November 2018 - Ffmpeg-devel-irc

ffmpeg-devel.log.20181109
by burek 10 Nov '18

10 Nov '18

[11:41:42 CET] <cone-790> ffmpeg 03Paul B Mahol 07master:16d91b7e99cd: avfilter/af_afir: kill useless code which increases latency for no reason [13:43:36 CET] <atomnuker> oh hey my server's time which my client is on and my system time match again [13:45:44 CET] <atomnuker> fuck dst, it shouldn't have existed at all, or should have varied with latitude, sunlight at 03:00 is torture [14:02:48 CET] <iive> the whole world should move to UTC [14:10:25 CET] <atomnuker> yep [17:05:00 CET] <philipl> BtbN: the glitching is baffling [18:53:16 CET] <cone-790> ffmpeg 03Paul B Mahol 07master:29fdaaa80d7d: avfilter/af_afir: fix hang because video is first requested [19:35:25 CET] <JohnsonAskot> When using the "gdigrab" format, is it possible to set a specific codec programmatically? I've tried using av_set_dict "vcodec" when passing the options to avformat_open_input, but that doesn't seem to work. [19:41:56 CET] <kepstin> JohnsonAskot: this sounds like more of a user problem, not ffmpeg development. [19:42:14 CET] <kepstin> JohnsonAskot: anyways, gdigrab gets bitmap images from windows apis. whatever you do with them after that is up to you [19:42:38 CET] <kepstin> JohnsonAskot: you'd probably want to decode those, use swscale to convert to a format, then encode with a codec of your choice. [19:46:52 CET] <JohnsonAskot> My bad. [20:07:22 CET] <durandal_1707> i have no ideas what to do next, so... [20:10:52 CET] <atomnuker> new lavfi api [20:17:07 CET] <kierank> durandal_1707: build statue of carl [20:17:11 CET] <kierank> like in india they do [20:26:33 CET] <durandal_1707> something related with audio... [20:29:52 CET] <atomnuker> write a silk encoder? [20:30:51 CET] <durandal_1707> why? [20:32:34 CET] <atomnuker> for fun if you're into slow linear filters [20:36:19 CET] <durandal_1707> better not [20:38:47 CET] <atomnuker> yeah, better not, if you know simd just doing completely unsimdable code is enough to get on your nerves [21:56:05 CET] <cone-790> ffmpeg 03Martin Vignali 07master:75625c555cbe: avfilter/af_headphone : fix mem leak [00:00:00 CET] --- Sat Nov 10 2018

1 0

ffmpeg.log.20181109
by burek 10 Nov '18

10 Nov '18

[00:08:35 CET] <fella> ffmpeg -i MOV.SUF1 -i SND.SUF2 -map 0:0 -map 0:1 -map 1:0 -filter_complex "[0]volume=0.5[some_name];[some_name]amix" OUTPUT.SUF1 [00:31:17 CET] <petecouture> When compiling ffmpeg from source with rtmpdump, do you need the full rtmpdump library or can you just use the included librtmp ver 2.4? [00:33:42 CET] <JEEB> I think librtmp is what FFmpeg utilizes, but check yourself. for the record, there's an internal rtmp implementation as well in libavformat for RTMP(E) [00:37:52 CET] <petecouture> @JEEB is that why the latest compliation build no longer include rtmp dump in them? [00:38:14 CET] <JEEB> I have no idea, there are no official compiles [00:38:25 CET] <JEEB> FFmpeg only provides source code of FFmpeg [00:38:47 CET] <JEEB> I think there are some 3rd party builds linked on the site, but those are 100% under the control of those providing those [00:39:04 CET] <petecouture> I spent the night learning rewriting the librtmp build so it would work with the ffmpeg command line compile. [00:39:16 CET] <furq> i assume librtmp still does something that the internal rtmp stuff doesn't [00:39:18 CET] <JEEB> also the internal support for RTMP(E) is a rather old thing. [00:39:22 CET] <furq> otherwise someone ought to have got rid of it by now [00:39:28 CET] <furq> but nobody has ever been able to tell me what that is [00:39:33 CET] <petecouture> It used to be included but I wondered why it was removed in the latest compliation script ffmpeg.org provides [00:39:46 CET] <JEEB> I don't think we provide compilation scripts? [00:39:54 CET] <JEEB> other than possibly something on the community wiki [00:40:13 CET] <JEEB> furq: mostly what I've seen is that the protocols' AVOptions don't match [00:40:21 CET] <petecouture> Let me pull up the link [00:40:36 CET] <JEEB> so people cannot just use the same command line with ffmpeg.c, for example [00:40:42 CET] <petecouture> https://trac.ffmpeg.org/wiki/CompilationGuide/Centos [00:40:44 CET] <furq> that's a pretty lame reason [00:40:53 CET] <JEEB> yes, that is 100% community based [00:40:55 CET] <petecouture> That guide used to include a librtmp build I believe [00:40:57 CET] <furq> there have been at least two major versions since then [00:41:36 CET] <JEEB> the only other thing is that there's those DMCA'd half-v10 supporting code dumps [00:41:43 CET] <JEEB> but I am not sure how many actually use that [00:42:01 CET] <petecouture> :-/ [00:42:03 CET] <furq> well that's an actual feature so that's ok [00:42:17 CET] <JEEB> it's not in an official rtmpdump release or the repo [00:42:25 CET] <furq> nice [00:42:28 CET] <JEEB> that's what I mean with "how many actually use that" [00:42:44 CET] <JEEB> since most people who build rtmpdump don't generally go searching for some shady repos [00:43:09 CET] <petecouture> I think youll find a bit of people new to ffmpeg that might try [00:43:49 CET] <petecouture> Are there shady repos of it? [00:43:52 CET] <furq> yeah configure could make it clearer that rtmp is builtin anyway [00:43:54 CET] <JEEB> usually no, those who go for that stuff tend to know more or less either what they need or that they need some specific less-official version of rtmpdump [00:44:21 CET] <JEEB> the nr1 reason for wanting to build with librtmp that I've seen is that people know the AVOptions [00:44:24 CET] <JEEB> and guides are made around them [00:44:34 CET] <JEEB> compared to the internal RTMP thing [00:44:48 CET] <furq> based on this channel, the number one reason seems to be that people didn't know about internal rtmp [00:44:52 CET] <petecouture> So building with RTMP is more powerful then using the built in library? [00:44:57 CET] <petecouture> lol [00:45:31 CET] <JEEB> furq: ok, yes. that's the nr1 thing, and then those who try the "native" RTMP support attempt to use the same option names and get stuck or something [00:45:38 CET] <petecouture> I was going to fork the rtmpdump repo and with the changes I made to the build script so it can be compiled in [00:45:45 CET] <furq> petecouture: if you don't know that you need librtmp then you probably don't [00:45:54 CET] <furq> but yeah you might need to rewrite your command [00:46:00 CET] <petecouture> I know enough to be dangerous [00:46:30 CET] <petecouture> Im used to building with it for the raspi arm [00:47:19 CET] <petecouture> Can I pull up the difference between using the librtmp and the internal libraries? [00:47:24 CET] <petecouture> Just to do a compair [00:47:58 CET] <furq> https://ffmpeg.org/ffmpeg-protocols.html#rtmp [00:48:00 CET] <furq> that's the internal one [00:48:13 CET] <furq> i don't think the librtmp stuff is documented on site [00:48:35 CET] <furq> oh nvm it's just not sorted properly [00:48:38 CET] <furq> https://ffmpeg.org/ffmpeg-protocols.html#librtmp-rtmp_002c-rtmpe_002c-rtmps… [00:49:27 CET] <petecouture> @furq thank you! [01:30:43 CET] <orbea> Hi, I am experiencing a crash with ffmpeg while running a single alephone scenario (Game), any ideas if this is a ffmpeg or alephone bug? Or maybe both? https://pastebin.com/YZ1qSnCC I made an alephone issue for now with more details - https://github.com/Aleph-One-Marathon/alephone/issues/121 [01:31:10 CET] <orbea> this is with the ffmpeg master (As of today) [01:41:43 CET] <MarioMey> Hi, there. I have a movie with japanese and english audio streams. I want to add a spanish one, from another file. But I need to recode the audio stream, because by copying it, it doesn't work. Command to copy all streams is: [01:42:23 CET] <MarioMey> ffmpeg -i movie_jap_eng.mkv -i movie_esp.avi -map 0 -map 1:1 -codec copy movie.mkv [01:42:52 CET] <MarioMey> (taken from https://stackoverflow.com/a/44136072/3868690 ) [01:43:12 CET] <MarioMey> How should I encode only the spanish audio stream? [01:46:51 CET] <dreamseeker> can you show us what error you get when you try to copy the Spanish audio stream? [01:47:03 CET] <dreamseeker> like, the full command and output, in a pastebin [01:59:13 CET] <furq> MarioMey: -c:a:0 copy -c:a:1 copy -c:a:2 aac [01:59:15 CET] <furq> or whatever codec you need [02:50:25 CET] <MarioMey> dreamseeker: by encoding, it shows more or less 16 errors like this: [02:50:26 CET] <MarioMey> [matroska @ 0x56026f190960] Starting new cluster due to timestampate=9804.8kbits/s speed=26.4x [02:50:46 CET] <MarioMey> VLC open the file, but audio is unsynch... and shows some other errors. Like: [02:50:54 CET] <MarioMey> [00005623dca7acb0] vlcpulse audio output error: digital pass-through stream connection failure: No soportado [02:51:03 CET] <MarioMey> [00005623dca7acb0] main audio output error: module not functional [02:51:23 CET] <MarioMey> [00007f9200ce04a0] main decoder error: failed to create audio output [02:51:46 CET] <MarioMey> Now, I'm encoding again with what furq said. [02:52:15 CET] <MarioMey> I didn't know that "-c:a:0" way to do it. [08:44:14 CET] <LigH> Hi. [08:45:36 CET] <LigH> Is it possible to let ffmpeg cut n seconds from the end of a clip, without first discovering its length and using it explicitly as parameter? (provided the length can be determined by ffmpeg from header data) [09:01:32 CET] <LigH> I guess --sseof is the one. [09:02:31 CET] <skident> Hi guys, does anybody know is it ok to call function avfilter_register_all (void) from different threads and more than once? [09:03:47 CET] <LigH> Hi ... it's quiet here. America asleep? [09:11:37 CET] <LigH> Bye. [12:50:13 CET] <skident> Let's ask again :) [12:50:14 CET] <skident> does anybody know is it ok to call function avfilter_register_all (void) from different threads and more than once? [12:51:27 CET] <durandal_1707> yes, and note that such function have been deprecated in latest version [12:51:30 CET] <skident> I did some tests an everything work fine, but maybe somebody knows something about it for sure? [12:54:47 CET] <atomnuker> I think they were safe to call before they were deprecated as well given every action was locked [16:17:09 CET] <Pinchiukas> Can I somehow suppress the printing of library information in ffprobe? I can't find an option in the manpage. [16:18:30 CET] <TheAMM> -hide_banner [16:19:48 CET] <TheAMM> -loglevel panic (or such) if you want to hide the rest [16:31:31 CET] <Pinchiukas> -hide_banner doesn't hide it. [16:31:36 CET] <Pinchiukas> It only hides the banner. :) [18:49:27 CET] <ingodid> anyone have a good one-liner for extracting a volume table from an audio file? [18:52:09 CET] <durandal_1707> ingodid: volume what? [18:53:15 CET] <ingodid> a CSV (or STDOUT) table of frames and volumes [18:53:42 CET] <ingodid> i've found something similar for video here "https://stackoverflow.com/questions/38056970/ffmpeg-txt-from-audio-levels" [18:54:04 CET] <ingodid> just starting to wrap my head around the CLI switches [18:55:40 CET] <ChocolateArmpits> ingodid, do you want the volume value for every sample? [18:56:00 CET] <ingodid> wait -- that does seem to work with a WAV [18:56:03 CET] <ingodid> indeed [20:29:53 CET] <ingodid> now onto FFT fun :) [21:19:17 CET] <`St0ner> i'm trying to extract OPUS audio from a YouTube WEBM file losslessly into a format playable by my iPod touch (so either ALAC or WAV). a WAV file is 42.6MB while the ALAC M4A file is 120MB. is it normal for ALAC to be triple the file size of WAV? i thought ALAC is supposed to use lossless compression [21:21:22 CET] <Pocari> `St0ner, how did you go from the Opus embedded in the YouTube video to WAV and ALAC? [21:22:21 CET] <Pocari> what you're saying is not possible... at most, the ALAC file's audio stream will be as big as the uncompressed WAV, only being the same in the case of uniform white noise [21:22:30 CET] <Pocari> so maybe there's something going wrong during the conversion [21:22:52 CET] <`St0ner> for WAV: ffmpeg.exe -i "<SourceFileName>" -acodec pcm_s16le -ar 48000 -ac 2 "<OutputPath><OutputFileName>.wav" [21:23:11 CET] <`St0ner> for ALAC: ffmpeg.exe -i "<SourceFileName>" -acodec alac "<OutputPath><OutputFileName>.m4a" [21:23:44 CET] <`St0ner> ffmpeg build 4.0.2 win64 static dll [21:25:08 CET] <Pocari> `St0ner, check if both WAV and the ALAC, as well as the original Opus, have the same audio hashsum: ffmpeg -i your_file.foo -vn -f md5 - [21:27:13 CET] <Pocari> so, since you're seemingly on Windows, ffmpeg.exe -i your_file.foo -vn -f md5 - [21:33:52 CET] <`St0ner> the MD5= line is different [21:34:03 CET] <Pocari> for all three?? [21:34:41 CET] <`St0ner> sorry didnt run the original. the original and the wav match, the ALAC doesnt [21:35:14 CET] <Pocari> right, that figures... so it's the ALAC indeed that has an issue, as the overbloated filesize indicated [21:36:38 CET] <`St0ner> ok, and is this a problem with the ffmpeg build or should i be feeding non-default parameters into the "acodec alac" section? [21:37:10 CET] <Pocari> `St0ner, try using the WAV then to go to the ALAC; ffmpeg.exe -i track.wav -acodec alac track.m4a [21:39:16 CET] <`St0ner> 31.5MB when i do it that way, which is lower than the WAV file [21:39:23 CET] <`St0ner> so that seemed to have worked [21:39:32 CET] <Pocari> good sign, now to make sure, md5 check that one too [21:40:22 CET] <`St0ner> yes, md5 matches [21:40:42 CET] <Pocari> excellent, so that's a genuine lossless copy... not too sure why the Opus one failed, tbh [21:41:23 CET] <`St0ner> hmm. how can i avoid doing a 2step process from the youtube files? [22:14:27 CET] <ChocolateArmpits> `St0ner, youtube-dl has some encoding options [22:15:01 CET] <ChocolateArmpits> you can download the audio track along with it [22:15:39 CET] <`St0ner> i ended up using AnotherGUI, converted source to WAV, then used the second pass section to convert WAV to ALAC, then used the postprocessing section to delete the WAV. seems to work fine [22:17:01 CET] <saml> how can I write a filter like mandelbrot? [22:17:01 CET] <Pocari> you could have also written a batch file to process the downloaded Opus file with ffmpeg, I guess [22:17:09 CET] <saml> I want my own visualization [22:35:56 CET] <saml> https://www.ffmpeg.org/ffmpeg-filters.html#silenceremove how can I make it also remove video so that there won't be no a-v sync [22:48:39 CET] <threebar> hey everyone, what's the first version of ffmpeg to support 64bit windows builds? [23:03:29 CET] <JEEB> threebar: pretty sure it goes far enough it doesn't really matter [23:06:02 CET] <JEEB> officially change log gained win64 related stuff @ http://git.videolan.org/?p=ffmpeg.git;a=commit;h=61b1f0f909f6adb5e1e32c8b45… [23:06:23 CET] <JEEB> but it clearly seemed to build post-2008? [23:06:52 CET] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=commit;h=dcc01c06650b193c11d3021346… [23:07:01 CET] <JEEB> because this is the first commit that mentions win64 [23:08:25 CET] <kepstin> win64 stuff wasn't really usable until vista or win7 time period, so around 2008 is a place in the timeline that makes sense. [23:12:04 CET] <threebar> so, what would be the earliest usable/compilable version? [23:13:06 CET] <JEEB> decide by yourself whatever hits your criteria http://git.videolan.org/?p=ffmpeg.git&a=search&h=HEAD&st=commit&s=win64 [23:13:17 CET] <JEEB> clearly it was compilable (?) in 2008 already [23:14:04 CET] <JEEB> I know around 2010 people started to want to use it more [23:14:14 CET] <JEEB> which is when the change log entry appeared [23:16:04 CET] <kepstin> win 7 came out in 2009, so that would make sense as being when people started actually caring - upgrades from 32bit xp to 64bit win7. (vista was kinda unpopular) [23:17:13 CET] <kepstin> if you're just going be release tarballs, i wouldn't be surprised if the 0.5 (oldest version still listed on ffmpeg.org/olddownload.html ) would compile, if you can get the right compiler versions. [23:21:43 CET] <JEEB> a snapshot of dec, 2010 by archive.org of fate shows a gcc 4.4.4 x86_64 mingw32 test :P [23:21:54 CET] <JEEB> https://web.archive.org/web/20101221074132/http://fate.ffmpeg.org:80/ [23:22:31 CET] <JEEB> which matches with teh win64 change log part [23:22:45 CET] <JEEB> 727/737 tests pass [23:23:07 CET] <durandal_1707> WOAH [23:26:14 CET] <threebar> nice [23:26:15 CET] <threebar> thank you! [23:27:05 CET] <JEEB> you'd have to grep the IRC logs and the mailing list regarding whenever someone was able to test building for win64 [23:28:09 CET] <threebar> how do i match this to an ffmpeg version number> [23:28:10 CET] <threebar> ? [23:29:14 CET] <JEEB> https://github.com/FFmpeg/FFmpeg/commit/61b1f0f909f6adb5e1e32c8b45a692f67a5… [23:29:18 CET] <JEEB> that's the change log entry [23:29:26 CET] <JEEB> only linking on github because it shows a list of branches [23:29:31 CET] <JEEB> in which the commit is contained [23:30:05 CET] <JEEB> 0.7.1 by the looks of it? [23:30:14 CET] <JEEB> which was released like half a year later [23:30:26 CET] <JEEB> unless those releases are not in the git repo [23:41:17 CET] <threebar> hmm [23:41:44 CET] <threebar> 0.7.1's ./configure says "--target-os=win64" is not a valid option for target-os [23:42:05 CET] <JEEB> not surprising [23:42:50 CET] <JEEB> whatever was used back then with all the configure parameters :P [23:43:02 CET] <JEEB> I don't even know why you're attempting to build a version from 2011 in 2018 [23:43:20 CET] <threebar> getting old code to run [23:44:11 CET] <JEEB> then you should have much better ideas regarding how old something you need is [23:44:14 CET] <JEEB> like SONAMEs [23:47:07 CET] <JEEB> just because it's highly unlikely what I just explained to be what you need :P [00:00:00 CET] --- Sat Nov 10 2018

1 0

ffmpeg-devel.log.20181108
by burek 09 Nov '18

09 Nov '18

[01:33:10 CET] <atomnuker> BBB: so you accepted the ridiculous pr to not adapt but didn't even check its coding style [01:33:44 CET] <atomnuker> honestly, predicted branch or not its utterly ridiculous in case of cdf size one, as in the bool decode function [01:40:38 CET] <BBB> ? [02:49:27 CET] <JC_Yang> questions: should the rtsp client handle/workaround buggy rtsp server implementations? if there're known buggy server implementations, should we add workaround in the client codes? or we should report the bugs to the devices vendor and urge them to fix their own shits? how about if the latter path does not work? [03:39:27 CET] <Compn> JC_Yang : yes [03:39:38 CET] <Compn> as long as you document the buggy ness in the code with some good docs [03:43:35 CET] <Compn> both i mean [03:47:50 CET] <Compn> tlol [04:28:37 CET] <JC_Yang> ok, get it [06:01:39 CET] <Lord_Nightmare> Compn: I saw you put a notice about asking you about codecs on the https://wiki.multimedia.cx/index.php/TwoCC page [06:01:54 CET] <Lord_Nightmare> The "Yamaha ADPCM" codec is not listed on that page [06:02:12 CET] <Lord_Nightmare> but it can be found at https://web.archive.org/web/20160323215457/http://smaf-yamaha.com/jp/tools/… [06:04:45 CET] <Lord_Nightmare> its a plain old Jayant ADPCM codec, almost identical to jayant's 1973 paper at https://ia902601.us.archive.org/13/items/bstj52-7-1105/bstj52-7-1105.pdf and his 1974 paper at https://ia902600.us.archive.org/31/items/bstj52-7-1119/bstj52-7-1119.pdf (see page 23) [06:04:57 CET] <Lord_Nightmare> its also used on a bunch of yamaha ADPCM playback chips [06:05:15 CET] <Lord_Nightmare> like the ym2608 and ym2610 [06:06:14 CET] <Compn> I mean you edit the wiki , you can add it :D [06:06:25 CET] <Compn> do we already have yamaha decoder? [06:08:16 CET] <Lord_Nightmare> not sure, i think ffmpeg does have one. if not, how to make one is pretty easy [06:08:38 CET] <Lord_Nightmare> it uses the same coefficients as jayant's 1974 paper does, just multiplied by 256 (iirc) [06:09:27 CET] <Lord_Nightmare> i.e. 0.8 0.8 0.8 0.8 1.2 1.6 2.0 2.4 [06:09:47 CET] <Lord_Nightmare> but twice, since the sign bit doesn't affect the magnitude of the step size multiplier [06:09:57 CET] <Lord_Nightmare> for each 4 bit sample [06:10:21 CET] <Lord_Nightmare> (MSADPCM is weird and has the table laid out differently, plus adds the value '3.0' for the -8 case) [06:10:52 CET] <Lord_Nightmare> (and BSMT2000 adpcm is sort of like MSADPCM i think except they don't add the 3.0 value?) [06:11:14 CET] <Lord_Nightmare> (and they don't have the 2-previous-samples IIR filter either, which MSADPCM has) [06:12:42 CET] <Lord_Nightmare> i'm still trying to figure out how the 'ADPCM2' mode of the oki MSM6295 works [06:14:10 CET] <Lord_Nightmare> afaik dialogic oki adpcm is 100% identical to the MSM5205 except differences with the preset value for the first sample(?) [06:14:27 CET] <Lord_Nightmare> i'm not sure the difference between 0x0010 OKI ADPCM and 0x0017 Dialogic OKI ADPCM [06:14:47 CET] <Lord_Nightmare> oki adpcm is a subset if IMA ADPCM afaik [06:14:51 CET] <Lord_Nightmare> *of [06:15:02 CET] <Compn> yeah i havent a clue [06:15:06 CET] <Compn> i just collect the twocc :D [06:15:18 CET] <Compn> try to find acm files when people ask [06:15:40 CET] <Lord_Nightmare> I was trying to hack together a codec for 0x0005 IBM CVSD IBM Corporation which was later bought by Digispeech [06:15:47 CET] <Compn> and yes ffmpeg has adpcm_yamaha decoder [06:16:24 CET] <Lord_Nightmare> i don't think there was ever a 'real' acm codec for IBM CVSD (4-bit cvsd), i think the real codec was just a driver that passed the speech data along to an isa card and the PS/2 speech adaptor [06:16:44 CET] <Compn> btw yamaha is on there [06:16:44 CET] <Compn> 0x0020 Yamaha ADPCM [06:16:49 CET] <Compn> on the twocc page [06:17:19 CET] <Lord_Nightmare> yes, and theres a Yamaha_ADPCM page as well [06:17:31 CET] <Lord_Nightmare> i didn't even KNOW about the twoCC page until i checked recent edits [06:18:01 CET] <Compn> yeah i wanted to put it in the main nav box [06:18:15 CET] <Compn> but i think mike gave me the runaround when i asked him howto get it there [06:18:58 CET] <Lord_Nightmare> afaik the PS/2 Speech adapter isa card has an MC3418 4-bit cvsd codec chip on it, and an adjustable clock generator for it for 8 fixed samplerates http://arcarc.xmission.com/Tech/Datasheets/MC3418.pdf [06:19:15 CET] <Compn> Lord_Nightmare : theres also this page I maintain https://wiki.multimedia.cx/index.php/Category_talk:Video_FourCCs [06:19:42 CET] <Compn> both used to get a lot of hits from google. but i cant track the hits anymore since the wiki upgrade took out the "popular pages" and hit counts out :( [06:19:52 CET] <Lord_Nightmare> the CVSD typically used on arcade games and pinballs though is the MC3417 3-bit cvsd codec chip, and its harris clones the HC55516, HC55532 and HC55564, and the decode-only HC55536 [06:20:08 CET] <Compn> i just sold my pinball games :( [06:20:09 CET] <Compn> ehe [06:20:39 CET] <Lord_Nightmare> motorola made a later chip called MC34115 which i believe is an updated MC3417 with multiple filters and a bandpass to select which one is active, like the HC55564 has [06:20:59 CET] <Lord_Nightmare> the hc55516 filters are tuned for 16khz, and hc55532 is tuned for 32khz [06:21:19 CET] <Lord_Nightmare> hc55564 uses some cute internal circuit with an analog bandpass to select one of two sets of filters, or so i believe [06:21:28 CET] <Lord_Nightmare> depending on how fast the clock in is [06:21:44 CET] <Lord_Nightmare> sorry for info dump [06:22:40 CET] <Lord_Nightmare> jeri ellisworth made an fpga reimplementation of the 3-bit cvsd chip to fix one of her pinballs, but i believe she forgot one of the analog filters when doing so. it still seems to sound ok, though, surprisingly... [06:23:33 CET] <Lord_Nightmare> https://www.youtube.com/watch?v=2FRGwuxFDE4 [06:23:42 CET] <Compn> have the pinmame people done some of these codecs ? [06:24:05 CET] <Lord_Nightmare> MAME has an hc55536 implementation used on exidy's mouse trap arcade game [06:24:19 CET] <Lord_Nightmare> but again i'm not sure how accurate it is [06:24:34 CET] <Lord_Nightmare> its also used on some williams games i believe [06:25:19 CET] <Compn> hah i just sold my getaway [06:25:32 CET] <Compn> williams game [06:25:35 CET] <Compn> and my taxi [06:25:43 CET] <Compn> nother williams [06:26:24 CET] <Lord_Nightmare> a real cvsd chip has two analog filters: one controls the rise and fall of the slope magnitude value if it received 3 or 4 bits of the same value for the most recent 3 or 4 bits [06:26:58 CET] <Lord_Nightmare> and another filters this magnitude value as a lowpass when it is applied to the comparator op-amp vs the input [06:27:21 CET] <Compn> dont think i'd wire in an fpga though [06:27:21 CET] <Lord_Nightmare> jeri emulated the first of these as if it is perfectly symmetrical gain and loss, and didnt' emulate the second one at all [06:27:35 CET] <Compn> easier to fix the boards , get the roms on new chips and put it back together [06:27:55 CET] <Lord_Nightmare> yeah, i don't think obtaining a NOS or used but working CVSD chip is as hard as jeri made it out to be [06:27:58 CET] <Compn> unless the entire sound board was toast. then... [06:28:01 CET] <Compn> ah [06:28:12 CET] <Lord_Nightmare> I'm going to email her and see if she still has the blown out chip, since i'd love to get a micrograph done of it [06:28:26 CET] <Compn> the chip or the board ? [06:28:34 CET] <Compn> you can just straight out buy new chips [06:29:25 CET] <Compn> probably you mean board :D [06:30:16 CET] <Compn> and when you cannot get boards anymore, just toss in the emulator on an rpi [06:30:27 CET] <Lord_Nightmare> nah she had a working board with just that one chip blown [06:30:51 CET] <Lord_Nightmare> sound worked except the speech was just noise [06:30:56 CET] <Compn> ah [06:31:05 CET] <Compn> then yeah its as easy as getting a new chip [06:33:19 CET] <Compn> or re-programming the old one if its just corrupt [06:33:40 CET] <Lord_Nightmare> you can't reprogram a CVSD chip, its a fixed function mixed-analog-digital ASIC of sorts [06:33:48 CET] <Compn> oh i'm thinking of the sound samples [06:33:49 CET] <Compn> my bad [06:33:52 CET] <Compn> not the chip [06:33:57 CET] <Compn> my brain is off right now [06:34:06 CET] <Lord_Nightmare> in theory you MIGHT be able to replace it with a very fast PIC emulating a CVSD chip [06:34:15 CET] <Lord_Nightmare> and emulating the analog filters digitally [06:35:43 CET] <Compn> sorry [06:35:44 CET] <Compn> ehe [06:37:23 CET] <Compn> proc thats what its called [06:38:00 CET] <Compn> https://www.multimorphic.com/store/circuit-boards/p-roc/ [06:38:44 CET] <Compn> oh that uses the existing boards, i'm completely incorrect about it haha [06:39:13 CET] <Compn> thats pretty neat replacing a sound decoder chip with a fpga though [07:05:43 CET] <JC_Yang> do we internally have any indicator whether AVStream.avg_frame_rate is valid? or any guidelines about when it is usable as a reference? [10:49:52 CET] <BtbN> Scientific build systems sure are fun. [10:50:30 CET] <BtbN> This one expects to be cloned to .../src/github.com/syslabs/singularity. Otherwise it will fail to build. [10:51:00 CET] <BtbN> The only place I learned about that was in a github issue about it. [11:03:14 CET] <nevcairiel> some people just care about working on their code, not making it easy on other people [11:06:23 CET] <BtbN> This is aparently how go is supposed to work. [11:07:01 CET] <BtbN> Even though you don't even see it's Go, cause their self-built mconfig script downloads it in the background [11:07:14 CET] <nevcairiel> thats at least something [13:07:45 CET] <cone-132> ffmpeg 03Paul B Mahol 07master:fcbde5e6a4a5: avfilter/af_afir: remove redundant code [13:07:46 CET] <cone-132> ffmpeg 03Paul B Mahol 07master:8dc5eb43b070: avfilter/af_afir: return early when giving video frame [13:07:47 CET] <cone-132> ffmpeg 03Paul B Mahol 07master:1d9fe1fdf65b: avfilter/af_afir: implement rate option [13:07:48 CET] <cone-132> ffmpeg 03Paul B Mahol 07master:0f0f84071659: avfilter/af_aiir: implement rate option [13:39:05 CET] <cone-132> ffmpeg 03Carl Eugen Hoyos 07master:5ff080fdfa25: lavf/apngdec: Return EOF for incomplete files. [15:21:51 CET] <cone-132> ffmpeg 03Paul B Mahol 07master:09fdfc11b796: avfilter/avf_showspectrum: add cividis colormap [18:00:35 CET] <cone-132> ffmpeg 03Paul B Mahol 07master:70463b702697: avfilter/avf_showspectrum: add terrain color map [18:38:08 CET] <BtbN> philipl, 415 is out [18:44:25 CET] <philipl> BtbN: yes. I'll be trying it out tonight/tomorrow [20:40:28 CET] <durandal_1707> FFmpeg - officially dead [20:43:00 CET] <atomnuker> wouldn't be so dead if companies didn't stop making their own custom codeccs [20:44:56 CET] <durandal_1707> it is dead, no new commits in latest days except mine [20:47:55 CET] <kepstin> is everyone busy on dav1d instead? :) [20:48:13 CET] <atomnuker> yeah [20:48:42 CET] <atomnuker> wish there were some 4k samples to test its speed, I'm getting 220+ fps on 1080p and its so fast its boring [20:49:36 CET] <mmu_man> plop [20:49:37 CET] <mmu_man> anyone seen my metadata patch? [20:50:48 CET] <durandal_1707> this one: [20:50:51 CET] <durandal_1707> [FFmpeg-devel] [PATCH] Optimize libavformat/metadata.c? [20:51:35 CET] <durandal_1707> ah, michael replied to another metadata patch [20:51:56 CET] <mmu_man> ah yes, just saw [20:52:17 CET] <jamrial> atomnuker: elecard has some 4k sampes [20:59:36 CET] <BBB> netflix also, no? [21:00:06 CET] <atomnuker> netflix had chimera, which can curiously fully load my cpu, but a similar resolution and bitrate from youtube can only half load it [21:00:36 CET] <atomnuker> maybe the gop length is different [21:02:09 CET] <BBB> how weird [21:02:14 CET] <BBB> maybe bitrate? [21:02:30 CET] <BBB> with most things simded, bitrate significantly affects cpu load [21:02:35 CET] <BBB> so netflix may be better for your purposes [21:02:41 CET] <BBB> since bitrate/ec-ratio is higher [21:03:14 CET] <BBB> bitrate (= entropy coding-ratio) is higher [21:03:19 CET] <BBB> you know what I mea [21:03:25 CET] <atomnuker> I'm getting the same performance over 300 frames too, 1.4 seconds to decode from both [21:03:36 CET] <atomnuker> its just the load is higher on one than the other [21:03:47 CET] <kurosu__> there also may be some tool that both is more consuming and increase the delay for frame multithreading ? [21:03:51 CET] <atomnuker> yeah, on netflix the ec overhead is higher [21:05:06 CET] <kurosu__> atomnuker, btw, for msac_decode_symbol, pmulhuw exists [21:06:51 CET] <atomnuker> yeah, I know [21:07:12 CET] <atomnuker> the elecard 4k sample kills performance though, barely 44fps or so here [21:10:44 CET] <BBB> oh no, only 44 fps" [21:10:59 CET] <BBB> I used pmulhuw in some of my tests instead of psraw, it did seem to help, yes [21:11:02 CET] <BBB> and prevents stack use [21:11:23 CET] <atomnuker> its the attitude you need to optimize further :) [21:11:33 CET] <BBB> true [21:11:47 CET] <BBB> I think after having spent several months writign the decoder, I want to do encoding work again [21:12:10 CET] <BBB> it does what it needs to do, Im happy to do more work on it but Id also like to sort of sit back and see if people want to use it etc. [21:12:11 CET] <atomnuker> your opening quote mark was converted to some non-ascii one but your closing quote mark is ascii I think [21:12:29 CET] <BBB> "now it's ok"? [21:12:38 CET] <BBB> my irc client is shit [21:13:29 CET] <atomnuker> yeah, its okay now [21:14:02 CET] <atomnuker> could be eviler, it could detect dates and convert them to messed up formats or worse, detect fractions and change the dot to a comma [21:14:38 CET] <atomnuker> and remove spaces/thin spaces for thousands separations and replace them with dots and or commas [21:15:30 CET] <BBB> mine also changed dashes [21:15:38 CET] <BBB> dav1d's CLI doesn't scan for unused arguments [21:15:52 CET] <BBB> so if you copypaste my non-standard dashes on the CLI, it doesn'tknow it's a dash and it ignores the argument [21:15:54 CET] <BBB> remember? :) [21:16:27 CET] <atomnuker> yeah, that took some head scratching to figure out, maybe with a different font they'd be more visible [21:28:48 CET] <atomnuker> elecard's 1080p version is about 115fps here, so its slightly more than 2x slower [21:29:07 CET] <atomnuker> they must be using some bottlenecking bitstream features then [22:10:06 CET] <cone-132> ffmpeg 03Paul B Mahol 07master:698e67bef2ee: avfilter/af_afir: do not leak frame and clear pointer to freed frame [22:10:07 CET] <cone-132> ffmpeg 03Paul B Mahol 07master:36348d77b86a: avfilter/af_afir: add minp/maxp options to control latency and speed [22:19:46 CET] <cone-132> ffmpeg 03Andreas Rheinhardt 07master:8d1cf2d89481: cbs_h264: Fix handling of auxiliary pictures [22:22:20 CET] <cone-132> ffmpeg 03Martin Vignali 07master:105891cd580b: avcodec/prores_aw : change src pix_fmt and alpha description [22:22:21 CET] <cone-132> ffmpeg 03Martin Vignali 07master:4d32c2b4e0df: avcodec/prores_aw : cosmetic indent and add comment to describe frame header value [22:22:22 CET] <cone-132> ffmpeg 03Martin Vignali 07master:ff3b4f91cd56: avcodec/prores_aw : add support for prores 444 with alpha [22:41:31 CET] <cone-132> ffmpeg 03Michael Niedermayer 07master:07bc603757ca: avcodec/qpeg: Optimize full width runs in qpeg_decode_intra() [00:00:00 CET] --- Fri Nov 9 2018

1 0

ffmpeg.log.20181108
by burek 09 Nov '18

09 Nov '18

[11:31:04 CET] <Akien> Hi there! I'm trying to debug a video playback issue in the emulator PPSSPP when built against FFmpeg 3.1+ (upstream uses 3.0.2 for official builds). I have no experience with FFmpeg and I'm not the PPSSPP dev who wrote the video playback code either, but I've done some bisecting and thought maybe some of you know what the current code misses. The [11:31:04 CET] <Akien> issue: https://github.com/hrydgard/ppsspp/issues/11490 [11:31:24 CET] <Akien> TL;DR: Issue triggered by an API change in FFmpeg 3.1 for AVStream: https://github.com/hrydgard/ppsspp/issues/11490#issuecomment-436589320 [11:32:18 CET] <Akien> And this is the code that works well with 3.0.x but produces stuttering in 3.1+: https://github.com/hrydgard/ppsspp/blob/master/Core/HW/MediaEngine.cpp#L245… [11:33:06 CET] <Akien> From what I understand of https://github.com/FFmpeg/FFmpeg/commit/9200514ad8717c63f82101dc394f4378854… I guess the AVStream needs some more configuration now? [11:35:28 CET] <Akien> (This was done in part here, fixing the garbled output, but not the choppiness: https://github.com/hrydgard/ppsspp/commit/7a7c655615a4f607ea2e6bc58dc8ca3e4… ) [11:40:38 CET] <RandomUser8787> hello all. im new to developing with libav libs and im currently having some issues with memory leaks which i dont quite understand. im getting UYVY422 pixel data from a capture device which i copy to a frame. this frame i then run through a filtergraph and then the filtered frame i transfer to the gpu for encoding (nv12). after each frame is encoded i free all frames but im seeing leaks from the filter graph, avcodec_send_frame [11:42:30 CET] <RandomUser8787> i have looked at the ffmpeg filter and vaapi encode code examples but obviously im missing something... [12:58:24 CET] <RandomUser8787> hmm looks like i should have done unref instead of free [13:44:34 CET] <RandomUser8787> or no that didnt really change anything [14:41:10 CET] <zanijwa> Does the precompiled versions of ffmpeg not have https support enabled by default? I'm using youtube-dl and it gave an error 'https protocol not found, recompile FFmpeg with openssl, gnutls or securetransport enabled' [14:41:21 CET] <zanijwa> SO did say to recompile ffmpeg [14:41:59 CET] <zanijwa> So if I do this by myself, should I pass the entire path to my compiled ffmpeg whenever I need to call it? [14:42:34 CET] <zanijwa> Because I prefer automatic updates provided by the Ubuntu package maintainers [14:54:28 CET] <^Neo> good morning friends, is the idea with 4.X now that av_frame_get_best_effort_timestamp / av_frame_get_pkt_duration / av_frame_get_pkt_size is to access the data members directly? [14:56:24 CET] <kepstin> ^Neo: yeah, just access the data members directly. [14:59:14 CET] <ariyasu> zanijwa it does just add it to the whitelist [15:22:10 CET] <santamarta> I am trying to make a long presentation to record in a bluray under linux. It has 21 images and 18 long audios (~1 hr each, but different length each), mostly voice, with some music here and there. The whole audio encoded to .m2ts is ~200M long. The unencoded .jpg images are ~25M. I tried to encode the images to h264, each image is static and most of the audios show only one image, but some of them have two. The result was a 3.7G file. [15:23:07 CET] <santamarta> my question is... There is a way to encode the images in a lower size file that is compatible to dvd or bluray format ? [15:23:25 CET] <durandal_1707> there are some options for still picture [15:23:38 CET] <furq> -tune stillimage [15:23:48 CET] <santamarta> yes, I am using that [15:23:53 CET] <furq> pastebin the full command [15:24:03 CET] <kepstin> the requirements for bluray video specify a fairly short keyframe interval, so the video's gonna be big [15:24:06 CET] <kepstin> no way around that [15:24:09 CET] <furq> how short [15:24:35 CET] <santamarta> it is small: ffmpeg -f concat -safe 0 -i Covers.txt -vf "scale=1920:1080,format=yuv420p" -c:v h264 -tune stillimage portadas2.mp4 [15:25:00 CET] <furq> that's not actually bluray compatible so i'm surprised it turned out that big [15:25:03 CET] <kepstin> I think 1 second gop length [15:25:15 CET] <kepstin> i.e. 30 frames for 30fps [15:26:05 CET] <kepstin> (although many players apparently accept longer values, particularly the ps3 and software players that also do arbitrary h264 high stuff) [15:27:02 CET] <furq> well yeah you need -x264-params bluray-compat=1 [15:27:10 CET] <furq> and maybe also -g 30 but i guess bluray-compat might set that [15:27:13 CET] <furq> or -g 25 rather [15:27:53 CET] <santamarta> thanks, furq [15:27:58 CET] <furq> bear in mind bluray also supports 720p [15:28:06 CET] <furq> so i guess if you care about size then 720p24 is good [15:28:48 CET] <santamarta> I will try... 1080p is important because the images are text, but maybe coud go with more images [15:46:24 CET] <santamarta> could this be solved using a lower framerate ?. Time resolution is not critical at all, a 5 seconds difference causes no problem [15:46:41 CET] <kepstin> I think 24fps is the minimum [15:47:01 CET] <kepstin> but given that the keyframe interval is set based on realtime, that's not likely to make a noticable difference [15:49:30 CET] <kepstin> hmm, it sounds like you can use longer gop when the bitrate is lower. up to 2 seconds if you're under 15mbit/s, which should be true of a video made from mostly still images [15:50:53 CET] <kepstin> if you want the video to fit within a particular size, i'd suggest using a 2-pass bitrate encode with bluray compatible options, and just see if the quality is acceptable [15:51:45 CET] <kepstin> and if you don't actually need blu-ray compatibility, the video can be made a lot smaller :/ [17:01:28 CET] <gh0st3d> Hey everyone... I'm trying to take a video file and create a gif from 1s to 11s that also includes an overlay... I previously got it to include the overlay while creating the gif, but the colors on the overlay were messed up because of the palette thing. For now, I'm making it add the overlay to the video, then creating the palette and the gif [17:01:47 CET] <gh0st3d> However, it definitely was much faster the other way... Is there anyway to generate the palette from 2 files separately? Here's my commands: https://pastebin.com/eRcRY89E [17:02:31 CET] <TheAMM> You'll have to overlay when creating the palette [17:02:50 CET] <TheAMM> You don't have to render the overlay on the video and encode that to a file, you can add the overlay filter before the palettegen [17:03:05 CET] <gh0st3d> Ok, let me try rearranging that and see if I can get it figured out [17:03:05 CET] <TheAMM> The paste isn't loading for me right now so I can't comment on the commands [17:03:43 CET] <TheAMM> but ffmpeg -i in.mkv -vf "overlay=<...>, palettegen" palette.png [17:12:13 CET] <gh0st3d> Oh sweet, got it this time. I thought I tried something similar before but just got a bunch of errors. Is there anyway to compress the gif output with FFMPEG or do you need other software installed to do that? [17:21:27 CET] <kepstin> not sure what you mean by "compress", ffmpeg can generate gif animations directly, but depending on the original content it may be possible to optimize them better manually or with other tools. [19:12:03 CET] <mooniac> is it possible to use two videos of the same length, and make one a smaller inset in one of the corners of the other video? I think that's sometimes called a "mashup"? [19:14:03 CET] <pink_mist> https://www.daniel-mitchell.com/blog/video-in-a-video-ffmpeg/ [19:14:58 CET] <mooniac> woohoo, thanks. [19:15:50 CET] <pink_mist> never tried doing that myself though, so I can't speak to the effectiveness of that blogpost [19:16:14 CET] <mooniac> but it's a good start for me. If it doesn't work, I'll be back here :) [19:22:07 CET] <ChocolateArmpits> mooniac, actually it's called picture in picture "PiP" [19:22:20 CET] <mooniac> thx [19:32:14 CET] <Aerroon> i have a bunch of audio files [19:32:46 CET] <Aerroon> and i'd like to put a static image and stretch it across the entire file to basically make a "video" wher eit plays the audio and shows that one specific image [19:33:04 CET] <Aerroon> what should i do to have the "fastest render time"? [19:33:20 CET] <Aerroon> the normal way of doing it that i figured out i'm basically just rendering out a new video [19:41:29 CET] <pink_mist> no idea, something like: ffmpeg -i pic.png -i audio.mp3 -c:a copy -c:v libx264 -tune stillimage out.mkv perhaps? [19:48:14 CET] <atomnuker> keep in mind that'll make a video [19:48:37 CET] <atomnuker> if you want a single frame you have to encode to a temporary file [19:48:53 CET] <atomnuker> and then use stream copy on that single frame stream along with your audio stream [19:49:28 CET] <atomnuker> some players like vlc won't display it correctly though, but most will, including browsers [19:50:03 CET] <atomnuker> it'll save you a lot of bits though so its worth it [19:50:43 CET] <atomnuker> but if the container supports it you can add the cover art disposition which will make it work with pretty much anything [19:54:27 CET] <Aerroon> i see [19:54:28 CET] <Aerroon> thanks [19:54:45 CET] <Aerroon> the bits bit isn't important for me, i'm more interested in it not taking a long time to encode [19:54:50 CET] <Aerroon> but your solution likely does both [20:37:44 CET] <Aerroon> from tests it seems i'm gonna go with [20:37:47 CET] <Aerroon> ffmpeg -loop 1 -i image.png -i audio.wav -shortest -c:v libx264 -tune stillimage -c:a copy result.mkv [20:38:06 CET] <Aerroon> but this means it'll have to encode it as though it's just a normal video [20:38:20 CET] <Aerroon> any recommendations for a faster video codec? [20:38:38 CET] <Aerroon> ohh, ig uess i could try gpu encoding [20:39:10 CET] <pink_mist> mjpeg? [20:39:56 CET] <Aerroon> that doesn't seem faster [20:42:03 CET] <Aerroon> hmm, it seems the encoding speed isn't even limited by that [20:42:14 CET] <Aerroon> i think it might be the loop 1 argument [20:43:40 CET] <durandal_1707> -loop 1 -- will decode image all over again [20:43:57 CET] <Aerroon> yep, that would be consistent with what i'm getting [20:44:11 CET] <Aerroon> i guess that was the point of the earlier stream copy comment [20:45:06 CET] <mmu_man> plop [20:45:11 CET] <mmu_man> anyone seen my metadata patch? [20:45:34 CET] <durandal_1707> wrong channel for patch talk [20:49:21 CET] <mmu_man> eh [20:59:56 CET] <analogical> I just encoded a sound file to FLAC but the the resulting file become 24-bit although the source file is 16-bit. How do I force FFmpeg to encode to 16-bit FLAC ?? [21:02:34 CET] <dreamseeker> run an ffmpeg -i input_file.flac command and pastebin the output for us [21:02:49 CET] <dreamseeker> er [21:02:57 CET] <dreamseeker> ffmpeg -i input_file.foo I mean [21:03:04 CET] <dreamseeker> whatever extension the input file is [21:09:30 CET] <Aerroon> oof [21:11:07 CET] <Aerroon> speed=1.14e+03x [21:32:10 CET] <Aerroon> >Too many packets buffered for output stream 0:1. [21:32:21 CET] <Aerroon> it's an audio stream copy [21:32:32 CET] <Aerroon> thats 2.5 hours long [21:32:50 CET] <Aerroon> -max_muxing_queue_size 4000 fixes it but i have no idea what that means [21:34:30 CET] <JEEB> Aerroon: ffmpeg.c wasn't for one reason or another capable of initializing the muxer until then [21:35:00 CET] <JEEB> so it makes a queue and sticks the AVPacket there [21:35:38 CET] <JEEB> and the option you set controls the size of that queue [21:35:54 CET] <Aerroon> i see [21:36:21 CET] <JEEB> the better question is why it decides it cannot initialize the muxer (the thing that writes them packets into a container and into a file/stream) [21:49:25 CET] <Aerroon> i don't know either [21:49:43 CET] <Aerroon> because i did demux this audio stream with -c:a copy from another file [21:50:38 CET] <Aerroon> i get the too many packets buffered for output stream 0:1 (which is audio [21:50:42 CET] <Aerroon> from a file i made with [21:50:44 CET] <Aerroon> ffmpeg -stream_loop -1 -i image.mp4 -i audio.m4a -shortest -c:v copy -c:a copy out.mkv [21:51:07 CET] <Aerroon> image.mp4 is a video that consists of one frame [21:51:39 CET] <Aerroon> but the problem appears on the audio stream which demuxed just fine before i put it together with the image [23:04:41 CET] <SneakySnake> Do I really have to do wizardry like this to mix an audio stream with a video while also lowering the original video's volume? https://superuser.com/a/769195 [23:04:55 CET] <SneakySnake> Is there a simpler way? [23:05:11 CET] <SneakySnake> I can't even comprehend what is going on here [23:23:26 CET] <SneakySnake> Okay, I think I figured it out. [00:00:00 CET] --- Fri Nov 9 2018

1 0

ffmpeg-devel.log.20181107
by burek 08 Nov '18

08 Nov '18

[12:00:36 CET] <BtbN> Hm, I wonder if ffmpeg_cuvid.c can just be removed by now. It looks as generic as it gets. [15:41:51 CET] <kwizart> Hello, FYI fsf raised a statement on the fdk-aac license (on Fedora/RH request after a patent disabled fdk-aac package was introduced) [15:41:54 CET] <kwizart> https://www.gnu.org/licenses/license-list.en.html#fdk [15:42:09 CET] <kwizart> in case you have missed it [15:43:03 CET] <kwizart> half related, I've recently found this project https://github.com/yugr/Implib.so [15:43:53 CET] <kwizart> which might allow to dlpen any libraries at runtime (probably once the headers and entries point are added in ffmpeg) [15:44:09 CET] <kwizart> would it be something appropriate for use in ffmpeg ? [15:44:25 CET] <kwizart> (for such cases as fdk-aac decklink or else ?) [15:44:56 CET] <nevcairiel> its questionable this influences the license situation. Some technical mean should not influence how law is interpreted. this is not something ffmpeg s hould consider. if you want to build like that [15:45:18 CET] <nevcairiel> .. you can do that w ithout ffmpeg knowing about it, its supposed to be transparent [15:59:14 CET] <kwizart> yep I think it's just a matter to error as appropriate if end-users request the library but it's not available at runtime [15:59:54 CET] <atomnuker> I wish people would stop trying to find ways to make fdk appear as legal and open source as possible and just use opus [16:00:18 CET] <kwizart> the question is would ffmpeg source tree would be able to have the generated wrapper code or this have to be a downstream use case ? [16:02:00 CET] <gnafu> atomnuker: Hear, hear! [16:03:51 CET] <kwizart> atomnuker, I agree with that too, but the averrage end-user is dumb when it comes to audio codec [16:04:10 CET] <JEEB> also there's the plastic boxen, although the internal AAC encoder is generally speaking OK [16:05:01 CET] <j-b> yes [16:05:09 CET] <j-b> fdk-aac is non-free [16:05:10 CET] <kwizart> atomnuker, and my problem as a distro maintainer is that some scammer repo still advertise ffmpeg built with fdk-aac without fear breaking "good repos" that tries to avoid such licensing issue [16:05:13 CET] <j-b> ffaac is good enough [16:05:19 CET] <j-b> and opus is amazing [16:05:38 CET] <kwizart> j-b, note quite, it's GPL incompatible as stated by fsf, but it might be floss [16:05:47 CET] <j-b> it is not. [16:05:54 CET] <kwizart> (they don't say it's not) [16:06:09 CET] <kwizart> they say it should be avoided when possible, which is good [16:06:32 CET] <j-b> "You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec software or your modifications thereto." [16:07:13 CET] <JEEB> "This is a free software license as far as it goes. It is incompatible with any version of the GNU GPL." <- whatever this means @ the FSF page [16:07:21 CET] <JEEB> "as far as it goes" [16:07:33 CET] <JEEB> that just raises more questions as a wording than anything else [16:07:50 CET] <kwizart> j-b, I don't see that in the current fdk-aac license at all [16:10:06 CET] <kwizart> j-b sorry my bad, it's there, but it's not an issue [16:12:13 CET] <j-b> https://opensource.org/osd [16:15:25 CET] <jamrial> jkqxz: ticket 7538, libva-x11 pkg-config apparently doesn't pull x11 ldflags and cflags [16:16:16 CET] <jamrial> while libva needs to fix it upstream, configure should also make vaapi_x11 depend on xlib, so it may include its ldflags [16:17:41 CET] <kwizart> j-b, maybe ffmpeg/vlc should publish their reading of the fdk-aac license , it would be interesting to compare with the fsf statement wording [16:18:15 CET] <kwizart> but anyway, thx for all advices and sorry for raising that issue again [19:02:40 CET] <cone-228> ffmpeg 03Paul B Mahol 07master:975510758cc7: avfilter/avf_showspectrum: improve magma colors [19:02:41 CET] <cone-228> ffmpeg 03Paul B Mahol 07master:ed58db4237c8: avfilter/avf_showspectrum: add viridis color map [19:02:42 CET] <cone-228> ffmpeg 03Paul B Mahol 07master:0c6d4e7ba4d7: avfilter/avf_showspectrum: add plasma color map [20:57:18 CET] <jkqxz> jamrial: True, but that doesn't appear to be the problem there because it's complaining about libavutil? [20:57:25 CET] <jkqxz> To get libva_x11 you need to have xlib, and xlib gives you those libraries (<http://git.videolan.org/?p=ffmpeg.git;a=blob;f=configure;h=00b5d9795e0daf5c…>). [20:57:29 CET] <jamrial> jkqxz: hwcontext [20:57:52 CET] <jkqxz> Seems to work fine for me. --disable-xlib indeed loses the X flags and libva_x11 too. [20:58:06 CET] <jamrial> nothing is making libavutil depend on (aka, include ldflags from) xlib [20:58:20 CET] <jamrial> or at least nothing obvious [21:05:40 CET] <atomnuker> I think we wanted to get rid of specific vaapi backends eventually and just always used a dri device path [21:06:32 CET] <jkqxz> Getting rid of the X11 backend would be sensible, it's not useful to anyone. [21:07:24 CET] <jkqxz> Ah, I see. VDPAU works because it includes -lX11 explicitly in its own flags, and autodetect pulls that in. [21:07:49 CET] <jkqxz> Whereas VAAPI doesn't have anything making that dependency, and libva-x11 also lacks it. [21:11:30 CET] <jkqxz> Is <https://0x0.st/sIFj.diff> right? It seems to work, but I'm not sure how those configure parts interact exactly. [21:13:00 CET] <jamrial> yeah, that should do it [22:33:45 CET] <atomnuker> BBB: btw gave up on doing adaptation as a separate function, I'll make the whole decoding function in asm [22:34:27 CET] <atomnuker> call overhead was almost 2x the overhead for doing the basic full avx2 adaptation [22:35:43 CET] <atomnuker> with the latest blend avx2 simd decode_symbol_adapt had the most overhead for chimera at 13% the whole decoder [22:38:24 CET] <BBB> yes I saw that too [22:38:25 CET] <BBB> ok [22:39:16 CET] <BBB> this is a good sign btw, now that ec is becoming most runtime [22:39:19 CET] <BBB> it means we did well :) [00:00:00 CET] --- Thu Nov 8 2018

1 0

ffmpeg.log.20181107
by burek 08 Nov '18

08 Nov '18

[00:26:37 CET] <benbro> furq: padding/scaling before format seems to fix it [00:26:39 CET] <benbro> thank you [00:27:02 CET] <benbro> I still don't understand why it sometimes succeeded even with odd width/height [03:36:34 CET] <ZeroWalker> can i trim a video so that the audio starts at the same time as the video (currently it's negative, so audio plays a bit before the video, so i just want to cut out that till the video starts) [03:36:55 CET] <ZeroWalker> so basically, cut audio till the first key frame [03:52:21 CET] <mixfix41> which video editor would you use just ffmpeg? [03:52:31 CET] <mixfix41> maybe use a gui video editor [03:53:06 CET] <frendShot> if he's asking in #ffmpeg, he's probably asking how to use ffmpeg to do that... [03:56:14 CET] <mixfix41> oh ok so just trim the file to the file starts both audio and video with ffmpeg [03:57:51 CET] <mixfix41> yea i do sound a bit brash but i was actually visualizing using kdenlive and all but yeah a quick trim off the beginning should do the work [04:40:26 CET] <mixfix41> ZeroWalker: i think you could use this https://superuser.com/questions/138331/using-ffmpeg-to-cut-up-video for single clips then but later used this https://superuser.com/questions/820747/slicing-video-file-into-several-segm… due to a better file/multiple trims [11:15:16 CET] <mmu_man> plop [11:15:19 CET] <mmu_man> is trac down? [11:25:30 CET] <c_14> works here [11:41:25 CET] <BtbN> It's been insanely slow the last couple days for me [11:42:08 CET] <zerodefect> In what respect? [11:44:36 CET] <zerodefect> Using the C-API and when dealing with sRGB, what do I set the AVFrame's color primaries too (ie. AVColorPrimaries). AVCOL_PRI_BT709 ? [13:03:56 CET] <zerodefect> Hmmm...and if want to convert between bt.709 and bt.601, can I get away with swscale or am I forced to utilize colorspace filter. [13:05:17 CET] <furq> zerodefect: use colorspace or zscale [13:07:02 CET] <zerodefect> Does swscale not make the grade? [13:07:59 CET] <zerodefect> My hesitation with using filters is that the graphs are very static....cannot change input parameters without recreating graph. [13:08:44 CET] <furq> you could also just flag the output as 709 if that's an option [13:09:28 CET] <zerodefect> Not sure what you mean, can you elaborate? [13:10:56 CET] <furq> i mean just not converting colourspace and setting the output primaries etc to 709 [13:11:11 CET] <furq> if that's not an option then you're stuck with using filters afaik [13:12:17 CET] <zerodefect> Ok. I guess I ask this because I note that swscale has a `sws_setColorspaceDetails(..)` function. [13:17:29 CET] <mmu_man> [ogg @ 0x5597871796c0] Error reading attached picture data. 708 != 5120 [13:17:44 CET] <mmu_man> no wonder I can't get a thumbnail inside [13:17:49 CET] <mmu_man> seems to be truncated [13:32:44 CET] <mmu_man> uint8_t line[1024]; [13:32:46 CET] <mmu_man> bleh& [13:34:33 CET] <mmu_man> oh [13:35:35 CET] <mmu_man> & [13:38:22 CET] <mmu_man> \o/ [13:38:30 CET] <mmu_man> en forçant le buffer à 256k ça passe [13:38:40 CET] <mmu_man> sorry [13:38:49 CET] <mmu_man> forcing the buffer size to 256k worked [13:44:42 CET] <mmu_man> hmm aviobuff:ff_get_line & ffmetadec:get_line could use some factoring [14:51:05 CET] <mmu_man> patch sent [14:54:28 CET] <wr> hello, need a help on converting a mp4 video file on the ffmpeg to ogv, how can do it? [15:29:52 CET] <saml> wr, ffmpeg -i a.mp4 -c:v libtheora -c:a libvorbis a.ogv [15:30:37 CET] <wr> ah thanks [15:54:50 CET] <Fenrirthviti> Hey folks, I know this isn't really your problem, but looking for any leads on what might be causing this crash: https://obsproject.com/logs/9OJ_4cLDPVgO6i8h [15:55:02 CET] <Fenrirthviti> NVENC startup crash when using ffmpeg as the output [15:55:51 CET] <Fenrirthviti> Curious if anything similar has been seen with ffmpeg natively or not [16:20:31 CET] <BtbN> Impossible to say without debug symbols, but I'd blame a broken driver. [16:20:48 CET] <BtbN> Or really horribly broken parameters. [16:23:33 CET] <zerodefect> Agreed about debug symbols. c0000005 means `Access Violation` from memory. It's not very happy. [16:24:54 CET] <Mavrik> Feels violated. [16:32:03 CET] <wr> saml, the video just turned to be more low quality than the mp4, any way i increase it? [16:32:46 CET] <relaxed> wr: https://trac.ffmpeg.org/wiki/TheoraVorbisEncodingGuide [16:33:17 CET] <saml> you mean it became retro and better [16:33:35 CET] <wr> saml, too much retro lol [16:45:49 CET] <kepstin> well, you asked for theora, you got retro ;) [16:47:37 CET] <kepstin> why do you want theora anyways? pretty much the only thing that's needed for is obsolete versions of firefox, afaik. [18:49:56 CET] <pi-> I'm experiencing a strange problem. [18:50:07 CET] <pi-> I have a source .caf waveform. [18:50:42 CET] <pi-> If I convert it to .wav with Audacity (which if I remember correctly uses ffmpeg), my DSP code processes it correctly. [18:51:09 CET] <pi-> However if I use `ffmpeg -i X.caf X.wav`, the code now fails to process it correctly. [18:52:07 CET] <kepstin> can you pastebin the output of that ffmpeg command please? [18:52:38 CET] <kepstin> and also what do you mean by "fails to process it correctly?" how/where does it fail? [18:54:43 CET] <kepstin> with that command, ffmpeg will convert the audio to 16-bit little-endian pcm in the wav file, but it should preserve the sample rate and number of channels from the input. [18:56:41 CET] <pi-> kepstin: https://paste.pound-python.org/show/44sRGMU0wu3bZ3eRysnL/ [18:57:43 CET] <pi-> If I examine hexdumps of the generated WAVs (both from audacity and from ffmpeg), they do look strange. Every single 4-cluster ends ??00 or ??ff. [18:58:10 CET] <kepstin> yeah, that's converting from big-endian audio (in the caf) to little-endian (in the wav). If your DSP code expects the opposite endian and doesn't read the wav headers properly that could cause issues [18:58:38 CET] <pi-> I am attempting to extract an ultrasound signal, so 'fails' is difficult to define. [18:58:51 CET] <pi-> It is actually a partial failure. [18:59:05 CET] <pi-> One part of the signal gets recognised successfully, another part doesn't. [18:59:42 CET] <pi-> I wrote my own WAV reader, so it's possible I'm not processing the header right. [18:59:50 CET] <kepstin> how in the world would you get ultrasound stuff from a 48kHz audio file? if it's lowpassed correctly, it won't have anything above 24kHz. [19:00:00 CET] <pi-> well near-ultra [19:00:03 CET] <pi-> 18-20kHz [19:00:11 CET] <kepstin> ah, yeah. [19:00:40 CET] <kepstin> one thing you can try is to add "-c:a pcm_s16be" to the ffmpeg command, to leave the audio in big-endian in the wav file, see if that helps [19:01:23 CET] <kepstin> if it does, then your app is expecting big-endian audio, and it isn't reading the wav headers correctly or isn't doing the required conversion before processing. [19:02:20 CET] <kepstin> oh, nvm, wav doesn't support big endian at all [19:02:23 CET] <kepstin> my mistake [19:02:27 CET] <pi-> the endianness appears to be consistent between ffmpeg and audacity output. [19:02:30 CET] <kepstin> (it's a PC/windows thing) [19:03:05 CET] <pi-> I'm guessing the xx00 and xxff everywhere means small positive and small negative numbers [19:03:37 CET] <pi-> i.e. between -/+ 255 over a total range of -/+ 32768 [19:18:19 CET] <kepstin> yeah, that makes sense. [19:19:50 CET] <kepstin> in the caf file, i'd expect the values to all look like 00xx or FFxx [19:30:39 CET] <GlidingHorse> hi, can I mux HEVC into an mpegts with ffmpeg 4.0? [19:31:03 CET] <GlidingHorse> i'm not asking how ,just asking if it's supported [19:31:13 CET] <durandal_1707> no [19:31:45 CET] <GlidingHorse> :( [19:31:59 CET] <kepstin> pi-: hmm. one thing to note is that ffmpeg usually adds some metadata to the wav file, you want to make sure your wav reader correctly locates the start of the "data" chunk. [19:32:32 CET] <durandal_1707> GlidingHorse: or yes, dunno i thought you ask about av1 [19:32:55 CET] <GlidingHorse> :) [19:33:47 CET] <GlidingHorse> anyway i will try by myself, but wanted to gain some time while ffmpeg downloads [19:34:15 CET] <durandal_1707> it should be possible, i see lines with HEVC in mpegts muxer [19:34:40 CET] <GlidingHorse> excelent :D [20:08:25 CET] <GlidingHorse> indeed, ffmpeg muxed hevc video and e-ac-3 audio into mpegts :D [20:08:35 CET] <GlidingHorse> thanks for the support :D [20:08:37 CET] <GlidingHorse> o/ [20:08:38 CET] <GlidingHorse> \part [00:00:00 CET] --- Thu Nov 8 2018

1 0

ffmpeg-devel.log.20181106
by burek 07 Nov '18

07 Nov '18

[00:02:05 CET] <jamrial> jkqxz: if you're also pushing them to release/4.1 yourself, do the same with a4fb2b1150 and e9d2e3fdaa while at it [00:03:47 CET] <jkqxz> Ok, sure. [00:07:00 CET] <cone-309> ffmpeg 03Mark Thompson 07master:6bdb7712ae02: cbs_av1: Fix header writing when already aligned [00:07:01 CET] <cone-309> ffmpeg 03Mark Thompson 07master:f5894178fb80: cbs_av1: Support redundant frame headers [00:17:37 CET] <cone-309> ffmpeg 03Mark Thompson 07release/4.1:066ff02621ce: configure: Add missing IVF muxer BSF dependency [00:17:38 CET] <cone-309> ffmpeg 03Mark Thompson 07release/4.1:ec1b5216fc65: configure: Add missing V4L2 M2M decoder BSF dependencies [00:17:39 CET] <cone-309> ffmpeg 03Mark Thompson 07release/4.1:af3fccfeff74: cbs_av1: Fix header writing when already aligned [00:17:40 CET] <cone-309> ffmpeg 03Mark Thompson 07release/4.1:10506de9ad1f: cbs_av1: Support redundant frame headers [01:09:27 CET] <cone-309> ffmpeg 03Mark Thompson 07n4.1:HEAD: cbs_av1: Support redundant frame headers [01:25:13 CET] <cone-309> ffmpeg 03Mark Thompson 07n4.2-dev:HEAD: cbs_av1: Support redundant frame headers [02:35:21 CET] <cone-309> ffmpeg 03James Almer 07master:beaa350e2416: avcodec: libdav1d AV1 decoder wrapper. [10:51:55 CET] <cone-871> ffmpeg 03Linjie Fu 07master:87368884a52b: lavc/qsvdec: flush buffered data before reinit [10:51:56 CET] <cone-871> ffmpeg 03Zhong Li 07master:d96ae9d5ea1f: lavc/qsv_overlay: suppress code scan complain [12:40:56 CET] <funman> sys.path.append(os.path.join(sys.MEIPOASS, 'out00-PYZ.pyz')) [12:42:55 CET] <JEEB> what's the best way to figure out which part is blocking with f.ex. ffmpeg.c? [12:43:28 CET] <JEEB> as in, is it filter chain with splits etc, or is it encoding or what [12:43:45 CET] <JEEB> what are good tools to profile such stuff? [12:44:05 CET] <nevcairiel> i would go low-tech and test the invidiual steps individually [12:44:14 CET] <nevcairiel> measure their speed [12:44:41 CET] <JEEB> it's probablematic with live inptus [12:44:50 CET] <JEEB> I could of course dump and check [12:45:09 CET] <JEEB> but then you're not limited in input speed unless I use something like multicat [12:45:22 CET] <JEEB> which outputs MPEG-TS into UDP with speed control [12:45:32 CET] <nevcairiel> of course, but if filtering and encoding is like 10x realtime, its not going to limit you [12:46:27 CET] <JEEB> true. it's just that I have a hunch that I have a case of a filter chain getting stuck or so on one of the outputs or something like that [12:46:42 CET] <JEEB> but yea, I think that makes sense. dump some and see the steps separately [12:46:48 CET] <JEEB> also the overall load is probably one thing [14:06:11 CET] <Compn> ahhh [14:06:16 CET] <Compn> we still don't have qdmc decoder ? [14:06:54 CET] <Compn> no wait we do [14:06:55 CET] <Compn> hmm [14:07:53 CET] <Compn> is my vlc so old it does not have this mapped ? [14:08:45 CET] <durandal_1707> yes [14:09:58 CET] <Compn> durandal_1707 : btw i have imm5 sample (and other samples) for you [14:10:32 CET] <durandal_1707> i accept only money [14:17:43 CET] <Compn> sent you link [14:18:02 CET] <durandal_1707> link is not money, ignored [14:18:21 CET] <Compn> j-b iirc said he would pay you for imm5 decoder [14:18:30 CET] <Compn> there is imm5 sample in there [14:18:44 CET] <durandal_1707> what is in sample? [14:19:06 CET] <Compn> boring video , security cam or dash cam [23:07:03 CET] <durandal_1707> i'm retiring [23:13:18 CET] <durandal_1707> nobody cares, eh! [23:14:35 CET] <jamrial> i do! [23:14:36 CET] <j-b> retiring from? [23:16:14 CET] <Compn> durandal_1707 : retiring from ffmpeg ? [00:00:00 CET] --- Wed Nov 7 2018

1 0

ffmpeg.log.20181106
by burek 07 Nov '18

07 Nov '18

[00:38:52 CET] <petecouture> Building ffmpeg on centos Im getting an error thrown ERROR: libfdk_aac not found but the library was downloaded and installed as rpm which worked for librtmp. Im reading that there maybe an issue with the new version for centos 7 not including the lib into the name. Anyone else run into this? [00:39:58 CET] <pink_mist> doesn't centos require -devel versions of all such libraries if you want to build something that links to them? [00:40:55 CET] <petecouture> @pink_mist thanks I will review. I didnt run into this with the librtmp library. [00:43:41 CET] <pink_mist> I've never actually used centos, so I may well be mistaken [00:43:46 CET] <petecouture> @pink_mist that worked you rock! [00:43:51 CET] <pink_mist> oh good =) [10:28:46 CET] <mrtAkdeniz> Hey there [10:29:17 CET] <mrtAkdeniz> Im trying to use an image as camera source.. Can I do that with ffmpeg? [10:29:46 CET] <mrtAkdeniz> I tried to google it, but probably used wrong keywords.. [10:51:42 CET] <thin> If I want to take a bunch of webm's and supercut random 1-second intervals from all of them into a video, what's my best option [13:02:22 CET] <haasn> ffmpeg: code 3074: invalid colorspace definition [13:02:24 CET] <haasn> huh? [13:03:21 CET] <haasn> oh nvm I'm stupid [14:12:50 CET] <spaam> w [17:20:11 CET] <AlekseyVl89> Hello, huys [17:20:14 CET] <AlekseyVl89> Hello, guys! [17:21:02 CET] <AlekseyVl89> I'm stuck with the problem. ERROR: opus not found using pkg-config [17:21:26 CET] <AlekseyVl89> I'm trying to make ffmpeg for android. I tried on mac os and Ubuntu. And got the same error [17:21:44 CET] <AlekseyVl89> I've installed libopus with apt-get install libopus-dev [17:22:19 CET] <bencoh> what do you mean "for android"? [17:22:23 CET] <bencoh> is that cross-compilation? [17:22:30 CET] <kepstin> you need to build all the dependent libraries (e.g. libopus, etc.) in your cross-compile build environment. You can't use your system libraries (probably x86) for android (probably arm) [17:22:32 CET] <lays147> Hello guys, I am trying to cut a video, however, the cutting is not happening, the param '-c copy' is mandatory? because its apparently giving conflict with the -vf param [17:22:36 CET] <bencoh> if so, then installing a native package won't help [17:22:59 CET] <kepstin> lays147: if you're filtering video, then you can't copy it, because if you copy it it can't be modified by filters. [17:24:28 CET] <AlekseyVl89> bencoh, thank you. [17:24:44 CET] <AlekseyVl89> I managed to build libopus for android as static library [17:24:54 CET] <AlekseyVl89> How can I compile ffmpeg with it? [17:25:45 CET] <kepstin> AlekseyVl89: after installing libopus to your cross environment root, set PKG_CONFIG_PATH to contain the appropriate pkg-config directory in that corss env. [17:26:54 CET] <AlekseyVl89> like this? PKG_CONFIG_PATH="$HOME/ffmpeg_build/lib/pkgconfig" ./configure \ --pkg-config-flags="--static" \ [17:27:02 CET] <bencoh> and assuming you're properly generating your sysroot, you might want to have a look at PKG_CONFIG_SYSROOT_DIR and PKG_CONFIG_LIBDIR [17:29:56 CET] <AlekseyVl89> Thanks, guys [20:16:42 CET] <ChocolateArmpits> How is fpsprobesize different from probesize? I have a dv file with messed up audio that doesn't start at beginning, increasing probesize or analyzeduration doesn't help in detecting it initially, however using fpsprobesize helps. [20:26:42 CET] <frendShot> if I use the minterpolate filter to go from a 29.97 fps video to a 60fps one, is it normal for the audio to go out of sync? [20:31:24 CET] <No0n3Left> Do you mean the audio is distorted or that it is not in sync with the video? Assuming it would get a bit out of sync with the video just the framerate has changed and it isnt a multiple of the original, not sure though. [20:32:39 CET] <frendShot> as in, it's out of sync with the video; actions take place at one time in the video, the sound doesn't coincide with them [20:33:44 CET] <frendShot> the sound itself is identical; the minterpolate filter didn't like it when I gave it the video with the audio also muxed in, so I separated the two losslessly, and used the video stream alone with minterpolate [20:33:55 CET] <frendShot> and muxed the audio back in, and that's when it's out of sync [20:33:56 CET] <No0n3Left> Is the video still the same length? [20:34:41 CET] <frendShot> *exactly* the same length? I doubt it; I interpolated from 29.97fps to 60fps... but that shouldn't be that important, right? [20:34:59 CET] <No0n3Left> Do events take place at the same time if you play them back at the same time? [20:35:26 CET] <frendShot> hm? Sorry, I didn't understand [20:35:56 CET] <No0n3Left> Like open the videos at the exact same time and start them. Do they stay in sync, not the audio, but hte actual video itself? [20:36:14 CET] <frendShot> oh I see, the original 29.97 fps and the new 60 fps ones [20:36:16 CET] <frendShot> let me check [20:36:46 CET] <No0n3Left> Cause otherwise it may be something happening when you put the audio back in [20:39:38 CET] <frendShot> heh, the events take place at different times... I think I see what happened; the new interpolated frames occur before the sound [21:01:40 CET] <kepstin> frendShot: the minterpolate filter should preserve video timing. for best results you should have the audio and video in the same file so ffmpeg can manage timestamps for everything. [21:02:02 CET] <kepstin> it sounds like your video originally had an offset between audio and video start, and that was lost when you separated them [21:02:15 CET] <frendShot> kepstin, that's the thing, the minterpolate filter didn't like it when I gave it the video with the audio also muxed in [21:02:17 CET] <saml> if you change framerate, you have to reencode audio, right? [21:02:25 CET] <frendShot> so I had to demux it and then use the filter [21:02:37 CET] <kepstin> frendShot: minterpolate filter has nothing to do with audio [21:02:41 CET] <kepstin> frendShot: it should work fine [21:02:51 CET] <frendShot> well, I understand that, but ffmpeg threw some error about it [21:02:58 CET] <frendShot> I could try and replicate it and paste output here in pastebin [21:02:59 CET] <kepstin> frendShot: we can help you with that... [21:03:05 CET] <saml> usually, when I alter video with -c:a copy, there is a-v sync problem [21:03:11 CET] <frendShot> saml, I see [21:03:22 CET] <saml> so, i'm lazy and do -c:a aac or something [21:03:33 CET] <frendShot> yeah, I used -c:a copy to demux the audio, I used an m4a file since it was an AAC stream from a DVD [21:03:34 CET] <saml> there could be a smarter way [21:03:57 CET] <kepstin> saml: hmm, i'd really expect -c:a copy to keep the offsets/timing correct. [21:04:05 CET] <frendShot> ikr [21:05:13 CET] <saml> hrm yeah i could've done something else. if i experience a-v sync problem with exact command line i used, i'll report [21:16:30 CET] <markweston> how to change L/R audio balance in ffplay or somehow else? [21:18:22 CET] <ChocolateArmpits> markweston, for a single file ? [21:18:23 CET] <ChocolateArmpits> you'd have to use the pan audio filter [21:18:29 CET] <durandal_1707> markweston: stereotools is most user friendly way [21:24:29 CET] <markweston> hey, any way to change L/R audio balance? [21:24:39 CET] <markweston> sorry, wrong channel, I already asked [21:24:43 CET] <frendShot> lol [21:24:54 CET] <frendShot> I was going to say, didn't you see ChocolateArmpits and durandal_1707's answers [21:26:03 CET] <markweston> I just don't want to re-encode the whole movie and OpenBSD doesn't have stereotools in packages, so I'll see if I can do it on OS level with sndio or maybe there's some other media player that can do it on the fly [21:27:13 CET] <kepstin> stereotools is an ffmpeg filter, not a separate package. But yeah, many players will be able to do this, I'm pretty sure mpv can at least. [21:28:51 CET] <durandal_1707> markweston: mpv can do it in fly, get and copy this script: https://github.com/wiiaboo/mpv-scripts/blob/master/audio-balance.lua [21:29:59 CET] <frendShot> woops, wrong button [21:30:07 CET] <frendShot> kepstin, everyone else, https://pastebin.com/raw/1SBsjbrL this is the output [21:30:20 CET] <frendShot> the input is ".\ffmpeg.exe -i .\out2.mkv -filter "minterpolate='fps=60'" output.60fps.mkv" [21:30:25 CET] <frendShot> (Windows PowerShell) [21:30:35 CET] <frendShot> the few lines of errors are at the bottom [21:30:40 CET] <kepstin> frendShot: and your ffmpeg command line? [21:30:50 CET] <frendShot> .\ffmpeg.exe -i .\out2.mkv -filter "minterpolate='fps=60'" output.60fps.mkv [21:31:00 CET] <frendShot> that line had my name in it, so I didn't copy it [21:31:03 CET] <kepstin> oh, the issue is that you used -filter without a stream specifier [21:31:15 CET] Action: frendShot eyes bug out [21:31:15 CET] <kepstin> either use -filter:v to apply it only to video streams, or use -vf (short option) [21:31:21 CET] <frendShot> I see [21:31:36 CET] <markweston> durandal_l1707: thanks man, it's ironic I got the link for the script on #ffmpeg and not on #mpv, where I asked too. The guy there only said "iirc there was a script to do that" but didn't say anything else [21:32:25 CET] <kepstin> frendShot: you'll want to specify some codec options, something like "-c copy -c:v libx264" will copy everything except the video, which will be re-encoded with x264. [21:32:54 CET] <frendShot> I see, and to retain lossless video I use -crf 0 with libx264 right? Even though I guess the file will be huuuuge [21:33:06 CET] <kepstin> frendShot: lossless video is a bad idea [21:33:06 CET] <frendShot> i'm not sure how lossless video will work with framerate change though [21:33:24 CET] <kepstin> use a crf value that gives no quality loss you notice, and that's good enough [21:33:32 CET] <frendShot> right, so transparent [21:33:47 CET] <kepstin> frame interpolation requires making changes to video content, so it is by definition not lossless [21:34:04 CET] <frendShot> yeah, that's what I figured ... so what exactly does -crf 0 here do then? [21:35:07 CET] <kepstin> -crf 0 in 8-bit x264 (the general option for all modes is actually -qp 0) turns on lossless mode, where when you decode the resulting video, you'll get back the exact same bits as were sent to the encoder [21:35:17 CET] <frendShot> i see [21:35:23 CET] <kepstin> this requires huge amounts of space compared to normal lossy video encoding. [21:35:36 CET] <frendShot> yeah, and in the past has given me compatibility issues, too [21:35:58 CET] <kepstin> yes, it requires the decoder support a profile that many hardware decoders and players don't do. [21:36:05 CET] <frendShot> yeah [21:37:17 CET] <kepstin> basically, the only reason to use lossless mode is for certain special archiving tasks where you need to ensure the encoder does not add any artifacts whatsoever, and as a temporary intermediate format to avoid generation loss from multiple encoding/decoding steps. [21:37:36 CET] <frendShot> sweet [21:37:44 CET] <frendShot> all right, lemme try these commands out and report back, thanks [21:50:22 CET] <frendShot> kepstin, thanks, this seems to have done the trick [21:50:38 CET] <frendShot> I guess somewhere along the muxing and demuxing steps, something got misaligned [21:51:06 CET] <aleph-> Hey is there any way at all to get multi core/parallelized encoding with av1? [21:51:48 CET] <kepstin> aleph-: are you a programmer familiar with codecs and threading? maybe you can help! :) [21:52:27 CET] <aleph-> Threading somewhat, codecs not in the slightest. [21:52:43 CET] <aleph-> Taking that as a no as well. :P [21:53:24 CET] <kepstin> av1 isn't really a codec ready for general usage yet, there's a lot more work on the encoder needed. [21:53:35 CET] <aleph-> Aye I figured. [21:53:38 CET] <frendShot> isn't it super slow at the moment? [21:54:03 CET] <aleph-> Indeed. [21:54:13 CET] <kepstin> well, iirc rav1e is only very slow rather than super slow, but iirc it's also rather less efficient? [21:54:27 CET] <kepstin> libaom is super slow, yeah [21:54:46 CET] <aleph-> How is dav1d? [21:55:02 CET] <kepstin> dav1d is a decoder, not encoder [21:55:06 CET] <aleph-> Oh doy. [21:55:33 CET] <aleph-> And I've heard rav1e is rather limited on input type right? [21:55:56 CET] <kepstin> aleph-: most standalone encoders are. Handily, most accept input piped from ffmpeg. [21:56:16 CET] <aleph-> Aye [21:56:49 CET] <kepstin> i mean, if you want to write a standalone tool that accepts arbitrary input, the way to do that is to use the ffmpeg libraries to decode the input :/ [23:08:37 CET] <benbro> [libx264 @ 0x557ffb0db440] width not divisible by 2 (329x308) [23:08:44 CET] <benbro> Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height [23:08:59 CET] <benbro> I'm using 'format=yuv420p,scale={0}:-2' [23:09:15 CET] <benbro> shouldn't it handle width 329? [23:09:50 CET] <klaxa> nope [23:10:14 CET] <benbro> klaxa: any workaround? [23:10:28 CET] <pink_mist> a lot of formats require a multiple of 8 or something like that I believe [23:10:46 CET] <klaxa> *2 / 2? [23:10:52 CET] <klaxa> err... [23:10:55 CET] <benbro> ? [23:10:57 CET] <pink_mist> you could add black bars to pad it out to a multiple of 8 [23:11:02 CET] <c_14> yuv420 ist even numbers only [23:11:16 CET] <pink_mist> ah, ok, multiple of 2 for that one then [23:11:20 CET] <c_14> -t [23:11:53 CET] <benbro> no automatic way to handle it? [23:12:09 CET] <c_14> there is [23:12:13 CET] <benbro> what? [23:12:14 CET] <c_14> eeeh [23:12:43 CET] <klaxa> mathematically, integer division by 2 and multiplication by 2 should yield even numbers [23:13:02 CET] <benbro> -vf "pad=ceil(iw/2)*2:ceil(ih/2)*2" ? [23:13:12 CET] <benbro> found this here: https://stackoverflow.com/a/20848224 [23:13:15 CET] <c_14> yes, that [23:13:16 CET] <benbro> but feels dirty [23:13:18 CET] <c_14> but only for the first one [23:13:20 CET] <c_14> the second can stay -2 [23:14:46 CET] <c_14> and you can use your {0} variable for iw [23:15:29 CET] <benbro> my {0} variable is divisible by 2 [23:15:56 CET] <c_14> then that should work [23:16:18 CET] <benbro> checking [23:21:32 CET] <benbro> the error is intermittent [23:21:41 CET] <benbro> sometimes I see the error and sometimes not [23:21:56 CET] <benbro> I don't understand how it's possible [23:21:59 CET] <c_14> log your {0} value, maybe your code is bugged? [23:22:04 CET] <frendShot> then your variable sometimes isn't divisible by two [23:22:12 CET] <frendShot> have your code print out what it is at each step to debug it [23:22:56 CET] <furq> you should probably be doing scale,format and not the other way around [23:23:12 CET] <c_14> oh, yeah [23:23:23 CET] <furq> although you'd get a different error message in that case [23:23:24 CET] <c_14> if your input isn't even that would break it [23:23:32 CET] <c_14> *should [23:23:51 CET] <furq> that's just generally good practice anyway [23:26:25 CET] <benbro> furq: I'll try to switch [23:26:38 CET] <benbro> can it explain why it sometimes fail and sometimes doesn't fail? [23:27:10 CET] <furq> yuv420p resizes the chroma planes to half width/height of the luma plane [23:27:23 CET] <furq> so if the width is an odd number then it can't do that [23:28:01 CET] <benbro> so it should always fail for the same video [23:28:03 CET] <furq> i guess it could just make up a pixel but it's probably better to just get you to do it [23:28:12 CET] <benbro> ok. trying [23:28:21 CET] <furq> yeah if it's intermittently failing on the same source clip then idk what the problem would be [23:29:11 CET] <benbro> furq: ok. trying to scale before format [23:35:37 CET] <frendShot> is there a preferred order when specifying filters? like, if I wanted to upscale and interpolate in the same command [23:38:56 CET] <furq> frendShot: it depends on the filters [23:39:29 CET] <benbro> furq: I have predefined resolutions but if the video width is smaller than all variations I just use it as is [23:39:33 CET] <furq> generally you want to do stuff before scaling so it has more information to work with [23:39:42 CET] <benbro> and in this case it was uneven number [23:39:51 CET] <furq> but if it's really slow (like minterpolate) then you might want to do it afterwards [23:39:52 CET] <frendShot> furq, to be specific, minterpolate='fps=60' and scale=1080:720 [23:39:57 CET] <frendShot> I see [23:40:10 CET] <benbro> furq: should I pad with 1 pixel in this case? [23:40:19 CET] <furq> benbro: pad or scale, it's up to you [23:41:06 CET] <benbro> furq: is it better to add or remove a 1 pixel? or maybe doesn't matter? [23:41:09 CET] <furq> i would normally just crop a pixel off, but it's entirely down to preference [23:41:43 CET] <benbro> crop instead of scale? [23:41:44 CET] <furq> if you don't mind the black bar then padding is the least destructive [23:42:24 CET] <benbro> 'scale=440:-2' if the original is 339px will just stretch one of the pixels? [23:42:54 CET] <furq> if you mean 340 then yeah [23:43:13 CET] <benbro> sorry, I mean 439 [23:43:39 CET] <benbro> I want to add/remove/scale without changing much the size and least cpu overhead [23:44:03 CET] <benbro> I'll choose nicer numbers [23:44:19 CET] <pink_mist> I believe it's better to pad if it's just one single line of pixels [23:44:39 CET] <benbro> if the original video width=499, should I "scale=500:-2", or pad or crop? [23:44:52 CET] <furq> like i said, it's entirely down to preference [23:45:06 CET] <furq> any way of adding or dropping one pixel is fine [23:45:30 CET] <benbro> "scale=500:-2" won't distort a 499px video too much? [23:46:02 CET] <pink_mist> try it and compare to the other options [23:46:27 CET] <furq> you could try pad and fillborders if you want to avoid the black bar [23:46:51 CET] <benbro> 1px black bar probably won't be noticible [23:47:14 CET] <benbro> scale is easier for me in the code because I'll treat all videos the same [00:00:00 CET] --- Wed Nov 7 2018

1 0

ffmpeg-devel.log.20181105
by burek 06 Nov '18

06 Nov '18

[01:10:58 CET] <jkqxz> jamrial: Sent a new version. [01:12:51 CET] <jamrial> jkqxz: can't see it in my inbox yet, but will test it once it's there, thanks [01:32:16 CET] <jkqxz> jamrial: Did you have any thoughts on including some AV1 streams in fate for testing? [01:33:29 CET] <jamrial> jkqxz: the aom testsuite is pretty barebones (just different quantizer setting during encoding), but it's enough for some basic tests [01:34:18 CET] <jamrial> aside from those, we could include some of the small samples i created using Tears of Steel to test parsing things like error resilence and redundant frame headers, timing info using decode model, and such [01:39:30 CET] <cone-223> ffmpeg 03Mark Thompson 07master:34429182b931: configure: Fix av1_metadata BSF dependency [01:41:34 CET] <jamrial> jkqxz: https://pastebin.com/qAxM1zBc [01:42:15 CET] <jamrial> file is unreadable after that. libaom can't read it, and i can't seem to get the ivf demuxer to output anything [01:42:56 CET] <jkqxz> Since it's read in order you lose the first tile group. [01:43:27 CET] <jkqxz> I was looking at trace_headers to show that the redundant frame header was used correctly. [01:43:54 CET] <jamrial> i did an av1_metadata passthrough and it works after your patch. checksum matches [01:44:19 CET] <jamrial> before it would crash and burn [01:44:49 CET] <jkqxz> Yep, your file passes through to be identical. I don't have any others to try, though. [01:45:30 CET] <jamrial> you can create them with aomenc by using error resilence mode with 2+ tile groups [01:45:58 CET] <jamrial> that's the only case where it adds redundant frame headers in between each tile group on every frame except key frames [01:57:16 CET] <cone-223> ffmpeg 03James Almer 07master:89a0d33e3a53: avcodec/cbs_av1: fix decoder/encoder_buffer_delay variable types [02:07:28 CET] <cone-223> ffmpeg 03James Almer 07release/4.1:1c98cf4ddd38: avformat/ivfenc: use the av1_metadata bsf to insert Temporal Delimiter OBUs if needed [02:07:29 CET] <cone-223> ffmpeg 03Mark Thompson 07release/4.1:acd13f12556b: configure: Fix av1_metadata BSF dependency [02:07:30 CET] <cone-223> ffmpeg 03James Almer 07release/4.1:398a70309e4f: avcodec/cbs_av1: fix decoder/encoder_buffer_delay variable types [02:18:06 CET] <nicolas17> can ffmpeg read AV1 without extra libraries? [02:18:12 CET] <jkqxz> If I made the bit comparison come one byte at a time rather than one bit at a time would that be better? [02:20:07 CET] <jkqxz> It could certainly be suppressed by splitting into read/write branches and just doing the necessary bit(read|write)er operations, but I think it's preferable to show something there. [02:21:12 CET] <jkqxz> Wrt refcounting, yes for read only. Again it needs to split the read/write branches out further, so code becomes even more horrible. [02:21:24 CET] <jkqxz> nicolas17: Depends what you mean by "read". [02:24:59 CET] <jamrial> nicolas17: decode? no. Parse, extract, remux? yes [02:47:32 CET] <BBB> nicolas17: for av1, some of us decided to write the decoder outside ffmpeg, so its a library to be linked in. some others are planning to eventually integrate the code directly into ffmpeg, which is fine but not yet done [13:41:32 CET] <dualz> I'm writing a filter that takes two inputs and has two outputs (audio and video). I noticed that filters with two input pads do not receive a filter_frame call back. Upon further reading it says I should use activate() so I messed around trying to get it to work and am dropping frames. I saw an example of a filter with multiple inputs (streamselect) and they use FrameSync, but FS doesn't work with multiple outputs [13:43:21 CET] <dualz> A little guidance on how to use activate() would be appreciated. I can't find a single filter that has two inputs and two outputs for an example [13:45:30 CET] <durandal_1707> dualz: what your filter will do that it needs 2->2 ? [13:46:37 CET] <dualz> for the time being I want a simple pass through [13:47:00 CET] <durandal_1707> framesync as name says, is for receiving synced frames from all inputs, eg, it will buffer to match same time before filtering [13:48:29 CET] <durandal_1707> dualz: afir filter have 2 audio inputs and audio output + video output, but its very specific usecase [13:49:11 CET] <durandal_1707> so doing only passthrough with activate is easy [13:49:40 CET] <dualz> is there a writeup somewhere that'll help? [13:50:04 CET] <durandal_1707> just consume frame from all inputs and output what you want with whatever pts you seem fit [13:51:35 CET] <durandal_1707> look at concat filter [13:51:55 CET] <durandal_1707> and feel free to ask questions if something is not clear [13:52:03 CET] <dualz> so activate { for each input { consume_frame } then for each output filter_frame } ? [13:52:18 CET] <durandal_1707> ff_filter_frame [13:52:27 CET] <dualz> can I call filter frame twice in one activate? [13:54:10 CET] <durandal_1707> dualz: activate does one thing only per call, so better no, best is output one frame and then have state what to push when [13:54:56 CET] <durandal_1707> also you need to signal that you need frames from inlinks [13:55:16 CET] <durandal_1707> and need to handle eof on inlinks and outlinks [13:58:07 CET] <dualz> if activate does one thing, how do I know to consume my audio input or my video input [13:58:50 CET] <dualz> sorry for the noob questions, I just started with ffmpeg [14:00:54 CET] <durandal_1707> dualz: by keeping state in some variable in filter private context [14:01:25 CET] <durandal_1707> and you can consume both video and audio at same time if its availabe [14:01:37 CET] <durandal_1707> just not output both... [14:05:03 CET] <dualz> so I consume both audio and video, I now how a AVFrame* video and *audio when is the best time to output these [14:05:28 CET] <dualz> if I can only output 1 per activate [14:06:30 CET] <dualz> store a reference in my filter context then wait for the next activate? [14:06:32 CET] <durandal_1707> you keep track of consumed frames in context, if you filter out consumed frame, you set it to NULL, so you need to pick new one, if its not NULL you output it [14:15:51 CET] <dualz> durandal_1707: is this the right idea https://gist.github.com/ryanmarin/213606922dff8fd0ae1a513727260483 [14:16:54 CET] <dualz> with this design will there ever be a activate() call where ctx.abuf is null and it'll try to output ctx.vbuf [14:18:46 CET] <durandal_1707> dualz: ff_inlink_consume_frame may return 0 if there is no frame available [14:19:27 CET] <durandal_1707> also your gist have wrong check for null/0 case [14:20:08 CET] <durandal_1707> also you do not need to consume all at once, just when you need it [14:20:28 CET] <dualz> how would i know when i need it [14:23:03 CET] <durandal_1707> dualz: ff_outlink_frame_wanted() [14:23:30 CET] <durandal_1707> you call that that to check if next filter in chain wants frame [14:24:54 CET] <durandal_1707> and if you need frame from inlink you call ff_inlink_request_frame() [14:37:53 CET] <dualz> durandal_1707: https://gist.github.com/ryanmarin/083afed8e909167da24d119d662ab6bf closer? [14:41:58 CET] <durandal_1707> dualz: something like that, missing eof handling and forwarding outlink status to other links [14:42:41 CET] <dualz> will ctx.vbuf ever be null for the ctx.abuf code to run? [14:44:40 CET] <durandal_1707> you can add check for other buf to not request new one if there is one to output [14:45:18 CET] <durandal_1707> AV case may have several small audio frames for each video frame [14:45:59 CET] <durandal_1707> and you can use consume_samples to keep AV in sync [15:05:32 CET] <cone-321> ffmpeg 03Martin Storsjö 07master:4e9cff282476: libavutil: Undeprecate the AVFrame reordered_opaque field [15:05:32 CET] <cone-321> ffmpeg 03Martin Storsjö 07master:882ae091d48b: libx264: Pass the reordered_opaque field through the encoder [15:05:32 CET] <cone-321> ffmpeg 03Martin Storsjö 07master:e7ed9d81bff0: flvdec: Rename FLV_STREAM_TYPE_DATA into FLV_STREAM_TYPE_SUBTITLE [15:05:32 CET] <cone-321> ffmpeg 03Martin Storsjö 07master:d7638d8dfc3c: flvdec: Export unknown metadata packets as opaque data [15:10:39 CET] <dualz> ok so I'm looking at consume samples [15:10:55 CET] <dualz> there's a min and a max argument [15:11:17 CET] <dualz> where do I get these from? [15:12:38 CET] <atomnuker> wbs: mate, 882ae091d48b112004b977314884176841f12cef looks very sketchy [15:12:49 CET] <atomnuker> wouldn't this be better done using side data? [15:13:26 CET] <atomnuker> would avoid the need for a codec cap flag too [15:13:49 CET] <atomnuker> but most importantly you wouldn't be stuffing pointers in random avframe fields [15:14:03 CET] <nevcairiel> this isnt exactly new api, that existed for years [15:14:10 CET] <nevcairiel> also, read ML if you want to review patches =P [15:14:10 CET] <wbs> atomnuker: does the encoder already bring side data along from the AVFrame into the AVPacket? [15:14:32 CET] <wbs> otherwise, yes, that's a bigger and more complete framework for the same thing; reordered_opaque is the old lightweight mechanism [15:14:52 CET] <wbs> and even then, a codec cap would be needed if you wanted a caller to know whether it really works or not for each encoder [15:15:02 CET] <atomnuker> old lightweight? its a hack. [15:15:25 CET] <wbs> feel free to propose patches that does things differently, this has been on the ML for weeks [15:15:26 CET] <atomnuker> you're putting a random pointer in there [15:15:54 CET] <durandal_1707> dualz: just use min = max = ff_inlink_queued_samples [15:15:56 CET] <nevcairiel> its up to the user to specify the content of the field, avcodec does nothing of that sort [15:16:24 CET] <dualz> ok [15:16:39 CET] <atomnuker> wbs: I'd rather see this reverted and done properly [15:16:45 CET] <wbs> atomnuker: what "properly"? [15:16:49 CET] <atomnuker> side data [15:17:26 CET] <wbs> atomnuker: I don't stuff pointers into ints anywhere in that patch, even though that's one thing the caller of libavcodec _can_ do if they want to [15:17:46 CET] <wbs> it's the converse; libavcodec has got an opaque int64_t field you can use. x264 similarly has got a void* you can use [15:18:09 CET] <wbs> to cater for systems where sizeof(int64_t) > sizeof(void*) I keep the int64_t in a buffer in libavcodec and have x264 point to it [15:18:22 CET] <nevcairiel> you dont get to ignore the ML for weeks and then cry for a revert when it hits, you missed your chance, besides, this doesn't add any new API or bad hacks that haven't existed before, reordered_opaque is as old as avcodec [15:19:42 CET] <atomnuker> its a high bandwidth ML, pushing it and saying you can't change it anymore is ridiculous [15:20:42 CET] <wbs> if you contest the use of reordered_opaque here, you can convince nevcairiel and other external users to remove the mechanism altogether. this is just using the existing api for what it was meant for [15:22:39 CET] <atomnuker> sure its meant to be opaque but its a bit of a stretch to put a pointer in there [15:22:50 CET] <nevcairiel> thats the users doing [15:22:57 CET] <nevcairiel> avcodec does nothing l ike that or even suggests it [15:23:26 CET] <wbs> atomnuker: I don't put a pointer in there, where do you claim I'm doing that? [15:24:43 CET] Action: durandal_1707 grabs something... ML is not high bandwidth [15:27:40 CET] <atomnuker> wbs: erm right, nevermind then, the x264 commit message mislead me [15:27:57 CET] <atomnuker> nothing wrong here [15:28:33 CET] <cone-321> ffmpeg 03Mark Thompson 07master:a4fb2b115071: configure: Add missing IVF muxer BSF dependency [15:28:34 CET] <cone-321> ffmpeg 03Mark Thompson 07master:e9d2e3fdaacb: configure: Add missing V4L2 M2M decoder BSF dependencies [15:29:05 CET] <wbs> exactly; if our reordered_opaque would have been 32 bit only, we _could_ have considered squeezing it into x264's void* opaque, but we can't, hence the buffer [16:43:08 CET] <BBB> atomnuker: why would tristans patch be incompatible with simd? arent the two complementary? [16:43:25 CET] <BBB> atomnuker: his patch basically just adds a branch before calling the simd code, which you need anyway [16:46:42 CET] <cone-321> ffmpeg 03Valery Kot 07master:be827e1d38cb: libopenh264enc: Handle sample_aspect_ratio [17:49:41 CET] <atomnuker> BBB: why would you need the branch at all? [17:49:59 CET] <BBB> thats what the header bit does [17:50:08 CET] <BBB> when that header bit is set, the bitstream cdfs are non-per-symbol-adaptive [17:50:29 CET] <BBB> the cpu branch predictor should take care of making that efficient, Id expect [17:50:33 CET] <atomnuker> yes, but you can avoid the branch by just changing the function pointer to a function which does nothing [17:50:53 CET] <BBB> that means an extra call [17:50:57 CET] <BBB> vs. a predictable branch [17:51:00 CET] <BBB> seems unfavourable [17:51:02 CET] <BBB> right? [17:51:21 CET] <atomnuker> I'd disagree in this case [17:51:32 CET] <BBB> the extra call is slower, right? [17:52:02 CET] <atomnuker> optimally we'd like to have no function calls at all which means integrating decoding and updating into a single piece of asm [17:52:46 CET] <atomnuker> yes, its slower, but given how little overhead updating has with simd I'm not sure its even worth doing in pure asm [17:53:10 CET] <atomnuker> e.g. disabling adaptation [17:59:48 CET] <BBB> but the two approaches are sort of mutually incompatible, riht? [18:00:11 CET] <BBB> and youd still need a branch if you wanted the whole function in asm [18:13:03 CET] <atomnuker> for adaptation I'd say it would be unnecessary [18:14:52 CET] <atomnuker> as good as a branch predictor is I'd rather not rely on it unless there's no other choice [18:23:38 CET] <BBB> but an empty function call is worse [18:23:42 CET] <BBB> so what else do you propose then? [18:26:42 CET] <JEEB> rcombs: you make me feel dirty [18:26:50 CET] <rcombs> same [18:26:53 CET] <JEEB> I had to read through dolby documents [18:27:09 CET] <rcombs> the sample file I was looking at when I sent that seems to decode as normal-ass HEVC [18:27:20 CET] <JEEB> yea, it seems to depend on the profile [18:27:26 CET] <JEEB> for which they made additional boxes, of course [18:27:36 CET] <JEEB> I linked both specs (ISOBMFF and the general profiles= [18:27:46 CET] <rcombs> the colors look weird but idk it's an HDR sample, they always try to look ~vibrant~ [18:28:04 CET] <rcombs> it's not like it has a convenient set of color bars I can verify [18:28:24 CET] <rcombs> and even if it is decoding wrong, I'm gonna assert that something is better than nothing here [18:36:13 CET] <Gramner> atomnuker: indirect function calls are branches too as far as the branch predictis is concerned [18:36:39 CET] <Gramner> and indirect branches are predicted worse than normal ones since they can go anywhere, not just two ways [19:10:09 CET] <cone-321> ffmpeg 03Vittorio Giovara 07master:febaa63b0f7f: proresenc_anatoliy: Rename a profile name with the correct one [19:10:10 CET] <cone-321> ffmpeg 03Vittorio Giovara 07master:cdc487bfc841: prores: Use profile names in the various encoders and decoders [19:10:11 CET] <cone-321> ffmpeg 03Vittorio Giovara 07master:aefbb2bf029f: proresdec2: Parse codec_tag and export profile information [19:10:12 CET] <cone-321> ffmpeg 03Vittorio Giovara 07master:d37faad0cdbb: mjpeg: Use profile names in the encoder and decoder [22:04:33 CET] <dualz> durandal_1707: I got it working! I just need to test if the audio is falling out of sync or not [22:10:33 CET] <dualz> can anyone tell me when exactly and at what frequency activate() gets called? [22:36:03 CET] <cone-309> ffmpeg 03Ruiling Song 07master:952a299fd390: doc/filters: add document for opencl filters [22:36:03 CET] <cone-309> ffmpeg 03Ruiling Song 07master:a587454fd07c: doc/filters: add tonemap_opencl document. [23:53:58 CET] <durandal_1707> dualz: whenever is needed, there is set_ready to make it more frequent [23:55:11 CET] <michaelni> jamrial, are you done with the av1 fixes ? or there is more to wait for ? [23:56:00 CET] <jamrial> michaelni: two patches by jkqxz on the ml that i already reviewed need to be pushed and backported, and that's it [23:57:34 CET] <michaelni> jamrial, ok no hurry [23:59:32 CET] <jkqxz> Just getting that now. [00:00:00 CET] --- Tue Nov 6 2018

1 0

ffmpeg.log.20181105
by burek 06 Nov '18

06 Nov '18

[00:02:04 CET] <limbo_> furq: If you can suggest some settings, I'm all ears. Burning in subtitles optionally. I currently use mpv to do this (slowly) because everything else I tried fails on embedded fonts. (that aren't also on the system) [00:02:37 CET] <limbo_> Which is most of the re-encoding I do, most other things a PS4 can play. Even without remuxing out of mkv. [00:07:33 CET] <furq> i don't think attached fonts are supported by any of the hardsubbing filters in lavfi [00:07:45 CET] <furq> also i take it that you actually need the styling so you can't just use an external srt [00:07:51 CET] <kepstin> my impression was that ffmpeg can burn in subtitles including fonts if you use the "subtitles" filter. [00:07:56 CET] <kepstin> and give it the mkv file as input [00:08:11 CET] <furq> https://github.com/libass/libass/issues/266 [00:08:52 CET] <kepstin> huh. [00:08:57 CET] <nicolas17> IIRC in closed captioning on TV you get two (or more?) lines that scroll, when a new line appears, what used to be the last line scrolls up to make space [00:09:01 CET] <nicolas17> what subtitle format supports that? [00:09:24 CET] <furq> i've never actually tried it but the subtitles filter doesn't call ass_set_fonts after ass_process_codec_private [00:09:30 CET] <furq> so i assume it doesn't work [00:09:58 CET] <kepstin> nicolas17: you can fake it with most formats if your renderer has good timing, just have one sub say "line1 / line2" then the next "line2 / line3" [00:10:27 CET] <kepstin> nicolas17: in theory webvtt specifies this sort of thing, in practice i've haven't seen it implemented. [00:10:28 CET] <furq> also my favourite kind of bug report is one where one of the main contributors says "this should be easy to fix" and then nothing happens for two years [00:10:52 CET] <nicolas17> I think Apple's video player supports that in webvtt [00:11:26 CET] <furq> also i've seen that faked with srt and it was incredibly annoying [00:12:14 CET] <kepstin> I was playing around with trying that for a web video player caption generation tool, and just doing them normal subtitle style was way less annoying, yeah. [00:12:54 CET] <furq> it makes sense for live subtitling because the timing is screwed up there anyway [00:12:54 CET] <nicolas17> the truth is that I want to overlay a synced chat log on a video ^^ maybe I should render it myself and then overlay [00:12:59 CET] <furq> so you just want everything on screen as long as possible [00:14:41 CET] <furq> you could probably do something like that with ass [00:14:49 CET] <furq> but it sounds easier to just render and overlay [02:33:02 CET] <mixfix41> nicolas17: could subtitle editor work for tha? [02:33:13 CET] <mixfix41> that application does everything i need [02:33:59 CET] <mixfix41> oh with an srt file that is [05:27:09 CET] <mixfix41> its actually difficult for me to use handbrakecli but i think the gui is good [05:28:24 CET] <furq> handbrake is ok but make sure to never use decomb because it's terrible [05:35:04 CET] <mixfix41> heh heh [06:19:11 CET] <KombuchaKip> What is the equivalent to avformat_open_input(), but to open from a file loaded in memory? [06:21:28 CET] <nicolas17> I think that needs "custom IO" [06:21:36 CET] <KombuchaKip> nicolas17: Ok [06:22:04 CET] <nicolas17> documentation for avformat_open_input says "If you want to use custom IO, preallocate the format context and set its pb field." [06:22:14 CET] <xiando> I'm trying to into VP9 encoding and I tried: ffmpeg -i gyongree-171129_downpour.mp4 -threads 8 -c:a opus -b:a 192k -strict experimental -c:v libvpx-vp9 -crf 33 -b:v 0 test.webm <- while it makes a file it only uses one core and takes ages. [06:22:20 CET] <xiando> How can I into multi-core VP9 encoding? [06:22:51 CET] <furq> with difficulty [06:23:00 CET] <furq> i think -row-mt 1 should be all you need these days [06:23:07 CET] <furq> but if you have an older libvpx then it's a huge pain [06:23:40 CET] <KombuchaKip> nicolas17: Like so? https://stackoverflow.com/a/20610535 [06:26:30 CET] <xiando> furq: uhm.. -row-mt 1 did work and now it puts most cores under 20-30% load. instead of 0.64x I get 0.84x on that very low-resolution file. [06:26:47 CET] <furq> yeah libvpx has bad multithreadnig in general [06:28:45 CET] <furq> if it's low resolution then that's probably the reason [06:28:57 CET] <furq> it only works effectively on high res sources [06:29:58 CET] <furq> you might want to try explicitly setting -threads, idk if that's still necessary [06:30:02 CET] <furq> the docs for vpx are pretty bad [06:30:36 CET] <nicolas17> furq: the command he posted had -threads 8 [06:30:51 CET] <furq> so it did [06:30:59 CET] <furq> i guess it's just going to be slow then [06:31:01 CET] <xiando> tried a 1080p, puts a bit more load accross cores but not near 100%, more like 60%. Still, it's an improvement [06:31:15 CET] <xiando> speed is like 0.22x [06:32:06 CET] <furq> you can set -cpu-used if you want to make it fast [06:32:06 CET] <furq> er [06:32:09 CET] <furq> but it'll reduce the quality [06:33:46 CET] <xiando> I must have quality, I can't have blurry kpop [06:34:32 CET] <furq> i take it you're doing this to upload it somewhere [06:34:38 CET] <furq> otherwise i'd just leave it as h264 [06:34:38 CET] <KombuchaKip> nicolas17: Like so? https://stackoverflow.com/a/20610535 [06:35:21 CET] <nicolas17> I guess? I never did it [06:35:52 CET] <xiando> no my tablet can't have files bigger than 4 GB since it can't ext4 the SD card for some reason so I must reduce 6-7GB files to below 4 with quality. So I can use HEVC or VP9 [06:36:01 CET] <xiando> I'm not sharing. [06:36:16 CET] <KombuchaKip> nicolas17: Ok [06:37:11 CET] <nicolas17> exfat? [06:38:39 CET] <xiando> it's one that says it only supports 32 GB SD cards but if I format 64 GB cards as FAT they work fine [06:39:06 CET] <xiando> I guess I could buy a new one but I don't want to [06:39:22 CET] <nicolas17> how big is the SD card? [06:39:30 CET] <xiando> 64GB [06:39:42 CET] <nicolas17> did you try formatting it as exfat? [06:40:03 CET] <nicolas17> it should support files bigger than 4GB [06:40:51 CET] <xiando> I'm fairly sure the card was exfat when I got it. The tablet doesn't "support" cards bigger than 32 GB [06:41:00 CET] <xiando> which is why I had to mkfs.vfat it to make it work [06:41:07 CET] <xiando> it's a thing with older technology [06:41:32 CET] <nicolas17> and i guess you don't want to split it into two video files? [06:41:56 CET] <furq> yeah i would just split it [06:42:00 CET] <xiando> Now there's a simpler solution isn't it [06:42:19 CET] <furq> there's no way to reencode it to <4GB with good quality without spending days [06:42:42 CET] <furq> x265 is quicker than libvpx though fwiw [06:42:47 CET] <furq> since it actually has frame multithreading [06:43:12 CET] <furq> but it's still extremely slow at high quality settings [06:44:55 CET] <xiando> yeah it takes hours with x265, quicker than vp9 but splitting the file took seconds [06:45:08 CET] <nicolas17> uh how did you split it? [06:45:44 CET] <xiando> avidemux-qt, just opened it, selected first half, save, selected second half, saved that, done [06:46:31 CET] <xiando> file operations are quick on raid ssds [06:46:47 CET] <nicolas17> ah nice :D [06:46:51 CET] <nicolas17> I'm still on spinning rust [06:48:00 CET] <xiando> what, there's a big difference for opening programs and everything, just get a SSD as OS disc, seriously. prices are going down and will go down more the coming months due to oversupply [06:48:13 CET] <xiando> we're talking $30 for 240GB [06:54:15 CET] <furq> mkvmerge can split by size [06:54:25 CET] <furq> --split size:4g [06:55:02 CET] <nicolas17> I'd have done like [06:55:25 CET] <nicolas17> ffmpeg -i gyongree-171129_downpour.mp4 -t 1:00:00 -codec copy part1.mp4 [06:55:31 CET] <furq> there's no particularly nice way to do it with ffmpeg [06:55:39 CET] <furq> without working backwards from how long the file is [06:55:55 CET] <nicolas17> yeah, I'd have just split it in half by time :P [06:56:22 CET] <furq> also if you did do it with ffmpeg you should use the segment muxer [08:46:44 CET] <KombuchaKip> Could anyone with knowledge of custom AVIOContext usage kindly provide a code review? I have attempted to implement my own AVIOContext for reading media from memory instead of disk. I just want to make sure this is performed correctly. https://pastebin.com/KA7s4bJs [08:50:09 CET] <squ> what is return type of distance [08:50:58 CET] <squ> it is assigned to pointer to char in one function [08:51:05 CET] <squ> and compared to int64 in another [08:51:51 CET] <squ> if should have space [08:52:17 CET] <squ> pointer to char is compared with 0 [08:58:35 CET] <KombuchaKip> squ: Thank you. I will take a look tomorrow. [09:45:20 CET] <beauty> hello [09:45:27 CET] <beauty> can i connect [10:41:46 CET] <spaam> beauty: i think so. [12:07:39 CET] <gaara4896> quick question, how long does it take for you guys computer to compile ffmpeg library? And what is your spec? [12:16:56 CET] <beauty> two minutes [12:17:00 CET] <beauty> twenty [12:54:12 CET] <skident> Hi there [12:56:42 CET] <skident> I'm trying to use audio filters for audio stream (applying them on the fly). After using some of these filters I got cut sound (sounds like part of audio was lost). [12:57:19 CET] <skident> For example loudnorm work as I described above [12:59:41 CET] <spaam> gaara4896: ~90s on macbook pro with i9 and running with -j12 with only configure without any options. [13:09:52 CET] <gaara4896> spaam: That was fast [13:10:57 CET] <gaara4896> spaam: Do we need to specify -j12? I put only make, and it seems like it not utilizing all the core, and i have a 6 core processor too, and it took me quite a while [13:11:18 CET] <gaara4896> I read online says without putting anything will spawn unlimited worker [13:12:45 CET] <Ke> make -j is unlimited parallelism [13:12:58 CET] <TheAMM> make -j is webscale [13:13:00 CET] <Ke> I use make -j$(nproc) [13:13:29 CET] <Ke> but what is best heavily depends on circumstances [13:13:37 CET] <Ke> might be nproc + 1 [13:15:22 CET] <gaara4896> I change to just -j, and its so freaking fast now... [13:19:27 CET] <Ke> -j is known to cause problems on large compiles, like distro kernels [13:19:49 CET] <Ke> running out of memory etc. [13:22:54 CET] <gaara4896> Ke: Just some thought experiment, if you have threadripper with 4GB ram and doing -j$(nproc+1) might get you out of memory right? Haha [13:24:10 CET] <Ke> gaara4896: depends a lot on, what you are compiling [13:24:26 CET] <Ke> C++ with debug symbols is a lot heavier than C without [13:25:27 CET] <gaara4896> But not people gonna pair 32core processor with 4GB ram I guess [13:43:05 CET] <BtbN> If you have a threadripper and only install 4GB RAM you have failed setting your priorities. [13:43:32 CET] <BtbN> I usually aim for at least 4GB of RAM per core, ideally 8GB+ [14:12:06 CET] <dfhh> hello [14:12:15 CET] <dfhh> hhh [14:12:28 CET] <dfhh> can i connect [16:22:43 CET] <kishmesh> hi. when trying to parse an mp3 file I see a message "header missing" Format mp3 detected only with low score of 1, misdetection possible! [16:22:55 CET] <kishmesh> what could be wrong and how to fix this? [16:24:25 CET] <kishmesh> basically this issue https://superuser.com/questions/827970/ffmpeg-mp3-file-header-restore [16:25:58 CET] <kishmesh> als this bug it seems https://trac.ffmpeg.org/ticket/5689 [16:30:59 CET] <kishmesh> similar with the same file as m4a I get "moov atom not found" [16:43:22 CET] <limbo_> kishmesh: you're missing the metadata section of the file, with mp4 files it's at the end. Either that, or it's not actually the format you think it is. [17:05:02 CET] <kishmesh> limbo_: thank you I figured it out. it was a file format problem. [20:21:13 CET] <puchu> hi [20:21:37 CET] <puchu> how can i convert an audio file (aac 2ch 44100Hz) to aac 2ch 48000Hz [20:23:32 CET] <kepstin> puchu: in general, don't - the lossy transcode will lose quality and that sort of resampling can normally be done by the player if needed. But if you have to, you can use the "-ar" output option in ffmpeg to tell it to resample when re-encoding. [20:24:01 CET] <puchu> kepstin: i want to merge two files with mkvmerge but i get [20:24:10 CET] <puchu> The sample rate of the two audio tracks is different: 48000 and 44100 [20:24:39 CET] <puchu> kepstin: so i have to change it to get i merged. or do you have another idea to force mkv to merge it [20:25:24 CET] <puchu> kepstin and where can i find a would documentation for ffmpeg? the man page seems to miss a lot of stuff [20:25:37 CET] <puchu> like list all available codecs with options [20:27:59 CET] <puchu> ahh found it ffmpeg -codecs [20:29:45 CET] <kepstin> adjusting the sample rate to allow concatenating files is a case where it makes sense [20:30:46 CET] <kepstin> ffmpeg has a builtin aac encoder (named "aac") that works ok, but if you have a build with fdk_aac available it'll probably give better quality at the same size [20:41:16 CET] <puchu> kepstin: thanks [20:41:37 CET] <puchu> kepstin is there a way to list all available options for a specific codec? [20:43:02 CET] <kepstin> there's a bunch of generic options available that apply to many codecs, e.g. bitrate [20:43:11 CET] <kepstin> and some codec specific ones [20:43:19 CET] <kepstin> no general way to get the full list [20:43:50 CET] <kepstin> for aac, just set a bitrate (-b:a) and that's probably enough [20:44:42 CET] <puchu> sucks a bit that there is no way to get a full list [20:44:45 CET] <puchu> thanks for your help [20:45:55 CET] <kepstin> for codec specific options, use " ffmpeg -h encoder=<codecname>" [20:46:07 CET] <kepstin> for general options,... read the ffmpeg man page :/ [20:50:50 CET] <puchu> kepstin: thanks [22:16:14 CET] <No0n3Left> Anyone have suggestions on converting between h264 in a mkv and theora in an ogg? Got the audio part fine, as its just copying an opus stream from the mkv, but the video conversion is quite horrid. Not only did it take 10 minutes for a 4 minute video (which isnt horrid really), it just doesnt render correctly. There are so many visual glitches from the conversion, and lots of laggy frames. The original [22:16:22 CET] <No0n3Left> h264 was 60fps, and mpv says the theora is also 60fps, but it looks more like a 15fps minimum quality jpeg slideshow. I just used the flags -vcodec theora -acodec copy, any suggestions on how to get better quality? [22:17:38 CET] <JEEB> No0n3Left: the theora encoder is notorious for dropping frames when it can't hit your bit rate [22:20:33 CET] <nicolas17> and you didn't set a bitrate [22:20:41 CET] <nicolas17> so maybe it's doing 200kb/s or something stupid [22:20:59 CET] <JEEB> the default in AVCodecContext for bit rate is indeed 200kbps for hysterical raisins [22:21:05 CET] <No0n3Left> It is, what bitrate would you suggest? Dont know much about bitrate [22:21:15 CET] <JEEB> or you just use some quantizer [22:21:20 CET] <JEEB> if theora supports that [22:28:10 CET] <dualz> Using anullsrc and testsrc as my inputs I'm getting very high FPS in my output. I set rate on testsrc to 30 and still no avail [22:28:13 CET] <dualz> https://gist.github.com/ryanmarin/7adbfec209c359296eb5dbfb4df96f9e [23:19:21 CET] <kepstin> yeah, just don't use theora if at all possible [23:19:40 CET] <kepstin> pretty much no reason to nowadays, vp8 (or even 9) is almost always a better choice [00:00:00 CET] --- Tue Nov 6 2018

1 0