.100000:first_pts=0 [17:13:56 CET] <aleek> maybe i should use apts=PTS-STARTPTS, since I use the same for video [17:15:04 CET] <cehoyos> Maybe you want to tell us how we can reproduce the crash? [17:15:21 CET] <cehoyos> Ah, the player not FFmpeg? [17:15:23 CET] <aleek> the player is crashing, not the ffmpeg [17:15:24 CET] <aleek> :) [17:15:37 CET] <aleek> let me play with setpts and asetpts [18:36:28 CET] <mys_721tx> void09: it's from ffmpeg-full. The maintainer said they have no issue importing the keys. I left them a message about which key server they were using. [00:00:00 CET] --- Sat Jan 25 2020

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ffmpeg-devel.log.20200123
by burek 24 Jan '20

24 Jan '20

[00:00:00 CET] --- Thu Jan 23 2020

1 0

ffmpeg.log.20200123
by burek 24 Jan '20

24 Jan '20

[02:40:51 CET] <kingsley> How do you tell version 1.7+ of libvpx to use multiple CPUs, and ideally, threads, to encode faster? [02:50:22 CET] <Hello71> ... what? [02:52:50 CET] <furq> -row-mt 1 -threads 8 [02:52:58 CET] <furq> and then cross your fingers because it probably still won't work very well [02:59:37 CET] <kingsley> furq: Thank you. My testing is consistent with your suggestion. [03:00:08 CET] <kingsley> I'll only add that it complained when I asked for more than 64 threads. [03:21:38 CET] <SirNapkin1334> um, so, I updated my entire WSL installation, but now apt-get reports that the newest version of ffmpeg is 4.1!! better than 3.7, but still, does anyone know how to actually update? [03:21:56 CET] <SirNapkin1334> i sudo do-release-upgraded to the latest version of ubuntu [03:22:26 CET] <SirNapkin1334> but it still says it's on the latest version 4.1, whereas launchpad.net says newest is 4.2 [03:58:41 CET] <Deep6> hey guys wondering if I can get some assistance on an "emergency" from my wife? I have to concatenate a small mov onto the front of a corporate mp4 file... [03:59:52 CET] <Deep6> I have tried a few things, converted the mov to an mp4, then concatenating, that seemed to work ok, except the audio was lost on the second video, ...googled that. turns out I needed to add a silent audio track...did that... [03:59:59 CET] <Deep6> still no joy... [04:00:53 CET] <Deep6> then upscaled the first video to the res of the second...and added the silent audio. no I get some sort of weird big gap between the files and then it's like the second is in fast forward [04:41:35 CET] <SirNapkin1334> hi, where is the public log for this channe:/ [04:41:39 CET] <SirNapkin1334> *channel? [04:43:48 CET] <Deep6> there's no one here SirNapkin1334 [04:43:50 CET] <Deep6> :( [04:44:05 CET] <SirNapkin1334> wait but it says its publicly logged [04:47:05 CET] <Deep6> yah, probably somewhere on the website it'll say where [06:30:44 CET] <AlaskanEmily> So I want to write a .au file with 8-bit PCM in it. Specifying a .au output apparently defaults to using the signed 16-bit format, and using -sample_fmt u8 doesn't seem to work, it tells me that the PCM16 encoder can't do u8. [06:49:26 CET] <furq> AlaskanEmily: -c:a pcm_s8 [06:50:05 CET] <AlaskanEmily> furq: Thanks! [07:05:05 CET] <void09> anyone here that can help me try to figure out why ffmpeg selects a video segment (encoding, not -copy) with 1 frame difference given the same -to time , but different -ss time ? 1 frame extra at the end of one of the encodes [07:05:12 CET] <void09> ffmpeg -i "video.mkv" -ss 00:02:15.844 -to 00:02:16.845 -crf 0 short.mkv <- ends up at the correct frame [07:05:17 CET] <void09> ffmpeg -i "vide.mkv" -ss 00:01:59.619 -to 00:02:16.845 -crf 0 long.mkv <- ends up at +1 frames [14:48:41 CET] <danilo82> hello, i want to merge 2 audio files, one is for the right side and the second if for the left side, i want to merge but keep it lossless [14:49:14 CET] <DHE> what do you mean keep it lossless? is it already lossless? [14:49:26 CET] <danilo82> yes [14:49:45 CET] <danilo82> rew file converted to wav [14:49:53 CET] <danilo82> raw* [14:50:39 CET] <DHE> ffmpeg -i left.wav -i right.wav -filter_complex amerge output.wav # I think this will work [14:51:17 CET] <danilo82> :P but how can i know if it worked? [14:51:35 CET] <DHE> I like to use ffprobe to see if the resulting wav is stereo.... or you know... play it [14:53:20 CET] <danilo82> they are not stereo [14:53:46 CET] <danilo82> no no [14:53:55 CET] <danilo82> they are stereo [14:54:10 CET] <DHE> then I will require a better description of what you want to accomplish [14:55:56 CET] <danilo82> music from a game, the left file is created to be played in the left speaker and the right is made to be played in the right speaker, they are almost the same except for some details [14:56:45 CET] <DHE> so merge both wavs to mono, then use each resulting mono channel as a speaker in a stereo config? [14:58:44 CET] <danilo82> i want to merge the files because they only sound right if they are merged [14:59:58 CET] <DHE> yes, but merging 2 stereo files produces 4 channels. how do you want that presented? [15:00:30 CET] <DHE> mix left+left and mix right+right straight up like 2 wavs are playing simultaneously into your ears? [15:02:56 CET] <danilo82> :/ [15:03:05 CET] <danilo82> but i think your first command worked [15:03:57 CET] <danilo82> left file "Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 32000 Hz, 2 channels, s16, 1024 kb/s" [15:04:17 CET] <danilo82> right file "Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 32000 Hz, 2 channels, s16, 1024 kb/s" [15:04:53 CET] <danilo82> merged with your command "Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 32000 Hz, 4.0, s16, 2048 kb/s" [15:05:35 CET] <pink_mist> that now has 4 audio channels [15:05:39 CET] <danilo82> i see [15:06:27 CET] <danilo82> so i should convert each file to mono first? [15:17:11 CET] <danilo82> https://trac.ffmpeg.org/wiki/AudioChannelManipulation [15:17:13 CET] <danilo82> there [15:17:22 CET] <danilo82> 2 × stereo stereo [15:17:30 CET] <danilo82> will this work for me? [15:42:32 CET] <cehoyos> Yes, but "this" are several different operations, only you can know which one is correct for your use case [16:27:44 CET] <filpAM> can ffmpeg decode .exo chunks? [16:28:27 CET] <JEEB> try just feeding whatever that is to ffprobe with `ffprobe -v verbose -i INPUT` [16:28:40 CET] <JEEB> if it comes up with anything semi-sane, you could at least attempt [16:31:54 CET] <filpAM> I am downloading some videos from youtube for my Mom. My mom had youtube premium so I downloaded them with it, but the videos aren't present in the smartphone download directory. I got pissed but then I checked the Android internal storage and figured where the downloaded videos are located. They happen to be divided into chunks with extension .v3.exo.. [16:32:44 CET] <JEEB> if it's DRM'd it's DRM'd and that's no longer within ffmpeg's area of expertise [16:34:06 CET] <filpAM> such a bullshit [16:35:17 CET] <JEEB> and if you just want to download normal videos from youtube, just use youtube-dl for it. probably if it's some premium DRM content it will not work, but at least the random cat videos will :P [16:36:49 CET] <filpAM> JEEB: I tried to be ethic(because these videos are for mom church) but youtube isn't being back. [17:03:43 CET] <void09> anyone here that can help me try to figure out why ffmpeg selects a video segment (encoding, not -copy) with 1 frame difference given the same -to time , but different -ss time ? 1 frame extra at the end of one of the encodes [17:10:44 CET] <pink_mist> I would presume it's simply floating point rounding errors [18:30:54 CET] <void09> I thought of that too [18:31:45 CET] <void09> as the video is 23.976fps [18:33:14 CET] <void09> which is actually 23,976023976 [18:40:02 CET] <Hejkki> hello. I have a file from my camera that is corrupted. The battery died while recording. How can i fix the file with ffmpeg or some other tool? if i try ffmpeg -i damaged.MP4 out.mp4 it will only say Invalid data found when processing input [18:40:30 CET] <Hejkki> the file size is about 2GB [18:41:45 CET] <Hejkki> using Linux [18:45:20 CET] <JEEB> most things unfortunately write globally indexed mp4 [18:45:35 CET] <JEEB> which means that unless you want to re-write the whole file, you need to write it at the end [18:46:02 CET] <JEEB> unfortunately, that index also happens to include various content-related information [18:46:21 CET] <JEEB> you can possibly fix it to a point if you already have some file with the exact same parameters, but working [18:46:45 CET] <JEEB> ah, found the project https://github.com/ponchio/untrunc [18:47:41 CET] <Hejkki> yes, i do have 2 files, one working and another not working [18:48:37 CET] <Hejkki> will try that, thanks :) [18:50:39 CET] <Hejkki> JEEB: it gives me track.cpp:57:11: error fatal: config.h: No existe el fichero o el directorio [18:50:51 CET] <Hejkki> oh sorry, i have it in spanish,.. [18:50:53 CET] <Hejkki> sec [18:51:05 CET] <JEEB> Hejkki: you probably need to actually read that project's documentation and build it :P [18:51:38 CET] <Hejkki> i am building it [18:51:44 CET] <Hejkki> the build process said like that [18:51:54 CET] <Hejkki> track.cpp:57:11: fatal error: config.h: No such file or directory [18:52:16 CET] <JEEB> yea, and clearly you need to go get support from that project since it is not FFmpeg (while it utilizes FFmpeg and I know of it) [18:52:30 CET] <JEEB> unfortunately I have not built it myself nor have I used it :P [18:52:37 CET] <JEEB> but it really sounds like you haven't read the documentation [18:53:14 CET] <Hejkki> i have [18:53:22 CET] <Hejkki> assholes in the internet all the day [18:53:54 CET] <JEEB> heh, and I tried to say it without too much negativity :P [18:54:08 CET] <JEEB> it just sounded like he hadn't configured the project, which I would hope is part of their build docs [18:54:55 CET] <pink_mist> it isn't [18:55:04 CET] <pink_mist> they suggest you just run a bare g++ command [18:55:09 CET] <pink_mist> not even make [18:55:49 CET] <JEEB> right [21:10:56 CET] <snooky> hi all [23:10:34 CET] <DrNefario> hey, how are you supposed to pronounce ffmpeg? [23:11:30 CET] <furq> eff eff emm peg [23:28:19 CET] <JEEB> äf äf ämpäri [23:31:21 CET] <pink_mist> electronice frontier foundation electroic frontier foundation enterprise metadata manager pci express graphics [23:46:14 CET] <DrNefario> oops i forgot i had this open [23:46:59 CET] <DrNefario> i guess i was pronouncing it right, thanks [00:00:00 CET] --- Fri Jan 24 2020

1 0

ffmpeg-devel.log.20200122
by burek 23 Jan '20

23 Jan '20

[00:00:00 CET] --- Wed Jan 22 2020

1 0

ffmpeg.log.20200122
by burek 23 Jan '20

23 Jan '20

[03:18:25 CET] <gregorycu> "An email was sent to --- with a token to verify your new address. Please check your email and enter the token in the form below." [03:18:34 CET] <gregorycu> I haven't got an email in 2 hours. Think something is busted. [06:22:57 CET] <SirNapkin1334> hi, I want to update ffmpeg with apt-get, and launchpad.net reports that ffmpeg has pushed 4.2, however, apt-get still says that 3.4.6 is the newest version, even after sudo apt-get update [06:28:28 CET] <kepstin> ah, you're using ubuntu bionic (18.04). ubuntu generally doesn't update to new major versions of ffmpeg within a release [06:28:51 CET] <kepstin> you can either switch to a newer ubuntu release, or install it from elsewhere [06:30:26 CET] <kepstin> fwiw, the only ubuntu version with ffmpeg 4.2 is focal, which hasn't even reached beta yet (comes out in april) [08:39:18 CET] <void09> anyone here that can help me try to figure out why ffmpeg selects a video segment (encoding, not -copy) with 1 frame difference given the same -to time , but different -ss time ? 1 frame extra at the end of one of the encodes [09:37:28 CET] <vlt> void09: Can you show the command you used? [09:37:34 CET] <void09> vlt: sure [09:38:29 CET] <void09> ffmpeg -i "video.mkv" -ss 00:02:15.844 -to 00:02:16.845 -crf 0 1.mkv [09:39:09 CET] <void09> this one produces the extra frame [09:40:49 CET] <void09> wait no, reset [09:41:12 CET] <void09> ffmpeg -i "video.mkv" -ss 00:02:15.844 -to 00:02:16.845 -crf 0 short.mkv <- ends up at the correct frame [09:41:29 CET] <void09> ffmpeg -i "vide.mkv" -ss 00:01:59.619 -to 00:02:16.845 -crf 0 long.mkv <- ends up at +1 frames [09:41:36 CET] <void09> video.mkv* [09:42:14 CET] <void09> I can give the video link if you want to try [09:46:59 CET] <termos> do I still have to subtract stream.start_time from every packet.pts when reading from UDP, since it can have a random start pts? [10:39:49 CET] <lavaflow> how can I create a video from several image files without using a single -i and wildcards? [10:40:00 CET] <lavaflow> using one -i for each image file doesn't seem to do what I want [10:40:14 CET] <lavaflow> I assume there is a way to tell it to concat them all, but I'm not sure how to do that [11:08:03 CET] <durandal_1707> lavaflow: see image2 demuxer documentation, assuming images are correctly numbered and same format and size [11:08:34 CET] <lavaflow> durandal_1707: the images are not in alphanumerical order [11:09:26 CET] <durandal_1707> thatn use multiple -i and concat filter? [11:20:47 CET] <funnybunny2> With avio_alloc_context, is a read function necessary as in the following example? I imagined ffmpeg could just read from the passed in buffer itself if no custom behavior is necessary https://ffmpeg.org/doxygen/trunk/avio_reading_8c-example.html#a22 [11:21:33 CET] <funnybunny2> In the above example, all the read does is memcpy into another buffer [11:21:39 CET] <funnybunny2> *read function [11:22:29 CET] <funnybunny2> What I'd like to know is whether avio_alloc_context(file_contents, file_size, 0, NULL, NULL, NULL, NULL); is wrong. I get a segfault, so I guess so [11:31:00 CET] <lavaflow> is there a way to use the concat filter without specifying each input explicitly? [11:48:03 CET] <funnybunny2> Sorry, I dced [12:02:28 CET] <funnybunny2> Nevermind. It seems to work without a custom read function [12:02:42 CET] <funnybunny2> I just forgot to allocate a format context [12:54:07 CET] <BabySoda> guys how can i reduce the size of my gif while retaining sharp edges? [12:54:37 CET] <BabySoda> i tried creating it using ffmpeg, and initially the file size was good, but the edges were all fuzzy [13:00:22 CET] <durandal_1707> BabySoda: do you use palettegen and paletteuse? [13:01:05 CET] <BabySoda> durandal_1707: er yeah, but i'm not sure if i did it correctly [13:01:28 CET] <BabySoda> i mean its in my filter_complex flags, but... [13:02:15 CET] <BabySoda> durandal_1707: and also, does the size of each frame PNG matter? [13:02:31 CET] <BabySoda> (as in can it be too high res) [13:02:53 CET] <BabySoda> at the moment each component frame is about 12K [13:10:38 CET] <BabySoda> anyone know how to improve this/ [13:10:40 CET] <BabySoda> ? [13:17:29 CET] <cehoyos> Please use pastebin or similar to provide the command line you tested together with the complete, uncut console output: FFmpeg is supposed to produce high-quality gifs [13:25:07 CET] <BabySoda> cehoyos: oh, i got it working somehow :) [13:25:19 CET] <BabySoda> oh wait, no i didn't [13:26:13 CET] <BabySoda> for some reason when i use this command (https://dpaste.org/VJxp) the gif only uses like 7 of the frames before reseting [13:27:20 CET] <BabySoda> cehoyos: any idea why that is? [13:28:27 CET] <BabySoda> cehoyos: the output of ffmpeg -> https://dpaste.org/J53v [13:29:06 CET] <BabySoda> ah it even sames frame=7 in that output [13:29:13 CET] <BabySoda> but i don't know why though [13:32:23 CET] <cehoyos> Please test current FFmpeg git head before requesting support here - you cannot scale pal8 input (you can but it produces the effects you see) [13:32:51 CET] <durandal_1707> why you use reverse? it uses lot of memory [13:35:22 CET] <BabySoda> what is pal8? [13:35:30 CET] <BabySoda> is that related to why its only giving me 7 frames? [13:36:01 CET] <BabySoda> durandal_1707: as in it takes ffmpeg longer to run? [13:36:07 CET] <durandal_1707> pal8 is palette pixel format used internally [13:36:19 CET] <durandal_1707> BabySoda: nope, more RAM [13:36:26 CET] <BabySoda> durandal_1707: ah ok [13:36:49 CET] <BabySoda> not really sure what that sentence means, "cannot scale pal8 input" [13:37:09 CET] <BabySoda> also, why does ffmpeg output say 256 colors generated [13:37:17 CET] <BabySoda> when my gif only has 2 [13:50:55 CET] <BabySoda> not sure what's going on tbh [13:52:09 CET] <c_14> BabySoda: the palette you're generating is 16x16, i.e. it defines 256 colors [13:52:37 CET] <BabySoda> c_14: ah, can i make it smaller? [13:52:37 CET] <c_14> and for the pal8 thing, try putting a format=rgb filter before the scale filter [13:52:40 CET] <BabySoda> 2x2? [13:53:04 CET] <c_14> no point, gif uses a 256-color palette [13:53:32 CET] <c_14> you can add max_colors=2 to the options for the palettegen filter if you want though [13:53:47 CET] <c_14> that just sets all unused palette entries to black [13:54:00 CET] <pink_mist> wouldn't 2x2 be 4 colours [13:54:29 CET] <BabySoda> pink_mist: sure, but i thought maybe it has to be an nxn format [13:54:47 CET] <BabySoda> pink_mist: also allows me to use more colors if i need [13:55:07 CET] <BabySoda> c_14: do i just add 'format=rgb,scale=...'? [13:55:13 CET] <c_14> yeah [13:55:30 CET] <BabySoda> gives me 'invalid pixel format "rgb"' [13:55:54 CET] <c_14> eeh, rgb24 [13:56:06 CET] <c_14> rgb0, rather [13:56:19 CET] <c_14> though rgb24 would also work [13:56:44 CET] <c_14> although, if your pngs have transparency you'll want rgba [13:57:12 CET] <BabySoda> c_14: ah yeah they do [13:57:16 CET] <BabySoda> let me try that [13:58:12 CET] <BabySoda> c_14: thanks, i think that works, only problem is the gif still is only using 7 frames [13:58:26 CET] <BabySoda> (even though the images/ directory has 100 images in it) [13:58:52 CET] <c_14> probably the glob messing up, try using cat *.png | ffmpeg -f png_pipe -i pipe:0 [14:01:00 CET] <BabySoda> c_14: er now my gif is mostly a single color, with the other color in the edges [14:01:11 CET] <BabySoda> is that a palette issue? [14:01:18 CET] <c_14> sounds like it [14:01:31 CET] <c_14> did you add the max_colors option? just remove it [14:01:42 CET] <BabySoda> c_14: hadn't added that yet [14:03:04 CET] <BabySoda> c_14: that's my command btw -> https://dpaste.org/8iES [14:03:16 CET] <BabySoda> ah crap wait [14:03:20 CET] <BabySoda> i think i know what i did [14:03:42 CET] <BabySoda> c_14: my bad, all working now [14:03:49 CET] <BabySoda> but still getting the 7 frames [14:04:06 CET] <BabySoda> only now the gif shows each frame a lot faster [14:04:44 CET] <c_14> does the ffmpeg output mention drop= somewhere? [14:05:20 CET] <BabySoda> c_14: doesn't look like it [14:05:52 CET] <BabySoda> c_14: can i have a filename like "my_first_frame.png"? [14:06:01 CET] <BabySoda> can ffmpeg not handle underscores or something? [14:06:17 CET] <c_14> nah, that should be fine [14:06:32 CET] <c_14> is one of the frames a different format or resolution perhaps? [14:06:45 CET] <c_14> or it might be the reverse filter? [14:06:51 CET] <c_14> try removing the reverse filter and using tac instead of cat [14:08:02 CET] <BabySoda> c_14: hmm that gives -> https://dpaste.org/8tKR [14:08:17 CET] <BabySoda> "Cannot determine format of input stream 0:0 after EOF" [14:08:28 CET] <c_14> zsh: no matches found: images/*.png [14:08:35 CET] <BabySoda> ah crap [14:08:43 CET] <furq> that's not how tac works [14:09:01 CET] <c_14> wait, you're right [14:09:08 CET] <c_14> you'd have to cat *.png | tac | ffmpeg [14:09:21 CET] <c_14> can you just rename the files so the glob picks them up in the intended order? [14:09:25 CET] <furq> ls -r * | xargs cat | ffmpeg [14:09:40 CET] <c_14> or try that^ if you don't have spaces in the filenames [14:09:54 CET] <BabySoda> c_14: i could, not it's not ideal [14:10:36 CET] <BabySoda> hmm still getting 7 frames, but not the non-reversed frames [14:11:00 CET] <BabySoda> c_14: these are composited .PNGs btw [14:11:14 CET] <BabySoda> i.e. they .PNGs made from other PNGs, could that be why it's not working? [14:11:21 CET] <BabySoda> *they are [14:12:18 CET] <c_14> maybe [14:12:30 CET] <c_14> if ffmpeg's png decoder doesn't like it [14:12:41 CET] <BabySoda> what's odd is that the file sizes are oh these images are different [14:12:53 CET] <BabySoda> for example the last 7 files are 21/21K, while the one before that is 14K [14:13:02 CET] <furq> png is compressed [14:13:16 CET] <BabySoda> furq: right but they should all be similar size [14:13:48 CET] <BabySoda> i'll try removing the ones that are "outliers" [14:13:54 CET] <furq> if they're only two colors then i wouldn't be surprised if there were big differences in compressibility [14:14:23 CET] <BabySoda> furq: but if the contents of each PNG is very similar, that shouldn't cause these big differences right? [14:15:50 CET] <c_14> BabySoda: if you try encoding without reversing the order, how many frames does it encode? [14:15:53 CET] <BabySoda> c_14, furq: ok that worked, removing the "odd" images now fixes the frames [14:16:38 CET] <furq> ffmpeg should be throwing an error somewhere if some of the files are broken [14:17:02 CET] <BabySoda> doesn't seem to [14:17:14 CET] <c_14> furq: don't think it does with the image2 demuxer [14:19:37 CET] <BabySoda> c_14: i think i stitched these other PNGs together using imagemagick, should i do this with ffmpeg instead? [14:22:51 CET] <c_14> if it works with that I guess [14:37:42 CET] <BabySoda> c_14: ok that seems to work, i think i was doing something wrong with imagemagick to concat the PNGs [14:38:09 CET] <BabySoda> c_14: would a faster cpu speed up this process of concatting images? [14:38:25 CET] <c_14> it shouldn't be that cpu intensive [14:38:34 CET] <c_14> so maybe, but probably not that much [14:39:39 CET] <BabySoda> c_14: ah, is it more gpu dependent or something? [14:40:08 CET] <c_14> it's probably mainly io and the scale filter [14:40:12 CET] <c_14> which is single-threaded [14:40:42 CET] <cehoyos> GPU is not involved [14:41:00 CET] <BabySoda> c_14: ah i see, so if i wanted to speed up this process significantly, i would have to use a group of machines or something? [14:41:16 CET] <BabySoda> each with the fastest single-threaded processor [14:41:24 CET] <cehoyos> You would first have to check which part of the process takes time [14:41:26 CET] <BabySoda> e.g. 9900K or something [14:42:34 CET] <cehoyos> If your images always use the same colours, you could speed the process up by pre-calculating the palette and not using palettegen for all frames [14:43:12 CET] <BabySoda> cehoyos: yeah they do use the same palette, like 4 colors max [14:43:19 CET] <BabySoda> *the same colors [14:43:33 CET] <cehoyos> It is likely that paletteuse is the cause for the slow encode, you cannot speed that up (only the other steps in the pipeline) [14:44:30 CET] <BabySoda> is my palette stuff optimized with this command? https://dpaste.org/LORL [14:44:47 CET] <BabySoda> do i have to generate the palette seperately or something? [14:45:54 CET] <cehoyos> You can pre-calculate the palette to avoid running palettegen [14:47:04 CET] <BabySoda> cehoyos: and i do that by running "ffmpeg -i frames/1.png -vf palettegen=16 palette.png"? [14:49:17 CET] <cehoyos> I am not sure if limiting it to 16 has any advantage but yes [14:49:47 CET] <BabySoda> and then i just add another "ffmpeg ... -i palette.png" and remove the palettegen from "...; [a] palettegen=resever_transparent=on:..."? [14:50:06 CET] <cehoyos> and the split [14:50:54 CET] <BabySoda> er what does "No output pad can be associated to link label 'b'" mean? [14:51:51 CET] <furq> you don't need the split any more [14:52:01 CET] <BabySoda> i removed it, but i still get that error [14:52:14 CET] <BabySoda> do i have to remove [b] from that same part? [14:52:23 CET] <BabySoda> or i remove [b] from the whole command? [14:53:42 CET] <BabySoda> this is my command btw -> https://dpaste.org/WRzW [14:54:29 CET] <BabySoda> i removed the "split" after "falgs=lanczos" and the palettegen before "reserve_transparent [15:03:10 CET] <BabySoda> got it working by removing that section after flags=lanczos [15:07:30 CET] <BabySoda> thanks for the help so far guys, really appreciate it [16:19:01 CET] <SirNapkin1334_> thank you kepstin, out of curiosity, how can I switch to a new version of the ubuntu? both on WSL and on an actual linux installation, since I use both [18:18:53 CET] <chaosblade> SirNapkin1334, in WSL its mostly just "sudo do-release-upgrade" iirc [18:19:14 CET] <chaosblade> i never recall if thats a ubuntu thing in general or a WSL thing [18:33:03 CET] <chaosblade> hmk, basic static binary with x264 and the other usual suspects compiled and working. Now to somehow hack nvidiot support in as non-static while maintaining the rest [18:58:09 CET] <kepstin> do-release-upgrade is a general ubuntu thing, is how you do release upgrades on bare metal systems too. [18:59:03 CET] <kepstin> If you're starting on an lts release like 18.04 it might need extra options to update to a non lts release [19:00:07 CET] <chaosblade> iirc i had to just apt update/upgrade before it would let me jump to 19.04 [19:00:16 CET] <chaosblade> the message says that when you try anyway, so thats friendly enough [19:00:40 CET] <chaosblade> you also need to edit a file somewhere between LTS and non-LTS upgradability, message points at that as well [21:44:22 CET] <chaosblade> hm, looks like my issue is very much like this https://trac.ffmpeg.org/ticket/6431 [21:44:44 CET] <chaosblade> only neither of the first two workarounds seem to work these days [21:44:54 CET] <chaosblade> maybe the way params flow to gcc changed a bit since [21:46:57 CET] <cehoyos> Please paste a backtrace [21:47:04 CET] <chaosblade> just about to [21:58:46 CET] <chaosblade> ah damn, didn't keep the last ffmpeg_g binary, ill rebuild and see [22:12:42 CET] <chaosblade> i'll have the rest up in a moment, but this is the last bit : #0 0x00007fffd35d01c1 in __GI__dl_allocate_tls_init (result=0x7fffd2d90700) at ../elf/dl-tls.c:465 [22:13:04 CET] <chaosblade> so seems like the known issue of static linking glibc breaking things? im not entirely sure [22:23:44 CET] <cehoyos> You could test with musl which is better suited for static linking and faster [22:29:50 CET] <chaosblade> thats an option, yeah. no experience with that toolchain at all though. [00:00:00 CET] --- Thu Jan 23 2020

1 0

ffmpeg-devel.log.20200121
by burek 22 Jan '20

22 Jan '20

[00:00:00 CET] --- Tue Jan 21 2020

1 0

ffmpeg.log.20200121
by burek 22 Jan '20

22 Jan '20

[00:00:27 CET] <cehoyos> SirNapkin1334: See https://ffmpeg.org/download.html "Linux static builds" [00:46:19 CET] <DHE> SirNapkin1334: there's a shell builtin command called `hash` which bash uses to remember where binaries are for future runs. you'll need to flush its cache [01:10:02 CET] <SirNapkin1334> DHE thanks, coming back later and having closed the shell earlier, it now works [01:16:38 CET] <DHE> that also works [01:16:45 CET] <DHE> :) [08:56:01 CET] <void09> anyone here that can help me try to figure out why ffmpeg cuts a video with 1 frame difference given the same -to time , but different -ss time ? [08:56:14 CET] <void09> 1 frame difference at the end, of course [11:50:55 CET] <BabySoda_> WHO HERE? [13:00:29 CET] <cybex1> hi all. I am trying to stream my USB camera to network multicast (udp) using `ffmpeg -i /dev/video4 -s 1280x720 -r 30 -g 60 -f mpegts udp://238.0.0.1:5000` but the quality is absolutely shite. Imagine a 256k modem with interference streaming a 144p video. I am using 5G wifi for this. Any idea why this is happening? [13:01:36 CET] <JEEB> you did not set any encoding parameters and wondering why you're getting something weird? [13:02:00 CET] <JEEB> you only said that you want MPEG-TS and over UDP [13:02:09 CET] <JEEB> and that your GOP an frame rate and video size are set [13:02:23 CET] <JEEB> everything else is some automagic logic, which can be mpeg-2 video with 200kbps :P [13:02:52 CET] <JEEB> you should preferably set the encoder you want to utilize, and the required minimum settings for that [13:03:43 CET] <JEEB> and yea, default video encoder for MPEG-TS is MPEG-2 Video and that most likely defaults to 200kbps [13:03:46 CET] <JEEB> :) [13:03:51 CET] <cybex1> Hi JEEB. Let me explain. I have a Intel RealSense D435 USB camera, and I want to get the RGB feed broadcasted over the network. To do this, I have a Raspberry Pi 4 with 5G capability. [13:03:59 CET] <JEEB> default audio encoder is mp2 audio [13:04:04 CET] <cybex1> The raspberry pi has hw encoding ability [13:04:15 CET] <cybex1> I thus used the following to recompile ffmpeg: [13:04:31 CET] <cybex1> but in short, I added OMX and enabled MMAL in my configuration [13:04:46 CET] <furq> you need to actually specify that you want to use mmal and omx [13:04:50 CET] <cybex1> ffmpeg setup script: https://pastebin.com/ci3KVqRc [13:04:58 CET] <cybex1> yeah, so I tried omx and mmal [13:04:58 CET] <JEEB> in general: don't let FFmpeg pick your encoder and at the very least know what parameters your encoder needs at the very least [13:05:03 CET] <cybex1> the problem is: [13:05:15 CET] <JEEB> -c:v encoder_name , and the parameters needed depend on the encoder [13:05:22 CET] <JEEB> -c:a for audio [13:05:39 CET] <cybex1> I kept getting this error: [13:05:39 CET] <cybex1> [video4linux2,v4l2 @ 0x1be1210] Cannot find a proper format for codec 'h264' (id 27), pixel format 'yuv420p' (id 0) Assertion *codec_id != AV_CODEC_ID_NONE failed at src/libavdevice/v4l2.c:811 [13:05:41 CET] <cybex1> when using: [13:05:48 CET] <cybex1> ffmpeg -f v4l2 -c:v h264_mmal -pix_fmt yuv420p -i /dev/video4 -s 1280x720 -r 30 -c:v h264_omx -b:v 4000k -f mpegts udp://0.0.0.0:5000 [13:05:56 CET] <cybex1> testing with local streaming only^ [13:06:09 CET] <cybex1> this is where I described my problem [13:06:09 CET] <cybex1> https://superuser.com/questions/1518680/stream-intel-realsense-camera-feed-… [13:06:10 CET] <furq> does your webcam support h264 [13:06:23 CET] <cybex1> uhm... [13:06:29 CET] <cybex1> *how do I check that* [13:06:31 CET] <furq> v4l2-ctl --list-formats [13:06:32 CET] <cybex1> xD [13:06:39 CET] <cybex1> oh right [13:06:42 CET] <cybex1> sec [13:07:02 CET] <cybex1> my video feed: v4l2-ctl --list-formats -d /dev/video4 [13:07:06 CET] <cybex1> ioctl: VIDIOC_ENUM_FMT [13:07:06 CET] <cybex1> Type: Video Capture [13:07:06 CET] <cybex1> [0]: 'YUYV' (YUYV 4:2:2) [13:07:06 CET] <cybex1> [1]: '' (36315752-1a66-a242-9065-d01814a) [13:07:28 CET] <cybex1> so my pix_fmt I set to yuv420p which seemed to work with libx264 sw encoder [13:07:36 CET] <cybex1> but got my CPU up to 100% usage [13:08:51 CET] <JEEB> because the default is preset medium and you're on an ARM box, yes [13:08:59 CET] <cybex1> yup [13:09:00 CET] <cybex1> armhf [13:09:06 CET] <furq> not sure why that list is so truncated [13:09:15 CET] <furq> try ffmpeg -f v4l2 -list_formats all -i /dev/video4 [13:09:20 CET] <furq> (also use a pastebin) [13:09:27 CET] <cybex1> oh [13:09:28 CET] <cybex1> sec [13:09:32 CET] <cybex1> getting pastebin redy [13:11:05 CET] <cybex1> well that is quite alot [13:11:05 CET] <cybex1> https://pastebin.com/aM8h2BDj [13:11:58 CET] <cybex1> also, not sure if this will help at all [13:11:58 CET] <cybex1> pi@raspberrypi:~ $ v4l2-ctl --list-devices [13:11:58 CET] <cybex1> bcm2835-codec-decode (platform:bcm2835-codec): [13:11:58 CET] <cybex1> /dev/video10 [13:11:58 CET] <cybex1> /dev/video11 [13:11:58 CET] <cybex1> /dev/video12 [13:11:59 CET] <cybex1> Intel(R) RealSense(TM) Depth Ca (usb-0000:01:00.0-2): [13:11:59 CET] <cybex1> /dev/video0 [13:12:00 CET] <cybex1> /dev/video1 [13:12:00 CET] <cybex1> /dev/video2 [13:12:01 CET] <cybex1> /dev/video3 [13:12:01 CET] <cybex1> /dev/video4 [13:12:02 CET] <cybex1> /dev/video5 [13:12:14 CET] <cybex1> (for clarity) [13:12:54 CET] <furq> not sure what's going on with "unsupported" but i've never really used v4l2 much [13:13:05 CET] <cybex1> :| [13:13:11 CET] <furq> either way -f v4l2 -input_format yuyv422 -i /dev/video4 [13:13:14 CET] <cybex1> thanks for the help though :) [13:13:16 CET] <furq> and then the output args as you had before [13:13:21 CET] <furq> should work ok on a pi 4 [13:13:28 CET] <furq> assuming this webcam does usb3 [13:13:33 CET] <cybex1> yup [13:13:41 CET] <cybex1> has usb3 (usb type c calbe) [13:15:47 CET] <cybex1> well, when using ffplay udp://238.0.0.1:5000 (streaming address), I get alot of errors, not sure if this means something but I see alot of `ac-tex damaged at`, `Warning MVs not available` and `invalid cbp -4 at` [13:16:03 CET] <cybex1> google doesn't provide that much assistance, thoughts on this? [13:16:46 CET] <cybex1> pastebin of ffplay: https://pastebin.com/tNYvmh7W [13:16:50 CET] <JEEB> just means that the decoder cannot properly decode it right away. if that says, then you might have issues [13:16:58 CET] <JEEB> but if it is until you get a rnadom access picture, then it's OK [13:17:09 CET] <furq> Stream #0:0[0x100]: Video: mpeg2video (4:2:2) ([2][0][0][0] / 0x0002) [13:17:10 CET] <furq> uh [13:17:14 CET] <JEEB> just the decoder complaining that before you get a random access picture, it cannot properly decode the images (d'uh) [13:17:16 CET] <cybex1> I get a picture, but then I get a helluva log of them [13:17:32 CET] <JEEB> furq: means he didn't specify an encoder :) [13:17:40 CET] <JEEB> since mpeg2video is the default for mpeg-ts [13:17:42 CET] <furq> yeah i meant the output args with h264_omx [13:18:03 CET] <furq> and add -pix_fmt yuv420p as an output option [13:18:50 CET] <cybex1> so, I should try -c:v h264_omx (after inpuit file option for encoder setting) and add -pix_fmt yuv420p after the encoder setting? [13:19:35 CET] <cybex1> e.g. ffmpeg -f v4l2 -i /dev/video4 -c:v h264_omx -pix_fmt yuv420p -s 1280x720 -r 30 -g 60 -f mpegts udp://238.0.0.1:5000 ? [13:19:38 CET] <cybex1> that look good? [13:19:55 CET] <furq> yeah [13:20:04 CET] <cybex1> well, trying that give a warning (in yellow) : [mpegts @ 0x1f22e20] Non-monotonous DTS in output stream 0:0; previous: 0, current: 0; changing to 1. This may result in incorrect timestamps in the output file. [13:20:09 CET] <cybex1> not sure if that is good or bad [13:20:15 CET] <furq> you probably also want -input_format yuyv422 -video_size 1280x720 as input options [13:20:40 CET] <furq> you can normally ignore non-monotonous dts errors [13:20:52 CET] <furq> unless the output is actually broken obviously [13:21:04 CET] <cybex1> trying that, seems like ffplay doesn't like that: [13:21:18 CET] <cybex1> using: ffmpeg -f v4l2 -input_format yuyv422 -video_size 1280x720 -i /dev/video4 -c:v h264_omx -pix_fmt yuv420p -s 1280x720 -r 30 -g 60 -f mpegts udp://238.0.0.1:5000 [13:21:24 CET] <cybex1> output of ffplay: [13:22:03 CET] <cybex1> https://pastebin.com/XWjB0PEx [13:22:11 CET] <cybex1> see the middle of the output, some warning: [13:22:31 CET] <cybex1> Could not find codec parameters for stream 0 (Video: h264 ([27][0][0][0] / 0x001B), none): unspecified sizeConsider increasing the value for the 'analyzeduration' and 'probesize' options [13:22:41 CET] <cybex1> should I change the analyzeduration and probesize? [13:33:34 CET] <squ> what is omx? [13:34:10 CET] <cybex1> openmax libraries used for accessing hardware encoding ability of devices [13:34:16 CET] <squ> thanks [13:34:24 CET] <cybex1> Want to use OMB-RPI for my raspbery pi 4 [13:34:27 CET] <cybex1> but having trouble with that [13:36:55 CET] <funnybunny2> I'm using avformat_open_input(&format_context, argv[1], NULL, 0) to read a file from disk. How do I read from a buffer in memory instead? [13:37:32 CET] <funnybunny2> (Assuming I have char *file_contents containing the file contents) [13:39:28 CET] <funnybunny2> The Doxygen search is not very good. [13:41:43 CET] <funnybunny2> Is there just like a function that takes a pointer to a buffer and size instead? [13:52:01 CET] <c_14> funnybunny2: https://ffmpeg.org/faq.html#I-have-a-file-in-memory-_002f-a-API-different-f… [13:52:17 CET] <funnybunny2> I hope this is the current way: https://ffmpeg.org/doxygen/trunk/avio_reading_8c-example.html#a22 [13:54:37 CET] <funnybunny2> c_14: Thanks [13:54:58 CET] <funnybunny2> Going through the ffmpeg docs, it's not very clear sometimes what is an internal function and what is an API function [14:01:52 CET] <funnybunny2> Why is there an av_malloc when there is already malloc? [14:02:26 CET] <funnybunny2> Why is there an av_file_map when there are already functions like fgets and read? [14:03:04 CET] <funnybunny2> I guess these functions are meant to be internal, but the examples use them, so it's confusing [14:08:40 CET] <c_14> funnybunny2: av_malloc handles alignment afair [14:08:57 CET] <c_14> funnybunny2: everything with an av_ prefix is public [14:13:44 CET] <funnybunny2> c_14: OK, that's good to know. Thanks [16:29:42 CET] <void09> anyone here that can help me try to figure out why ffmpeg cuts a video with 1 frame difference given the same -to time , but different -ss time ? [16:29:43 CET] <void09> 1 frame difference at the end, of course [16:39:10 CET] <JEEB> void09: look at result of -debug_ts which outputs all packets it handles? if your ss is different it might cause differing seeks? should be visible in the debug_ts output [16:40:01 CET] <void09> well shouldn't be frame accurate if operating on a properly indexed mkv ? [16:40:38 CET] <void09> it's only one frame off, but i found it happening two times in 3 minutes (in 2 scenes in the video) [16:41:34 CET] <JEEB> depends on a whole lot of things so look at the demux+ffmpeg etc lines' differences in the output [16:41:50 CET] <JEEB> also are you sure -tois not from the start pooint? [16:42:20 CET] <JEEB> not that I know, you would have to check how ut us implemwnted [16:42:31 CET] <void09> -ss 00:01:59.619 -to 00:02:16.845 [16:42:31 CET] <void09> -ss 00:02:16.845 -to 00:02:23.643 [16:42:56 CET] <void09> i have no idea myself , how it's implemented. [16:43:12 CET] <JEEB> i don't have your files, i don't know the rest of your stuff and i am on a train :p [16:43:23 CET] <void09> trying the debug thingie now [16:43:31 CET] <JEEB> that is why i tell yputo look at diffwrences in debug_ts [16:43:58 CET] <nicolas17> if -to was from the starting point set in -ss, then there would be no difference between -to and -t [16:45:01 CET] <void09> how to redirect ffmpeg console output to file please ? [16:45:12 CET] <void09> oh sorry nvm, you're on a train [16:45:24 CET] <void09> phone typing sucks :) [16:45:43 CET] <JEEB> yes, ut does [16:48:42 CET] <void09> ok, got the logs.. now what [16:56:34 CET] <friki> Hi, "ffmpeg ... -n -f tee a.mp4|b.mp4" does't seems to respect -n param. And efective overwrite ocurrs. [17:07:24 CET] <cehoyos> I created ticket 8492 [17:09:30 CET] <friki> cehoyos: thanks! [17:13:05 CET] <friki> cehoyos: may be -n should be added at the ticket [17:29:38 CET] <cehoyos> The option "-n" is an option of the cli, ffmpeg. The tee muxer doesn't know about the cli, the question is if the current behaviour (forced overwrite) makes sense or if a new option should be created (not for ffmpeg but for the tee muxer) that allows overwriting [17:35:58 CET] <friki> cehoyos: OK. In case of new "do not overwrite" option to 'tee' muxer it should be "linked" with -n option at the cli :-p [17:36:27 CET] <friki> default is overwrite at cli, so current tee behavour is good for defaults [17:40:07 CET] <nixonix> when i downloaded some videos with a browser addon (they showed as HLS, but they were pre-recorded, not live stream), the result was shown as h264/aac videos, but the sound was choppy every 4-5 seconds (i used VLC to watch) [17:40:31 CET] <nixonix> is it possible to get rid of the choppiness by using ffmpeg somehow? or any other solutions? [17:44:33 CET] <waleee-cl> nixonix: if you can, try youtube-dl and try different format strings. Eg: youtube-dl -f audio[ext==opus]+bestvideo URL_HERE [17:45:43 CET] <waleee-cl> * reservations for misstyped or miss-remembered syntax [17:46:28 CET] <nixonix> its not youtube, i used video downloadhelper that gets them fine otherwise, but with those sound choppiness. (i tried VDH's conversion options too, but for sound only couple of choises of the long list - didnt help) [17:47:30 CET] <waleee-cl> nixonix: youtube-dl is a program that works on far, far more than just youtube [17:48:04 CET] <waleee-cl> nixonix: and it allows for more control over the downloaded formats [17:49:31 CET] <nixonix> ok. still interested if those already downloaded files could be fixed afterwards, because there is no guarantee youtube-dl will work, and there might be endless list of different choises to try (like there is with conversion options if VDH) [17:49:54 CET] <nixonix> ..of VDH)* [17:52:53 CET] <nixonix> i noticed that converting them to mp4 straight in VDH (those videos then showed as mpeg4 / aac in ffprobe), the choppiness reduced a bit, but it was still there [17:53:04 CET] <nixonix> i suspect some kind of sync problem, but not sure [17:54:08 CET] <nixonix> and if it was a sync problem, there might be a solution to fix them afterwards, that would be a perfect solution [18:12:26 CET] <nixonix> i made more tests. first, just downloading with VDH produced sound chops couple times a second (i remembered it wrong above), but when i converted them straight with VDH's convert options either to re-encoded h264/aac or to mp4, those sound chops only appeared every 4 or 5 seconds [18:13:48 CET] <nixonix> without re-encoding, the result was also h264/aac, but there is option to re-encode it to the same format, and somehow that reduces the chops, but doesnt totally remove them (the result file only decreases by 20% for some reason, probably bit rate reduced) [18:15:27 CET] <nixonix> converting them in VDH to avi (result is mpeg4/mp3 according to ffprobe) removes the sound problems, but the video quality reduces [18:16:23 CET] <nixonix> now i tried to separate audio track from downloaded (choppy sound) videos - and it produced mp3 sound files with no chops [18:17:32 CET] <nixonix> the next i will try to produce a vide file without sound, then create a file with non-problematic sound file and the video-only file together, and see how it goes [18:18:17 CET] <nixonix> or just use ffmpeg to convert those straight, that should do the same, if i find the right way to use it... [18:49:57 CET] <chaosblade> Hi all. I've been following the generic, ubuntu and nvidia guides to get a (static) ffmpeg binary with nvenc enabled. I'm on an Azure VM starting from a Ubunutu Data Science (fancy name for 18.04 LTS with 440.30 nvidia drivers and CUDA) image. No matter what i do, --enable-libnpp causes "libnpp not found using pkg-config". [18:50:11 CET] <chaosblade> I'm suspecting a mismatch somewhere that im not picking up; Was anyone successful with that recently? [18:51:24 CET] <BtbN> chaosblade, does nvidia even offer a static libnpp? [18:51:42 CET] <chaosblade> That's why im putting static as an optional part here, i'll start with a dynamic one that builds and works [18:52:01 CET] <chaosblade> i dont mind that part staying dynamic and requiring the CUDA toolkit/sdk as long as the rest of ffmpeg stays static [18:52:08 CET] <BtbN> libnpp is also not part of just the driver. It needs the full CUDA SDK [18:52:28 CET] <BtbN> And as long as libnpp is on the linker path, it will "just work" [18:52:34 CET] <chaosblade> indeed, i followed nvidia's ffmpeg compile guide and installed the CUDA sdk as they supposedly say how to [18:52:51 CET] <BtbN> Well, libnpp is in fact the only remaining thing needing the CUDA sdk [18:55:07 CET] <BtbN> I'd say just use scale_cuda and call it a day. [18:55:33 CET] <chaosblade> Probably. My main use case is just the nvenc parts, h264_nvenc etc [18:55:38 CET] <chaosblade> so i can probably get by with that for now [18:56:11 CET] <chaosblade> i also failed to get (--enable-)libass to work, but that might be the same reason of a broken/missing linker path [18:56:30 CET] <chaosblade> that was an afterthought though, just because some of the other static builds ive been using include it, i don't do .ass subtitles anyway [19:01:14 CET] <chaosblade> Anyhow, if anyone managed to get nvidia hwaccel'd ffmpeg building and running lately with libnpp, do let me know, please. [19:01:48 CET] <BtbN> I can't see any reason why it wouldn't work. [19:01:59 CET] <BtbN> Check the config log and see what fails, and fix the issue. [19:02:16 CET] <chaosblade> Well, I'm suspecting something moved and their instructions mismatch with the actual versions on offer. [19:02:39 CET] <BtbN> Don't suspect, check the log [19:04:14 CET] <chaosblade> I did, gcc/linker not finding the three npp* libraries despite having an include and lib param that contains the right path to them. Leaves me a little confused. [19:07:43 CET] <nixonix> videos downloaded from that site produce files that ffprobe say are h264/aac, but it seems they are not quite standard or correctly encoded (audio chops couple of times a second). when i re-encode them with ffmpeg to the same h264/aac, the results is less audio chops but there are still those, and ffmpeg gives errors like this: [19:07:49 CET] <nixonix> "Number of bands (52) exceeds limit (42)." "decode_band_types: Input buffer exhausted before END element found" "Multiple frames in a packet." "invalid band type" "channel element 0.0 is not allocated" "Scalefactor (-1) out of range." "Prediction is not allowed in AAC-LC." "Scalefactor (-5) out of range." [19:09:57 CET] <nixonix> video works then, but still audio is choppy. now only every 4 or 5 seconds. any way to adjust the parameters given to ffmpeg so that it would produce a result with no audio chops (i gave the audio and video bitrates that ffprobe told me)? [19:13:28 CET] <chaosblade> without npp it builds and segfaults on using h264_nvenc, hm. more investigation required. [19:13:35 CET] <BtbN> chaosblade, what's your configure line? [19:14:26 CET] <chaosblade> currently, "configuration: --prefix=/home/USERNAME/ffmpeg_build --extra-cflags='-I/home/USERNAME/ffmpeg_build/include -I/usr/local/cuda/include' --extra-ldflags='-L/home/USERNAME/ffmpeg_build/lib -L/usr/local/cuda/lib64' --extra-ldexeflags=-static --pkg-config-flags=--static --bindir=/home/USERNAME/bin --disable-shared --enable-static --enable-gpl --enable-nonfree --enable-version3 --enable-libfreetype --enable-libmp3lame [19:14:26 CET] <chaosblade> --enable-libfdk-aac --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-cuda-nvcc --enable-cuvid --enable-nvenc" [19:14:58 CET] <chaosblade> the cuda pathes are symlinks to cuda-10.2 in the same location, made by the cuda package installer nvidia points you at [19:16:00 CET] <chaosblade> i need to check if my forcing of statics here is breaking something, it probably is. [19:16:16 CET] <chaosblade> i.e. drop disable-shared first [19:16:55 CET] <BtbN> The -static extra flag might prevent it from linking to shared libs [19:18:01 CET] <BtbN> --disable-shared --enable-static has no effect on linking [19:18:10 CET] <BtbN> it's the default even [19:18:43 CET] <chaosblade> yeah, i think that was a copy-pasta error in my experiments from the actual static builds of other libs (where no -dev/-devel package with .a is provided) [19:18:56 CET] <chaosblade> i seem to recall getting messages about those being ignored [19:21:53 CET] <chaosblade> BtbN, yeah, that was it. now i have a lot of dynamic deps for vorbis, theora, etc tho. But that's progress at least, thanks. [19:26:06 CET] <chaosblade> also, without --disable-shared, --enable-libnpp works as well [19:26:21 CET] <chaosblade> which makes sense [19:44:11 CET] <c_14> chaosblade: yeah, adding -static to the link flags disables dynamic linking. And if you want to link other deps statically except for cuda you'll have to edit the configure/pkg-config files to surround the deps with -Wl,-static -Wl,-no-static (or something like that) [19:44:20 CET] <c_14> it's been a while since I last touched that [19:51:43 CET] <cehoyos> I always (temporarily) remove the so files if I need static linking [19:52:00 CET] <cehoyos> the so symlinks... [20:09:20 CET] <LFSVeteran> I modified transcode_aac and now I get "Could not allocate output frame samples" (error 'Invalid argument') [20:09:21 CET] <LFSVeteran> https://pastebin.com/zckjEhLw [20:09:30 CET] <LFSVeteran> only the differences: https://pastebin.com/mWvGBrJE [20:49:07 CET] <nixonix> i got the sound work without chops by extracting the audio, then reattaching the extracted audio (using ffmpeg with both operations). but using either mp3 or aac for audio extract, the result was not not in sync, but audio was around 4-5% faster than video [20:49:41 CET] <nixonix> so i should now try to get them combined with ffmpeg so they would be in sync, i wonder how... [20:54:34 CET] <nixonix> there seems to be "-async" for that. i was looking for something to "stretch" the audio to the same time lenght than the video automagically, though... [21:01:18 CET] <durandal_1707> nixonix: atempo filter [21:10:08 CET] <nixonix> it seems it has changed, used to be -async, then i guess its atempo now. but it requires to calculate the length first, any way to make it automatically, if the start points are correct, so that it would either shrink or stretch the sound length to the same than the video? [21:10:52 CET] <durandal_1707> nope, you need to calculate factor by hand [21:11:16 CET] <nixonix> yeah. or by a script, but lets try by hand if it works [21:34:05 CET] <nixonix> seems like the speed difference is around 24/25 fps or vice versa, like i have used to fix when resyncing subtitles... in fact the other one was not exactly 24 but slightly less, if i just could remember the exact number... [21:35:03 CET] <nixonix> ah, probably that 23.976 (always wondered why not 24 but slightly under) [21:43:51 CET] <klaxa> nixonix: https://www.youtube.com/watch?v=3GJUM6pCpew i think that's a good explanation [21:44:18 CET] <klaxa> although it talks about 29.97 fps, this relates to the 23.976 [21:46:07 CET] <klaxa> the 23.976 is related to this: https://en.wikipedia.org/wiki/Three-two_pull_down [21:49:42 CET] <nixonix> thanks, i checked the wiki fast, but will read more in near future and put the youtube link in my notes [22:20:00 CET] <cehoyos> nixonix: The correct framerate is 3000/1001 [22:20:07 CET] <cehoyos> Or 24000/1001 [22:28:19 CET] <nixonix> ok. btw, resync worked. for some reason, although i extracted both mp3 and aac audio, then ended up using the aac as in my combining, the needed factor to get the lip sync was very close (but not exactly) to the result i got when calculated the length of mp3/video (not the aac, although i used the aac audio file). i wonder why [23:45:22 CET] <chaosblade> c_14, i'll look into that, much appreciated :] [23:45:39 CET] <chaosblade> also cehoyos :] [23:46:14 CET] <chaosblade> bit more of a nightmare than i imagined, getting an ffmpeg built "properly" when you don't have extensive experience with building under linux/gcc [00:00:00 CET] --- Wed Jan 22 2020

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