December 2018 - Ffmpeg-devel-irc

ffmpeg-devel.log.20181225
by burek 26 Dec '18

26 Dec '18

[00:45:30 CET] <cone-636> ffmpeg 03Michael Niedermayer 07master:7efe84aebd1f: avcodec/lagarith: Remove duplicate check [12:43:10 CET] <j-b> Merry 'Xmas, mates! [17:01:57 CET] <kierank> Merry Xmas [19:24:30 CET] <cone-376> ffmpeg 03Paul B Mahol 07master:8d0b8c50bd23: avfilter/af_sofalizer: speed up fast convolution [19:24:30 CET] <cone-376> ffmpeg 03Paul B Mahol 07master:6c7eb0708e65: avfilter/af_headphone: speed up fast convolution [19:41:38 CET] <durandal_1707> Compn: hello! [20:21:49 CET] <cone-376> ffmpeg 03Marton Balint 07master:d0e43cf1a89c: avformat/mxfdec: replace obsolete comment [20:21:50 CET] <cone-376> ffmpeg 03Marton Balint 07master:02935c2c93e8: avformat/mxfdec: support subsecond precision of decoded timestamps [20:21:51 CET] <cone-376> ffmpeg 03Marton Balint 07master:70c86deb8efe: avformat/mxfenc: support writing subsecond precision timestamps [20:25:24 CET] <atomnuker> why are people still encoding mxf files? [20:28:18 CET] <durandal_1707> atomnuker: to make you unhappy [00:00:00 CET] --- Wed Dec 26 2018

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ffmpeg.log.20181225
by burek 26 Dec '18

26 Dec '18

[00:51:57 CET] <luddite478> Hi, can suggest me the way to pass ffmpeg stream through my python script with opencv? Details here: https://stackoverflow.com/questions/53906765/pass-ffmpeg-stream-through-pyt… [08:21:03 CET] <Trilom> Greetings, looking to see if anyone is interested in taking a crack at something. I am following the compile guide at https://trac.ffmpeg.org/wiki/CompilationGuide/Centos for Centos7 and keep running into the same issue when I attempt to build FFMPEG after building all deps. Here is the error "STDOUT: ERROR: freetype2 not found using pkg-config" It is very similar if not identical to: https://superuser.com/questions/1366977/install-ffm [08:25:57 CET] <pink_mist> I mean ... did you try the obvious thing? installing freetype2? [08:28:09 CET] <Trilom> Yes, similar to the linked superuser post, pkg-config --exists freetype2 || echo 'no lib' shows freetype2 exists, rpm -q freetype shows freetype-2.8-12.el7.x86_64, and --cflags are set for pkg-config to -I/usr/include/freetype2 -I/usr/include/libpng15 [08:30:34 CET] <pink_mist> well [08:30:39 CET] <pink_mist> you're on centos [08:30:54 CET] <pink_mist> so having freetype installed doesn't mean you have all of freetype installed [08:31:02 CET] <pink_mist> you'll need freetype-devel too [08:31:31 CET] <Trilom> Here ya go: [vagrant@GS-l-centos-74 ~]$ rpm -q freetype-devel freetype-devel-2.8-12.el7.x86_64 [08:31:43 CET] <Trilom> Allegedly both should be there. [08:32:03 CET] <Trilom> I even tried downgrading to a lower version of freetype 2.4 or something. [08:33:11 CET] <Trilom> The compilation guide says to install freetype-devel, which freetype depends on. [08:33:35 CET] <pink_mist> no idea then ... ffmpeg compiles fine on my slackware machine [11:07:05 CET] <scramblez> Happy Xmas to all! I'm trying to concatenate 4 avi files but without the audio of the first file. So, I first used -na to remove the audio stream of the first avi file and then concatenated them. However, the result is a file with no audio stream. :( How could I fix this? [11:07:43 CET] <durandal_1707> full uncut ffmpeg command and output missing [11:15:48 CET] <scramblez> Thanks durandal_1707, here it is: https://pastebin.com/nrJT2Jbc [11:17:24 CET] <durandal_1707> please do not ever use concat protocol [11:17:53 CET] <scramblez> Ohh! What should I be using instead? [11:18:12 CET] <durandal_1707> use concat demuxer, and make sure that parameters like width/height and pixel format match [11:18:53 CET] <durandal_1707> http://trac.ffmpeg.org/wiki/Concatenate#demuxer [11:19:59 CET] <scramblez> Thanks, I think this is how I had done it last time I tried the same - I can't recall having problems back then. Will give it another go. [11:29:12 CET] <scramblez> Hmm ... still no audio, although it seems to find the audio stream from the second file onward: https://pastebin.com/Y3rqr3uN [11:30:37 CET] <scramblez> BTW, all video files parameters are the same, other than audio being removed from the first. [11:52:02 CET] <durandal_1707> you can not expect to have audio if you removed it from first [12:04:06 CET] <scramblez> durandal_1707: is it then possible to zero the volume on the first video's audio stream so it is not audible? This way all files will have their audio stream. [12:09:48 CET] <durandal_1707> scramblez: if you encode silence with same parameters as other files with same audio codec [13:04:35 CET] <scramblez> THanks durandal_1707, will give it a go later and see what I get. [13:14:10 CET] <bmcginty> I've tried searching for this with no luck. Is there a way to have ffmpeg fail to convert a file if it doesn't have at least one audio stream? (I know there's -map for selection, but it doesn't _require a stream type to be present I believe.) [14:44:09 CET] <JEEB> bmcginty: I think unless you specify a special thing with map it will fail you if there is nothing matching that [14:47:59 CET] <bmcginty> JEEB: I think I have to specify -map 0 because I basically want to copy everything except video streams, and I want to make sure there's at least one audio track. [14:49:19 CET] <JEEB> right, that is non-specific enough to not fail unless you have zero streams :P [16:40:26 CET] <kingmanor> is there a way to limit ffmpeg cpu usage without running it in a vm? [16:41:10 CET] <JEEB> depends on your OS [16:41:25 CET] <JEEB> linux for example has cgroups and both linux and windows has pinning to specific cores [16:48:25 CET] <kingmanor> i have a windows box rebooting in the middle of a long ffmpeg transcode. i tried it running the same job in a linux vm with 94% cpu entitlement and still happening. [16:49:01 CET] <durandal_1707> kingmanor: post full ffmpeg command? [16:49:49 CET] <kingmanor> ffmpeg -loglevel verbose -i gb_wintergames_2018.mp4 -c:v libx265 -c:a copy -map_metadata -1 -report gb_wintergames_2018.mkv [16:50:00 CET] <kingmanor> its a 2 hour high bitrate 1080 mp4 file [16:50:08 CET] <kingmanor> converting to hevc [16:50:26 CET] <kingmanor> using latest zeranoe windows build [16:50:30 CET] <durandal_1707> how much RAM you have? [16:50:38 CET] <kingmanor> 8gb [16:51:12 CET] <durandal_1707> wait, only vm reboots? [16:51:21 CET] <kingmanor> no the whole box reboots [16:51:27 CET] <durandal_1707> that that is hw issue [16:52:07 CET] <kingmanor> yes i am asking if there is a way to limit cpu without running in a vm that is capped at less mhz [18:04:00 CET] <ultrav1olet> I have a number of photos with varying sizing and aspect ratios and I want to get a video from them however I'm not sure what to do [18:04:19 CET] <ultrav1olet> I've resized them so that their maximum dimensions are either 1920 or 1080 [18:04:35 CET] <ultrav1olet> However when I try to encode them, ffmpeg resizes them and makes them all full screen [18:04:50 CET] <ultrav1olet> which breaks the aspect ratio completely and makes them look horrible [18:05:13 CET] <ultrav1olet> I've tried all the examples on the net and none of them work [18:05:44 CET] <ultrav1olet> my first image is black 1920x1080 or otherwise ffmpeg f*cks everything up [18:06:11 CET] <ultrav1olet> I now need to concatenate images without resizing them at all [18:09:12 CET] <ultrav1olet> Tried this https://stackoverflow.com/questions/44634765/ffmpeg-aspect-ratio-of-image-i… https://stackoverflow.com/questions/38937136/ffmpeg-different-sized-images-… https://superuser.com/questions/547296/resizing-videos-with-ffmpeg-avconv-t… [18:09:16 CET] <ultrav1olet> nothing works [18:09:20 CET] <ultrav1olet> damn it [18:57:03 CET] <kepstin> if you're trying to put photos into a 16:9 1080p video preserving aspect ratio, then you're going to have to pad them (ffmpeg 'pad' filter) to add bars to the sides. [18:58:09 CET] <kepstin> something like 'pad=1920:1080:(ow-iw)/2:(oh-ih)/2' should do the trick, assuming the pictures have already been resized [18:58:25 CET] <kepstin> er, wait, they already left [18:58:28 CET] <kepstin> ... :/ [21:24:38 CET] <docmax> whats the command line to copy video to udp://0.0.0.0:1234 AND play audio locally? [21:36:45 CET] <docmax> anyone? [00:00:00 CET] --- Wed Dec 26 2018

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ffmpeg-devel.log.20181224
by burek 25 Dec '18

25 Dec '18

[00:08:40 CET] <atomnuker> custom channel order? so codecs can just signal what the order is rather than reorder themselves? that'd be nicie [00:08:43 CET] <atomnuker> *nice [00:10:10 CET] <durandal_1707> that would need code that would do actual reorderring [06:26:31 CET] <cone-177> ffmpeg 03Steven Liu 07master:cdbf8847ea97: avformat/hlsenc: remove duplicate operation at hls_write_header [09:45:05 CET] <durandal_1707> so did I got any more comments about new channel layout API? [10:19:27 CET] <pross> avcodec_decode_audio5? [10:45:24 CET] <durandal_1707> no [11:44:23 CET] <kurosu> pross, avcodec_decode_audio5.1, just to be confusing (ignore me) [11:45:49 CET] <kurosu> (obviously 5_1, but being correct or relevant was not the point) [11:52:08 CET] <pross> this change wont break existing tests, except for those codecs that are being upgraded to take advantage of the new abi/api? [12:03:24 CET] <durandal_1707> pross: idea is to add new fields: int channel_order and int *channel_map instead of replacing channel_layout and channels with new struct [12:04:17 CET] <durandal_1707> with new struct one would need to touch and modify more than 100 files [12:12:41 CET] <pross> understood. seems reasonable, no objections here [13:21:06 CET] <nevcairiel> I don't see how a new structure would require any of that, as long as its side by side with the old value. I don't like separate lose values that have to be interpreted together, and if we define a new API, it should be absolutely future proof [13:22:54 CET] <durandal_1707> nevcairiel: using structure require touching all files that previously used old channels and channel layout [13:24:38 CET] <atomnuker> durandal_1707: any patches anywhere [13:25:10 CET] <atomnuker> and are you seriously adding a new decode function using the old-style api? [13:25:18 CET] <durandal_1707> Vittorio patches: https://github.com/kodabb/libav/commit/c023b553e6ad6da5af6d0d4ff067ff844b2f… [13:25:30 CET] <durandal_1707> atomnuker: no, that pross made up [13:58:41 CET] <nevcairiel> durandal_1707: no it does not, just keep the old value as well and then fill the new one from that if the Decoder didn't use it. Voila only code that wants to use it has to [14:03:33 CET] <nevcairiel> Two years time to eventually migrate all of it when the old one is removed. Also lazyness is no excuse for hacky api design [14:33:19 CET] <durandal_1707> its not hacky, my approach is perfect [14:34:42 CET] <atomnuker> I agree with nevcairiel [14:34:45 CET] <JEEB> I wish you a happy christmas eve, but could you at least argument yourself since clearly some people think there are better ways of going around with it [14:35:36 CET] <durandal_1707> you are perfect candidate to touch all those 500 files? [14:36:08 CET] <JEEB> what? [14:36:51 CET] <durandal_1707> to update to new ch_layout API [14:39:31 CET] <atomnuker> can't we leave some wrapper layer to convert such that we won't have to touch all codecs? [14:39:43 CET] <JEEB> ^ this is what was my 1st thought [14:39:51 CET] <durandal_1707> and filters and demuxers and muxers [14:41:07 CET] <durandal_1707> you can keep channel_layout and channels but ultimately you need conversion with ch_layout API [14:57:33 CET] <durandal_1707> s [15:02:42 CET] <durandal_1707> so you guys like ch_layout struct and want to use it instead of channels/channel_layout ? [15:09:05 CET] <durandal_1707> ? [15:16:31 CET] <durandal_1707> if not i will use my approach [15:28:55 CET] <cone-636> ffmpeg 03Paul B Mahol 07master:84d1adb118a7: avfilter/af_sofalizer: do not reduce LFE by 6dB [15:28:56 CET] <cone-636> ffmpeg 03Paul B Mahol 07master:3bc711a2675a: avfilter/af_headphone: do not reduce LFE gain too much [16:34:15 CET] <nevcairiel> I like a comprehensive solution that is able to either represent anything imaginable, or is extensible in the future, and yet remains sane to use, not some quick and dirty thing for one or two usecases [16:35:15 CET] <nevcairiel> In any case any API changes need to be discussed on the ML, and not I er Christmas when no one is around [16:46:50 CET] <atomnuker> true, can't miss exclusive events in mmos [16:52:07 CET] <durandal_1707> atomnuker: what events? [16:53:28 CET] <durandal_1707> i will commit it ASAP! [17:08:41 CET] <BradleyS> i'd pay a dollar to see that [17:08:46 CET] Action: BradleyS runs [17:50:02 CET] <kierank> durandal_1707: can you remove ndi [17:52:30 CET] <durandal_1707> kierank: nope, I'm the leader of FFmpeg [18:23:37 CET] <durandal_1707> should i keep 5.1(side) instead of 5.1(back) ? [18:25:35 CET] <atomnuker> keep both? they're both common, with side being a standard and back being a common mistake [18:26:27 CET] <durandal_1707> i will keep old behavior so fate stays green [20:59:14 CET] <cone-636> ffmpeg 03Paul B Mahol 07master:d6fc20a3ba17: avfilter/af_surround: fix code indentation [21:05:10 CET] <BradleyS> ^ that wasn't such a big change [21:05:43 CET] <BradleyS> durandal_1707: are you going to push chaos or do i get my dollar back [21:08:48 CET] <durandal_1707> BradleyS: it is on mailing list [21:09:11 CET] <BradleyS> mailing list is not enough chaos [21:10:14 CET] <BradleyS> o/ happy holidays [21:10:44 CET] <durandal_1707> there are no holidays here, only hard work on ffmpeg [21:12:15 CET] <BradleyS> as long as you enjoy it [21:13:26 CET] Action: durandal_1707 only enjoys NAKing Carl patches [21:43:06 CET] <JEEB> TIL SCTE-18 is just hard-coded PIDs [21:43:12 CET] <JEEB> not even mentioned in the PMT [21:44:19 CET] <kierank> JEEB: lol [21:44:32 CET] <kierank> JEEB: aint got time for PMT during emergency [21:46:55 CET] <JEEB> that kind of does make sense, although before re-reading the spec I did browse the PMT, and what I first thought was the SCTE-18 stream ended up being just HbbTV [21:47:24 CET] <JEEB> although I would expect such emergency alerts getting retransmitted quite a few times [21:49:35 CET] <JEEB> I think Japan has its own emergency alert system in broadcast, but I've never seen it action. it's always just broadcasters' own overlays on top of the video [21:49:48 CET] <JEEB> that way there's no way some receiver isn't showing it [22:17:01 CET] <cone-636> ffmpeg 03gxw 07master:f652c7a45c60: avcodec/mips: Fix failed case: hevc-conformance-AMP_A_Samsung_* when enable msa [22:21:16 CET] <cone-636> ffmpeg 03Rene Claus 07master:6a8cc8696339: avcodec/libvpxenc: add VP8/9 sharpness config option [00:00:00 CET] --- Tue Dec 25 2018

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ffmpeg.log.20181224
by burek 25 Dec '18

25 Dec '18

[01:59:55 CET] <RecursiveCollaps> test [01:59:58 CET] <RecursiveCollaps> Cool [02:00:41 CET] <RecursiveCollaps> Hey so ffmpeg has been causing some Problems(TM) for me. Crashing problems. I have a script set to use ffmpeg to convert a bunch of videos, but while this was running, any other application that uses any kind of video output would crash completely at random, with an error message from ffmpeg popping up at the same time. [02:00:49 CET] <RecursiveCollaps> I am not particularly interested in risking my computer in attempting to reproduce this because it ended with my video driver completely crashing and being forced to reboot and lose work. [02:01:36 CET] <RecursiveCollaps> Errors were just a popup box in front of everything that said something about ffmpeg 'failed to identify' and included a memory address [02:01:59 CET] <RecursiveCollaps> Any help with this would be greatly appreciated [02:32:58 CET] <klaxa> ffmpeg should not open any popup boxes [02:45:23 CET] <RecursiveCollaps> Well it did, or at least did indirectly, the popup box was identical no matter what program crashed. [02:45:47 CET] <RecursiveCollaps> Also the popup box said "ffmpeg:" in the header [02:47:01 CET] <RecursiveCollaps> Anything that rendered any images on the screen was also pereodically interfered with, any kind of rendering would glitch out and fail to draw for a few moments on occasion while it was running [03:57:59 CET] <Keshl> Okay, I have three computers, three different CPUs, all from vastly different time periods. All i7s, some with four physical cores, some with six. Some dual-channel memory, some quad. They're all pulling from different storage devices that only they have access to. Somehow, ffmpeg is running at the same speed on all of them. My command line is ffmpeg -i "Original.mkv" -vcodec libx265 -preset medium -crf 28 -movflags faststart -pix_fmt [03:58:00 CET] <Keshl> yuv420p -acodec aac -q:a 2 -map 0 -vf mpdecimate=max=0:hi=61*12:lo=64*5:frac=1 -af aresample=async=1:min_hard_comp=0.100000:first_pts=0 "Archive.mkv" . What could the bottleneck be? [07:21:28 CET] <NorthwestVegan> #chan #raspberrypi [07:22:02 CET] <NorthwestVegan> oops sorry guys [07:22:15 CET] <NorthwestVegan> ffmpeg is the best by the way [00:00:00 CET] --- Tue Dec 25 2018

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ffmpeg-devel.log.20181223
by burek 24 Dec '18

24 Dec '18

[08:21:21 CET] <cone-217> ffmpeg 03Gorilla Maguila 07master:bb586a970327: avformat/dashdec: refine uri path with double dot process [10:07:09 CET] <cone-217> ffmpeg 03Paul B Mahol 07master:b13fe6477d5b: avfilter/af_sofalizer: increase range for lfegain [11:05:09 CET] <cone-217> ffmpeg 03Paul B Mahol 07master:0c3481b43a19: avfilter/af_sofalizer: pick IR length after loading sofa [11:32:24 CET] <cone-217> ffmpeg 03Paul B Mahol 07master:d360a79c0a1e: avfilter/af_sofalizer: reset FFT pointers back to NULL [15:44:50 CET] <cone-281> ffmpeg 03Paul B Mahol 07master:3601eb047456: avcodec: add g732_1 parser [15:44:50 CET] <cone-281> ffmpeg 03Paul B Mahol 07master:f52dd8a55a98: avcodec/g723_1dec: use init_get_bits8() [18:53:45 CET] <cone-855> ffmpeg 03Paul B Mahol 07master:4096c670abef: avfilter/af_sofalizer: stop using easy API [20:25:49 CET] <cone-855> ffmpeg 03Michael Niedermayer 07master:fbf409cd91ac: avcodec/ilbcdec: fix integer overflow in energy [20:25:50 CET] <cone-855> ffmpeg 03Michael Niedermayer 07master:80cce5998c10: avcodec/ivi: Avoid mbs memleak [20:25:51 CET] <cone-855> ffmpeg 03Michael Niedermayer 07master:30a7a81cdc2e: avcodec/fic: Fail on invalid slice size/off [20:34:51 CET] <durandal_1707> michaelni: learn to use: for (_int_ t = 0; ... ) NOW! [20:51:16 CET] <cone-855> ffmpeg 03Paul B Mahol 07master:f89919d4fb7c: avcodec/fic: use init_get_bits8() [21:03:19 CET] <cone-855> ffmpeg 03Paul B Mahol 07master:e483606d4455: avcodec/g722dec: use init_get_bits8() [21:17:12 CET] <cone-855> ffmpeg 03Paul B Mahol 07master:c4a05ae65c39: avcodec/wmavoice: use init_get_bits8() [21:43:26 CET] <cone-855> ffmpeg 03Paul B Mahol 07master:ee64b64c0227: avfilter/af_sofalizer: set delays when interpolation is disabled [21:50:52 CET] <cone-855> ffmpeg 03Paul B Mahol 07master:7d5bb3a4d34f: avfilter/af_sofalizer: fix memory leaks [22:06:50 CET] <durandal_1707> why are our users so ignorant that they do not even read manuals - we should punish them! [22:11:34 CET] <kierank> durandal_1707: stop being mean [22:12:48 CET] <durandal_1707> it can't stop it [23:39:15 CET] <durandal_1707> i'm looking into adding custom channel layouts support but this would cause gross changes and mayor bump [23:40:33 CET] <durandal_1707> there is less intrusive way but at cost of uglier negotiation in lavfi [23:41:35 CET] <durandal_1707> what you guys think? [23:42:58 CET] <durandal_1707> no comments or you guys are already sleeping? [23:43:57 CET] <BradleyS> by custom channel layouts, you mean custom mappings user-definable at run time? [23:44:38 CET] <durandal_1707> ambusonics and others, not limited to 64bit mask [23:44:48 CET] <BradleyS> filtergraph-like [23:46:08 CET] <durandal_1707> i do not want to break compatibility with old code [23:47:55 CET] <durandal_1707> i gonna just add 2 new fields: channel_order and channel_mapping, last one being array with size 4 * number of channels [23:50:34 CET] <jamrial> durandal_1707: there's a patchset by vittorio that introduces a new channel layout api [23:51:06 CET] <durandal_1707> filtering negotiation would be combination of arrays [23:51:46 CET] <durandal_1707> jamrial: that api breaks everything [23:52:48 CET] <durandal_1707> i would need to touch all decoders and filters that touch channel map [23:52:49 CET] <jamrial> afaik, a compat layer for the deprecation period is what was missing, yes [23:53:44 CET] <durandal_1707> compat layer not possible, it changes Api and abi [23:56:43 CET] <durandal_1707> with my approach only new code that needs ambisonic and custom layout will set channel order and optionally mapping for custom ordering [00:00:00 CET] --- Mon Dec 24 2018

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ffmpeg.log.20181223
by burek 24 Dec '18

24 Dec '18

[03:53:33 CET] <kode54> is there an easy way to probe whether an HEVC video is either 8 or 10 bit? [03:59:08 CET] <kode54> ah, it says "Main" not "Main10" [06:03:27 CET] <fengshaun> I have transcoded an mp4/h264/aac file with ffmpeg libav api, but the resulting video has fps of 12 (transcoded) instead of 24 (original video), I haven't changed any parameters between the decoding and encoding context [06:04:03 CET] <fengshaun> except height, width, simple_aspect_ratio, time_base, and pix_fmt [06:04:28 CET] <fengshaun> time_base = av_inv_q(decoding_context->framerate) [06:04:54 CET] <fengshaun> what am I missing as far as fps is concerned? [06:05:00 CET] <fengshaun> I'd like to keep the original fps [06:06:55 CET] <fengshaun> as far as I understand, time_base is reciprocal of framerate, so in this case, the framerates of encoding_context and decoding_context (AVCodecContext *) should be the same [06:23:56 CET] <fengshaun> hmm looks like I can just set the encoding and decoding AVCodecContexts' time_bases the same and the resulting transcoded file seems to play fine [06:24:37 CET] <fengshaun> not sure if there is a reason to set the encoder's time_base to inverse of decoder's framerate instead of setting the encoder's time_base to that of decoder's time_base [07:52:53 CET] <mixfix41> 5~ [12:14:43 CET] <Alina-malina> after using amix and adelay the sound become so so bad, bearly can hear it, is there a way to make sound volume up? [12:16:49 CET] <durandal_1707> Alina-malina: full uncut console ffmpeg output missing [12:17:20 CET] <durandal_1707> you can use volume filter.... [12:17:28 CET] <Alina-malina> ok just a moment let me show you [12:18:16 CET] <Alina-malina> durandal_1707, https://pastebin.com/raw/vASLuVyP i know i should add the volume parameter, but i am not sure where [12:20:39 CET] <durandal_1707> Alina-malina: what are you attempting to do with so many adelay filters? [12:21:01 CET] <Alina-malina> mixing specific sound to that times [12:21:03 CET] <Alina-malina> why? [12:21:28 CET] <Alina-malina> special effects [12:21:29 CET] <Alina-malina> lol [12:22:11 CET] <durandal_1707> wrong approach, amix will reduce volume even when there is silence in most of inputs [12:22:49 CET] <Alina-malina> well thats from yesterday discussion [12:23:00 CET] <Alina-malina> what is the right approach then? [12:23:14 CET] <Alina-malina> i mean both mp3 files i can barely hear [12:23:25 CET] <Alina-malina> not 1 but both are bad very bad [12:23:48 CET] <durandal_1707> are you concating and fading or looping them or doing some special mixing? [12:24:27 CET] <Alina-malina> well the kaching.mp3 file is a 1 second sound of money [12:24:34 CET] <Alina-malina> the first mp3 file is a music background [12:24:45 CET] <Alina-malina> so i am applying that sounds to my game [12:24:53 CET] <Alina-malina> game is soundless [12:25:01 CET] <Alina-malina> so i am adding sound effects to game [12:25:30 CET] <Alina-malina> so game.mp4 sound background.mp3 and kaching.mp3 when get achievement [12:25:37 CET] <durandal_1707> you better use some specific library inside game directly [12:25:49 CET] <Alina-malina> well ye, that might work too [12:25:54 CET] <Alina-malina> i can write that [12:25:57 CET] <Alina-malina> but still [12:26:01 CET] <Alina-malina> need this first [12:26:09 CET] <durandal_1707> openal comes to mind [12:27:55 CET] <Alina-malina> do i need to add volume=0.2 to after each adelay? or i can do that in the very end of that string? [12:28:32 CET] <durandal_1707> volume=0.2 will make it even more silent [12:29:31 CET] <durandal_1707> you can use volumedetect filter first to measure output volume [12:30:10 CET] <durandal_1707> and after measuring replace volumedetect filter with volume filter to get final output [12:30:59 CET] <Alina-malina> i am not sure where to apply that volume filter, i am getting error when i try to add it [12:31:12 CET] <durandal_1707> at end [12:31:45 CET] <durandal_1707> before [aout] [12:32:09 CET] <durandal_1707> ... amix=inputs=51:duration=first:dropout_transition=32,volume=50[aout]" [12:32:34 CET] <Alina-malina> ok let me try that [12:33:17 CET] <Alina-malina> ye nice nice it works!! [12:33:20 CET] <Alina-malina> finaly [12:33:21 CET] <Alina-malina> lol [12:33:26 CET] <Alina-malina> thanks durandal_1707 <3 <3 <3 [13:59:39 CET] <__gh0st> Hi, how can I get the pict_type of a frame(access unit) from h.264 bitstream using ffmpeg api? Currently I am using av_read_frame to read packets into AVPacket struct and I need to get the pict_type of the frame, without decoding it. [14:00:26 CET] <__gh0st> Are there any facilities that ffmpeg api has to derive such information from AVPacket struct? [14:01:51 CET] <__gh0st> Or I need to manually look for pict_type using some offset from AVPacket.pos to look for pict_type in the stream? [14:06:28 CET] <__gh0st> There is a h.264 parse in ffmpeg, but I not entirely sure how to use it in my case. [15:21:49 CET] <Alina-malina> durandal_1707, https://www.youtube.com/watch?v=AvtCyRshAdk&feature=youtu.be in case if you interested what i was making bro, anyways its beginner steps, dont expect pro video there lol [18:20:50 CET] <oxek> this video plays horribly choppy https://send.firefox.com/download/42e7d21f5f/#J-JyN3vEvb6SVbj2yZKGxA [18:27:23 CET] <poutine> good to know [18:27:59 CET] <analogical> how do I use FFmpeg to trim a couple of seconds from a video file without re-encoding the video? [18:29:19 CET] <poutine> analogical, ffmpeg -ss 00:00:03 -i file -c copy out.mp4, potentially -t for duration based on where you're trimming it from, timing will be inexact will only cut on i-frames afaik [18:29:49 CET] <pink_mist> yeah, if you need more exact than i-frames, you need to re-encode I believe [22:36:39 CET] <Bertl_oO> greetings! I'm currently trying to record an RTSP stream as MP4 or MKV container [22:37:10 CET] <Bertl_oO> now the stream contains a data stream besides the video/audio streams and everything works fine if I drop the data with -dn [22:37:50 CET] <Bertl_oO> the question is, can I somehow store the data stream in the MP4 or MKV container as well, and if so, how? [23:07:03 CET] <kode54> looks like oxek's video is indeo 4.1 [23:07:13 CET] <kode54> good luck fixing that, ffmpeg [23:07:40 CET] <kode54> it appears to be arranging the frames in the wrong order [00:00:00 CET] --- Mon Dec 24 2018

1 0

ffmpeg-devel.log.20181222
by burek 23 Dec '18

23 Dec '18

[00:16:12 CET] <cone-779> ffmpeg 03Carl Eugen Hoyos 07master:6a87729e9da5: lavc/opus_rc: Cast a const pointer to uint8_t *. [01:26:01 CET] <cone-779> ffmpeg 03James Almer 07master:53d3a1c514bf: APIChanges: add missing entry for SMPTE2094-40 HDR Dynamic metadata API [08:15:12 CET] <pross> how to describe that feeling you get when your decoder output matches the reference implementation for the first time? [08:15:40 CET] <BradleyS> i believe the technical term is squee [08:16:08 CET] <BradleyS> congrats [11:16:24 CET] <cone-641> ffmpeg 03Paul B Mahol 07master:297e65c676e3: avformat/vividas: check if value from ffio_read_varlen() is too big [11:16:24 CET] <cone-641> ffmpeg 03Paul B Mahol 07master:14c2f7f65260: avformat/vividas: fix undefined behaviour [11:16:24 CET] <cone-641> ffmpeg 03Paul B Mahol 07master:e9909fe19418: avformat/vividas: check if size is enough big in xor_block [13:37:42 CET] <cone-641> ffmpeg 03Paul B Mahol 07master:d62cb29716a5: avformat/vividas: check if viv->sb_pb is not NULL [15:06:38 CET] <JEEB> I kind of remember us being OK with initializing an iterator within a for loop? [15:07:22 CET] <JEEB> and if we needed variables within each iteration we could put that in the beginning of the for? [15:08:57 CET] <atomnuker> yes [15:09:39 CET] <JEEB> ok, I just took a look at the patch by the hlsenc guy and he weirdly stuck everything in the beginning of the function [15:11:24 CET] <atomnuker> what's with the sea of dashenc and hlsenc patches and the total lack of dashdec patches? [15:12:02 CET] <JEEB> well just writing something without an XML parser in the middle is much simpler ;) [15:12:16 CET] <JEEB> also people generally just want to push to some HTTP end point [15:12:33 CET] <JEEB> since it seems like some video platforms are starting to support DASH things through HTTP [15:12:55 CET] <atomnuker> isn't dash http only? [15:12:58 CET] <JEEB> yes [15:13:11 CET] <JEEB> also I guess no further comments on https://patchwork.ffmpeg.org/patch/11430/ ? [16:21:23 CET] <cone-641> ffmpeg 03Paul B Mahol 07master:7ea4b928a264: avfilter/af_sofalizer: fix non-power of 2 IR length filtering in time domain [17:43:03 CET] <cone-641> ffmpeg 03Michael Niedermayer 07master:4325527e1c4f: postproc/postprocess_template: Avoid using %4 for the threshold compare [17:43:04 CET] <cone-641> ffmpeg 03Michael Niedermayer 07master:c1cbeb87db4b: postproc/postprocess_template: remove FF_REG_sp from clobber list [18:47:32 CET] <philipl> what's the best tool to use if I want to look at the SPS/PPS of an h.264 stream? [18:47:39 CET] <philipl> Will ffprobe do it? [18:49:39 CET] <kierank> philipl: h264parse [18:49:41 CET] <kierank> in bmxlib [18:53:43 CET] <philipl> kierank: thanks. [18:54:08 CET] <jamrial> philipl: or use the trace_headers bitstream filter [19:01:58 CET] <cone-641> ffmpeg 03Marton Balint 07master:dfd656632e89: avformat/avio: fix avio_feof documentation [19:09:15 CET] <philipl> Thanks guys. I was trying to see the differences between two streams to see if I could append them or not. Turns out I didn't need to dig that deep. One was Main and one was High... [19:27:05 CET] <JEEB> alright, no further comments I see. [19:27:29 CET] <cone-641> ffmpeg 03Mark Thompson 07master:b97a4b658814: cbs_av1: Fix reading of overlong uvlc codes [19:30:10 CET] <cone-641> ffmpeg 03Mark Thompson 07release/4.1:b420f2356682: cbs_av1: Fix reading of overlong uvlc codes [19:39:31 CET] <cone-641> ffmpeg 03Jan Ekström 07master:a1b4f120c031: avformat/mpegts: unset DTS/PTS for subtitle PES packets if PCR not available [20:52:37 CET] <cone-641> ffmpeg 03Paul B Mahol 07master:7c201e420acc: avfilter/af_headphone: fix filtering of non-power of 2 length IRs in time domain [20:55:18 CET] <durandal_1707> atomnuker: are you any near finishing non-power of 2 FFT ? [20:57:23 CET] <atomnuker> yes, the blocker is figuring out how to modify the twiddles, is all [21:07:49 CET] <atomnuker> I need to bug bofh to tell me again like he promised [22:34:17 CET] <cone-641> ffmpeg 03Michael Niedermayer 07master:8b53d1322fef: avformat/mxfdec: Do not process zero modified_date timestamp. [00:00:00 CET] --- Sun Dec 23 2018

1 0

ffmpeg.log.20181222
by burek 23 Dec '18

23 Dec '18

[03:37:04 CET] <utack> how does one do subtraction for the select filter? [03:37:14 CET] <utack> i want frame number n minus a fixed number [03:39:06 CET] <kepstin> utack: you can use the - operator to do subtraction as you'd normally expect... [03:39:17 CET] <kepstin> utack: see also the expression reference here: https://www.ffmpeg.org/ffmpeg-utils.html#Expression-Evaluation [03:39:28 CET] <utack> okay odd, thanks [03:42:22 CET] <utack> the "\," is to escape comma? [03:44:13 CET] <kepstin> yeah, but note that you might also have to escape the \ so that your shell doesn't interpret it (or put the expression into single quotes) [03:45:25 CET] <kepstin> (a useful trick is to run `echo -vf select=yourexpression` and make sure all the \ and , are in the right spot) [03:58:27 CET] <utack> works now, thanks for the help [04:14:22 CET] <crankharder> I want to process some large 4K footage to 'thumbnail' size -- e.g. aggressively compressed to save massive amounts of space, but i can still view it to get a sense for whats in the footage -- this seems to work decently well, just wondering if there are any flags to make it faster/smaller/better. thanks! [04:14:28 CET] <crankharder> ffmpeg -i in.mp4 -c:a aac -q:a 1.0 -c:v libx264 -vf scale=320:240 out.mp4 [04:19:25 CET] <kepstin> crankharder: just general libx264 stuff - use the -preset option to make it slower/better, use the -crf option to make it smaller/worse [04:20:09 CET] <kepstin> you can't get faster and smaller and better. [04:20:24 CET] <kepstin> only pick 2 [04:22:57 CET] <kepstin> the usual recommendation for using libx264 encoder is to use the -preset option to pick the slowest preset you can stand, then use the -crf option to adjust between the size vs. quality tradeoff. [04:23:21 CET] <kepstin> (you can use -preset to speed it up, too, the default is 'medium') [04:26:03 CET] <kepstin> also, last I heard, it's better to specify a bitrate when using the builtin aac encoder (-b:a 128K or something) rather than use the -q:a option. [04:32:56 CET] <crankharder> thanks for the tips! [04:47:19 CET] <Hello71> note that because x264 is bad, you must not do it the other way around [04:48:42 CET] <kepstin> well, it's not that x264 is bad, it's probably one of the best encoders around for any codec tbh. [04:49:09 CET] <kepstin> it's just that the -crf option isn't truely a quality setting, it's rather a bitrate vs quality tradeoff setting that depends on a bunch of other settings [04:49:21 CET] <kepstin> and changing preset changes a bunch of other settings [04:49:54 CET] <kepstin> (crf was originally part of how 2-pass bitrate encoding worked, and you were expected to use (mostly) the same settings for both passes) [07:26:13 CET] <cheapie> So I figured out how to get ffmpeg to produce files that Panasonic DVD players with "DivX" (not DIVX) support will play :) [07:26:17 CET] <cheapie> "ffmpeg -i /path/to/source/file -c:v libxvid -c:a mp2 /path/to/destination/file.avi" seems to do the trick. [10:48:34 CET] <pagios> hi all, ffmpeg -i rtmp://localhost/src/$name -o rtmp://localhost/src/stream2 <--- can i use this to convert stream1 to stream2? [10:50:40 CET] <pagios> hello? [10:51:33 CET] <pink_mist> be patient [10:52:22 CET] <pink_mist> there might be someone who knows the answer to your question and is willing to give you an answer, but you'll need to wait for said person to show up [10:52:27 CET] <pink_mist> I've got no clue [12:28:18 CET] <ariyasu> pagios try [12:28:37 CET] <ariyasu> ffmpeg -i rtmp://localhost/src/$name -c copy rtmp://localhost/src/stream2 [12:28:50 CET] <ariyasu> ffmpeg doesn't use -o to specify an output [12:29:23 CET] <pagios> is ffmpeg better than libav and gstreamer? [13:23:02 CET] <alone-x> hello any one can use qsv with ffmpeg under windows? [14:30:16 CET] <Bear10> which methods should i be looking at to grab a packet and get a frame to generate a preview image? [19:12:35 CET] <Alina-malina> I want to append a certain sound.mp3 to an existing video i mean basically i have a backgroundmusic.mp3 which is 00:10:00 duration and i need that on 07:15 to add a bass.mp3 file, but ti not lose the backgroundmusic.mp3, how to achieve this with ffmpeg? [19:12:55 CET] <Alina-malina> and its a file.mp4 file that both tracks should be applied to [19:14:38 CET] <ariyasu> you can concat them with ffmpeg [19:14:50 CET] <ariyasu> or copy /b works with most audio files [19:14:57 CET] <ariyasu> then use ffmpeg to mux them [19:15:12 CET] <ariyasu> https://trac.ffmpeg.org/wiki/Concatenate [19:15:43 CET] <ariyasu> or "copy /b audio1.mp3 + audio2.mp3 output.mp3" [19:16:04 CET] <ariyasu> ffmpeg -i file.mp4 -i output.mp3 -c copy output.mp4 [19:16:10 CET] <ariyasu> somthing like that should work fine too [19:17:10 CET] <Alina-malina> no not concat [19:17:41 CET] <Alina-malina> i need to apply 2 tracks at the same time [19:17:46 CET] <Alina-malina> not concatition [19:17:51 CET] <Alina-malina> concatination [19:18:03 CET] <furq> Alina-malina: -i bgm.mp3 -i sound.mp3 -lavfi "[0:v]atrim=end=7:15[tmp];[tmp][1:v]concat" [19:18:14 CET] <furq> s/v/a/ [19:18:30 CET] <Alina-malina> hmmm let me try this thanks [19:19:00 CET] <furq> that will just cut bgm.mp3 at 7:15 and play sound.mp3 [19:19:03 CET] <furq> i'm not sure if that's what you awnt [19:19:13 CET] <Alina-malina> no not cut [19:19:15 CET] <Alina-malina> append there [19:19:25 CET] <Alina-malina> no need to interupt bgm.mp3 [19:19:35 CET] <Alina-malina> though its interesting effect hehe [19:20:14 CET] <pink_mist> Alina-malina: append? but that's exactly what concatenation is [19:20:24 CET] <pink_mist> and you didn't want concatenation [19:20:26 CET] <c_14> sounds like you want adelay and amix [19:20:28 CET] <pink_mist> you said so [19:20:34 CET] <Alina-malina> c_14, yes [19:20:47 CET] <Alina-malina> pink_mist, i need both files tp play at the same time pink_mist [19:20:51 CET] <Alina-malina> to* [19:21:11 CET] <furq> -lavfi "[1:a]adelay=435000|435000[tmp];[0:a][1:a]amix" [19:21:21 CET] <Alina-malina> great great thanks [19:21:28 CET] <furq> -lavfi "[1:a]adelay=435000|435000[tmp];[0:a][tmp]amix" [20:28:31 CET] <Alina-malina> furq, Filter amix has an unconnected output :-/ [20:32:28 CET] <durandal_1707> Alina-malina: you need to first learn basics [20:42:03 CET] <Alina-malina> ok it worked [20:42:05 CET] <Alina-malina> i figured it out [20:42:55 CET] <Alina-malina> durandal_1707, no i think i need a good movie to watch now and relax, too much hard work already done today, enough working [21:58:11 CET] <Alina-malina> furq, i am not sure, but what if i want to do the same, but more then 1 time? like 2 times or more? i tried to find this in documents, but it doesnt say anything about multiple amix and adelays in 1 command? [00:00:00 CET] --- Sun Dec 23 2018

1 0

ffmpeg-devel.log.20181221
by burek 22 Dec '18

22 Dec '18

[11:33:02 CET] <thardin> youtube has started delivering av1 [11:33:33 CET] <durandal_1707> old news [11:56:10 CET] <durandal_1707> atomnuker: why you do not complete new channel layout API? [12:08:03 CET] <cone-532> ffmpeg 03Paul B Mahol 07master:c49307e7847f: avfilter/af_sofalizer: use fabsf() instead of fabs() [12:08:04 CET] <cone-532> ffmpeg 03Paul B Mahol 07master:47ba0854720e: avfilter/af_headphone: use fabsf() instead of fabs() [13:00:42 CET] <JEEB> wat [13:02:51 CET] <JEEB> I have a HLS key ending with .key [13:02:59 CET] <JEEB> and the HLS demuxer says nope [13:04:03 CET] <JEEB> http://up-cat.net/p/514b4f2b [13:05:38 CET] <JEEB> or is that just really badly mentioning to me that the file protocol is less open than http? [14:17:01 CET] <Compn> JEEB : encrypted [14:17:25 CET] <Compn> ? [14:17:48 CET] <Compn> i guess demuxer is saying it does not want to open some non hls (m3u8) file [14:19:47 CET] <Compn> oof that vividas demuxer [14:20:16 CET] <JEEB> Compn: yes, standard AES encrypted HLS stream that I was generating :P [14:20:23 CET] <JEEB> and reading locally [14:20:44 CET] <JEEB> there's seemingly an allowed_extensions AVOption that includes the key file [14:21:00 CET] <JEEB> could also be limited to the file protocol, but the message received says nothing about that [14:21:27 CET] <JEEB> > `-allowed_extensions <string> .D....... List of file extensions that hls is allowed to access (default "3gp,aac,avi,flac,mkv,m3u8,m4a,m4s,m4v,mpg,mov,mp2,mp3,mp4,mpeg,mpegts,ogg,ogv,oga,ts,vob,wav")` [14:25:13 CET] <Compn> http://helixproducts.real.com/hmdp/documentation/helixserver/1512_LR/html/C… [14:25:21 CET] <JEEB> I know how it works [14:25:30 CET] <Compn> Key files use the playlist file name and end with the file extension .key. [14:25:31 CET] <Compn> bah [14:25:35 CET] <Compn> then add .key :P [14:25:52 CET] <JEEB> the HLS specification AFAIK doesn't limit the file names for keys [14:25:57 CET] <JEEB> let me check the RFC [14:26:10 CET] <Compn> i was linking to helix specs for source :P [14:26:18 CET] <Compn> I know you know how it works [14:26:22 CET] <Compn> hehe [14:26:42 CET] <JEEB> https://tools.ietf.org/html/rfc8216#section-4.3.2.4 [14:27:14 CET] <JEEB> URI is just "The value is a quoted-string containing a URI that specifies how to obtain the key. This attribute is REQUIRED unless the METHOD is NONE." [14:27:45 CET] <JEEB> this is regarding the key files, but it doesn't limit the key files either https://tools.ietf.org/html/rfc8216#section-5 [14:28:01 CET] <JEEB> but yea, I have a hunch that these limits are only hit because I use file:// [14:28:05 CET] <JEEB> although I can't be 100% sure :P [14:28:19 CET] <JEEB> as in, I'm locally testing a stream I'm generating on that box with ffprobe [14:29:11 CET] <Compn> Playlists that [14:29:11 CET] <Compn> violate these rules are invalid; clients MUST fail to parse them. [14:29:48 CET] <Compn> it lists m3u and m3u8 .... [14:29:49 CET] <Compn> File extension(s): .m3u8, .m3u (see Section 4) [14:29:55 CET] <JEEB> that's for the playlists, yes [14:30:01 CET] <Compn> oh this is a segment then ? [14:30:02 CET] <JEEB> they don't limit the extensions for media [14:30:05 CET] <JEEB> or keys [14:30:15 CET] <Compn> okey [14:30:22 CET] <JEEB> #EXT-X-KEY:METHOD=AES-128,URI="das_key.key" [14:30:22 CET] Action: Compn scurries off [14:30:26 CET] <JEEB> :P [14:30:50 CET] <JEEB> otherwise it would have failed at loading the playlist or media [14:31:02 CET] <JEEB> both of which were OK, as you can see by the default white list :P [15:17:44 CET] <JEEB> ok so it is file:// specific http://ffmpeg.org/pipermail/ffmpeg-devel/2017-May/211796.html [16:52:42 CET] <durandal_1707> can I get primaries and transfer characteristic from colorspace transformation from YCC to Y'U'V' to RGB ? [17:02:48 CET] <funman> michaelni: i'm trying to make 420->422 faster by memcpy()ing the Y plane: https://pastebin.com/9teGMmAp [17:03:04 CET] <funman> any suggestion besides the obvious need to check that pixel format matches? [17:03:24 CET] <funman> https://github.com/kierank/ffmpeg-obe/commit/5a354020872bec61bea6be7604ec88… this is an old issue [17:03:46 CET] <kierank> durandal_1707: ^ [17:03:50 CET] <kierank> you are pixel format expert [17:04:00 CET] Action: durandal_1707 *hides* [17:04:26 CET] <funman> shame that swscale doesnt use bufferrefs, would've been just moving pointer rather than memcpy [17:04:52 CET] <funman> actually i wonder if a UV420 -> UV422 would make sense [17:08:21 CET] <durandal_1707> funman: please rewrite swscale [17:09:42 CET] <JEEB> wouldn't you just do something like zimg for most stuff :P [17:10:06 CET] <JEEB> durandal_1707: also not sure you can get those two separate things from just two images [17:10:26 CET] <funman> sure [17:10:54 CET] <funman> also can't pass NULL pointer or 0 stride for Y, check_image_pointers doesn't like it [17:11:02 CET] <durandal_1707> JEEB: this is about PhotoCD colors that predates all things in zimg [17:11:14 CET] <JEEB> durandal_1707: those were two separate comments [17:11:22 CET] <JEEB> first regarding "rewriting swscale" [17:11:29 CET] <JEEB> second regarding the photocd thing [17:11:56 CET] <JEEB> I think I might have some old old photocd CD somewhere [17:12:06 CET] <JEEB> although I mostly remember them already containing JPEGs [18:20:43 CET] <cone-779> ffmpeg 03Mohammad Izadi 07master:2744d6b364c0: Add HDR dynamic metadata struct (for SMPTE 2094-40) to libavutil [00:00:00 CET] --- Sat Dec 22 2018

1 0

ffmpeg.log.20181221
by burek 22 Dec '18

22 Dec '18

[00:04:59 CET] <Keshl> Just a little FYI: After setting VFR up for my weirdo specific use-case, I have /zero issues/ with seeking. I suspect it's because I used faststart so the entire MOV atom is available right away, and thus nothing ends up in "odd" spots. Regardless, everything's totally fine. The only "weird" bit is I can't seek to the middle of a long-duration frame, but I can jump to it's beginning or end easily enough, and I generally don't need the [00:05:00 CET] <Keshl> audio from those portions. [00:07:59 CET] <Keshl> Also, using mpdecimate, I have no dropped frames where I don't expect them or heavily desire them. Everything just worked out. o_o [00:08:31 CET] Action: Keshl hugs all the developers. -É-. [00:49:59 CET] <ntd> can ffmpeg be convinced to pull, say, a sep audio stream suing anything other than sep -i? [00:53:43 CET] <pink_mist> -i is how you tell it to pull things [00:53:57 CET] <pink_mist> your question really doesn't make much sense to me [01:45:17 CET] <fengshaun> is it necessary to run frames through filter if I only want to re-encode? [01:45:35 CET] <fengshaun> the transcoding example uses filters and I'm slowly trying to get rid of it [01:49:57 CET] <fengshaun> if I remove filter graph from the transcoding example, I get "corrupted size vs prev_size" [01:53:23 CET] <fengshaun> nevermind, it seems the segfault was coming from elsewhere [05:53:13 CET] <fengshaun> I'm trying to transcode a file with ffmpeg api, so far I have completed decoding, but when encoding, I'm getting the following error from libx264: "Input picture width (1280) is greater than stride (0)" [05:53:58 CET] <fengshaun> I've tried setting AVFrame.linesize[0 to 2] manually which results in segfault, and using avpicture_fill results in same segfault. [05:54:03 CET] <fengshaun> not sure what to do at this point [05:54:09 CET] <fengshaun> and I can't find any info on the web [05:54:21 CET] <fengshaun> using h264 decoder and encoder [05:54:26 CET] <fengshaun> there are no filters [05:54:44 CET] <fengshaun> it seems adding a filter in between works fine like transcoding example [05:54:53 CET] <fengshaun> but I'm trying to skip filtering [06:06:57 CET] <fengshaun> av_image_fill_linesizes(frame->linesize, pixel_format, frame_width) also results in return value of 0 and a segfault at send_frame [06:29:44 CET] <fengshaun> nvm again, looks like I was mis-using send/receive api [06:29:53 CET] <fengshaun> calling them repeatedly until EAGAIN [06:30:00 CET] <fengshaun> which messed up frame data [06:31:29 CET] <fengshaun> looks like it's working!! [13:26:04 CET] <th3_v0ice> How can I force the sending of PAT (Program Association Table) in MPEG-TS streams? [13:26:18 CET] <JEEB> pretty sure it's done by default, since it's a required table [13:26:27 CET] <JEEB> you would have to specifically disable sending PAT [13:26:40 CET] <JEEB> how often PAT is sent is a separate discussion, for that you'd have to look into the mpeg-ts muxer [13:55:52 CET] <th3_v0ice> JEEB, Well the control software is reporting that its not sent fast enough (>500ms). Will look into the MPEG-TS muxer. Thanks. [14:17:43 CET] <th3_v0ice> It seems that these options are configurable with input parameters. -pcr_period -pat_period. [17:25:24 CET] <tombb> hi all, can ffmpeg create mxf-atom files? (for avid environment) [17:26:17 CET] <JEEB> MXF, yes [17:26:31 CET] <JEEB> now what "MXF-Atom" means I have no idea [17:26:54 CET] <tombb> there's mxf op1a, which includes several tracks, video+audio [17:27:14 CET] <tombb> if u split it to different files, one per stream, they're atoms (as far as I understand) [17:29:01 CET] <tombb> if you're familiar with bmxtranswrap.exe, that what it does, split op1a to atoms.. I'm not really sure how that works and whats the difference between an mxf with a single stream to mxf-atom [17:31:15 CET] <^Neo> I'm finding some odd behavior on 4.0, I'm generating an HLS stream from a hardware device, using -use_wallclock_as_timestamps because I don't trust the hardware. I have '+append_list' in the hls flags. If I probe the resulting m3u8 and print the PTS's, and restart the HLS instance I see the PTS's jump by ~24 hours... [19:26:43 CET] <funyun> what's the best way to run ffmpeg in windows in the most linuxy way possible? [19:28:14 CET] <Rick627> just like linux [19:28:18 CET] <Rick627> in the command line [19:29:42 CET] <funyun> Rick627: okay thanks [19:30:26 CET] <^Neo> when I run '-fflags +genpts' do I need to specify a '-r' argument with a framerate? [00:00:00 CET] --- Sat Dec 22 2018

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