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October 2016

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ffmpeg-devel.log.20161020
by burek 21 Oct '16

21 Oct '16

[01:15:25 CEST] <Matador> Compn : yes I am using months ago ffmpeg for QSV [01:22:40 CEST] <Matador> just wish it was sorted so could use an updated ffmpeg w/ qsv [08:21:31 CEST] <beauty> why av_frame_get_channels(frame) return 0, but frame->channels is 2? [09:13:20 CEST] <cone-146> ffmpeg 03Muhammad Faiz 07master:56689a46b304: avfilter/showcqt: add cscheme option [09:13:20 CEST] <cone-146> ffmpeg 03Muhammad Faiz 07master:258c49d61451: avfilter/firequalizer: use zero phase kernel [09:56:59 CEST] <cone-146> ffmpeg 03Carl Eugen Hoyos 07master:c0e2846dcdc1: lavc/sheervideo: Increase av_get_codec_tag_string() input buffer size. [18:09:28 CEST] <cone-507> ffmpeg 03Steven Liu 07master:5702416c57af: avformat/flvenc: do not attempt to write duration and filesize when not seekable [18:11:39 CEST] <atana> michaelni: I installed ffmpeg from github repo but I don't see ffplay executable file [18:11:57 CEST] <atana> after make install [18:12:07 CEST] <michaelni> atana, you need libsdl [18:12:35 CEST] <atana> for testing the filter I need ffplay right? [18:13:09 CEST] <michaelni> you dont "need", its one way to test filters [18:14:28 CEST] <atana> what's the other way to test using command line. For ffplay do I need to install libsdl? [18:15:29 CEST] <michaelni> you could use ffmpeg [18:15:37 CEST] <michaelni> ffplay needs sdl yes [18:16:19 CEST] <atana> ffmpeg someaudio.mp3 myfilter=filterargs is this the format? [19:02:34 CEST] <Chloe> note that it needs sdl2 [19:06:41 CEST] <RiCON> nevcairiel: can you confirm dxva2-copy is bugged/broken since http://git.videolan.org/?p=ffmpeg.git;a=commitdiff;h=9b462a0b9 ? [19:24:09 CEST] <CFS-MP3> michaelni can you comment on this? http://ffmpeg.org/pipermail/ffmpeg-devel/2016-October/201358.html [19:30:18 CEST] <nevcairiel> RiCON: it probably needs more bracers if you compile both dxva2 and d3d11va [19:30:55 CEST] <nevcairiel> let me push a fix [19:37:33 CEST] <nevcairiel> RiCON: hopefully that solves it [19:37:34 CEST] <cone-507> ffmpeg 03Hendrik Leppkes 07master:8bd38ec5bde1: dxva2: fix surface selection when compiled with both d3d11va and dxva2 [19:46:08 CEST] <michaelni> atana, ffmpeg -i someaudio.mp3 -af ... outfile.wav [19:55:54 CEST] <cone-507> ffmpeg 03Andreas Cadhalpun 07master:ee5f0f1d355f: rsd: limit number of channels [20:00:42 CEST] <RiCON> nevcairiel: fixed, thanks [20:09:45 CEST] <Chloe> Any of the OPs use matrix.org? [20:11:22 CEST] <BtbN> Nah, I still use classic phone lines to log out. [20:13:08 CEST] <llogan> kind of looks like Slack [20:13:44 CEST] <BtbN> Which looks suspiciously like IRC. [20:15:02 CEST] <Chloe> Yeah, it's pretty much a federated slack. There's a nice matrix<->(freenode)irc bridge, but it requires setup from a channel OP afaik [20:16:02 CEST] <llogan> Most of my clients request that I use Slack. None have wanted to use traditional IRC. [20:17:49 CEST] <Chloe> err, by setup, I just mean it needs to be enabled, so matrix users can join. [20:23:50 CEST] <CFS-MP3> Thanks for taking a look at the SCTE-35 stuff, michaelni ... now, what's the next step to actually have this patchset merged? Who actually does the commit? [20:27:00 CEST] <BtbN> So it's also an over-complicated IRC client [20:27:35 CEST] <Chloe[m]> yes :) Looks like there's no setup needed though [20:28:02 CEST] <Chloe[m]> it's only if you want to link it to a channel on the matrix network rather than using the normal bridge alias [21:50:03 CEST] <CFS-MP3> michaelni - Can you merge this? https://github.com/FFmpeg/FFmpeg/pull/238 (or point me in the right direction if you're not the right person to ask) [21:54:40 CEST] <BtbN> we don't do github PRs. [21:54:43 CEST] <Chloe[m]> CFS-MP3: ffmpeg doesn't use github. [21:55:28 CEST] <Chloe[m]> CFS-MP3: it will get merged by a maintainer, assuming there is no longer anyone who still has an issue with it [22:08:11 CEST] <CFS-MP3> BtbN what's the process then? [22:08:50 CEST] <Chloe> CFS-MP3: I just said [22:09:40 CEST] <CFS-MP3> Ah sorry, missed that Chloe [22:09:45 CEST] <CFS-MP3> OK, so I'll just wait then [22:33:43 CEST] <atomnuker> michaelni: what do the scale and bits arguments do in ff_mdct_init()? [22:38:12 CEST] <atomnuker> (putting the scale to 1.0 should (?) make the mdct output normalized values, but what does the bits do, define operation precision?) [22:45:43 CEST] <peloverde> nbits is log2(number of points) [22:46:31 CEST] <atomnuker> duh [22:49:18 CEST] <atomnuker> can't use it then, my input's infuriatingly not a power of two [22:50:28 CEST] <nevcairiel> we also have the opus 960 mdct if that helps [22:52:59 CEST] <atomnuker> they're only inverse [22:53:38 CEST] <nevcairiel> hm i guess so [00:00:00 CEST] --- Fri Oct 21 2016

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ffmpeg.log.20161020
by burek 21 Oct '16

21 Oct '16

[03:02:36 CEST] <bencc> I'm doing screen capture with: [03:02:37 CEST] <bencc> -vcodec libx264 -pix_fmt yuv420p -preset:v ultrafast -crf 0 [03:02:42 CEST] <bencc> and saving to mkv [03:03:08 CEST] <bencc> the mkv file size increases monotonically even when there is no change on the screen for hours. [03:03:42 CEST] <bencc> is it possible to change something so the file size won't increase without affecting the cpu load too much? [03:06:48 CEST] <furq> using -preset superfast will make a big difference [03:07:13 CEST] <furq> there's no way to stop the filesize from increasing at all unless you want to patch ffmpeg to support vfr output [03:07:34 CEST] <bencc> furq: it's ok if the file size increases [03:07:44 CEST] <bencc> but 1GB per hour with no change seems to much [03:07:50 CEST] <bencc> I'll try -preset superfast [03:08:15 CEST] <furq> i just tested with a static 1080p image and superfast takes about 10% of the space [03:08:42 CEST] <bencc> furq: that's huge difference. what about cpu? [03:08:58 CEST] <furq> 300ish fps for ultrafast, 200 for superfast [03:09:33 CEST] <bencc> nice [03:10:08 CEST] <bencc> In my case I only need 30fps with: -framerate 30 [03:15:33 CEST] <furq> actually it looks like vfr output is supported [03:15:42 CEST] <furq> you can try -vf mpdecimate to drop duplicate frames [03:16:01 CEST] <furq> the defaults will drop some non-duplicate frames though so you'll need to tune it [03:17:49 CEST] <bencc> furq: I'll read about "-vf mpdecimate" and try it. thanks [03:20:19 CEST] <dougquaid> I'm not sure if this is the right place to ask, but is there a video codec that supports GPS metadata? For example, if I have a dashcam with GPS, can I save the route I drove in the video's metadata? [03:21:51 CEST] <furq> mp4 supports location metadata [03:21:57 CEST] <furq> i don't know if you'd be able to store an entire route in it though [03:22:23 CEST] <k_sze[work]> aren't there container formats that support arbitrary data streams? [03:22:29 CEST] <furq> yeah obviously you could do that [03:22:34 CEST] <furq> it won't be readable by any tools though [03:23:59 CEST] <k_sze[work]> What you can do is cheat: encode the GPS metadata as subtitles. [03:24:04 CEST] <k_sze[work]> :P [03:24:12 CEST] <dougquaid> That's actually not a bad idea [03:25:34 CEST] <furq> just make sure cohaagen doesn't get hold of it [03:27:24 CEST] <k_sze[work]> Vilos Cohaagen? [05:50:20 CEST] <beauty> hello [05:50:39 CEST] <beauty> why av_frame_get_channels(frame) return 0, but frame->channels is 2? [05:55:28 CEST] <k_sze[work]> Are there some likely causes for avformat_free_context to result in segfault? [06:41:25 CEST] <k_sze[work]> Hmm, ok, why does avcodec_free_context and then avformat_free_context cause segfault? [08:14:21 CEST] <ritsuka> k_sze[work]: apple uses time metadata in mov for gps route, not sure if ffmpeg supports it [08:15:24 CEST] <k_sze[work]> ritsuka: ffmpeg is probably agnostic to it. [08:15:46 CEST] <ritsuka> right, probably you can get the raw samples from the track [08:16:53 CEST] <k_sze[work]> One would still write their own formatter and parser. I don't think there is any magic, unless you use some Apple API, not ffmpeg. [08:17:35 CEST] <beauty> why av_frame_get_channels(frame) return 0, but frame->channels is 2? [09:46:33 CEST] <sasha047> äîáðûé äåíü [09:50:36 CEST] <ootje> trying again today: what is the -filter_complex equivalent of -loop 1 -i image.png -t 2 ? tried -filter_complex "movie=image.png:loop=0, setpts=PTS-STARTPTS" -t 2 but it just renders on and on and on without stopping. tried adding ", trim=0:2" also but nope.. any takers? [09:51:15 CEST] <ootje> (even if i do :loop=100 it only seems to render one frame, am i doing something wrong with the setpts?) [10:45:35 CEST] <A124> Hello, can someone please give me an example how to take audio from an a/v file, encode that to aac and put it in mp4 container? [10:45:56 CEST] <A124> I am getting Output file #0 does not contain any stream [10:46:07 CEST] <A124> Maybe nevermind... the source might not have audio. [14:09:31 CEST] <termos> I'm having the same issues as described here: https://trac.ffmpeg.org/ticket/5319 does anyone know of some secret progress on this? [14:10:07 CEST] <JEEB> I would bet on "nothing after that change 6 months ago" [14:14:18 CEST] <wildefyr> why can't ffmpeg cut video accurately to the frame? [14:14:46 CEST] <wildefyr> usually when I cut video the audio is cut precisely but the video is not leaving a freeze of 'about' a second at the beginning of the video [14:15:29 CEST] <wildefyr> or the player interprets this as no video data and skips over the audio for that second, retaining the sync, but it's no good when I import the file into the video editor, or upload to youtube [14:32:17 CEST] <c_14> Are you copying the video? [14:33:00 CEST] <wildefyr> yeh, I'd prefer not to do re-encoding but it's not a requirement [14:33:18 CEST] <c_14> If you don't reencode the video you'll always cut at the nearest keyframe [14:33:49 CEST] <wildefyr> interesting [14:34:53 CEST] <c_14> It shouldn't cause the issues you're describing, but it might. [14:35:15 CEST] <nonex86> if you need frame accuracy usually you need to reencode first and usually last gop [14:35:21 CEST] <c_14> What you can do is either reencode the video, or write a script that uses ffprobe to get the timestamp of the closest keyframe to where you want to cut and use that as the cutting point [14:36:10 CEST] <c_14> Or what nonex86 mentioned, though that'll require cutting out the "middle portion" and then cutting out the other portions reencoding and concatenating [14:36:10 CEST] <wildefyr> the latter seems more applicable [14:36:51 CEST] <wildefyr> how would I get the keyframe with ffprobe? [14:36:57 CEST] <wildefyr> closest* [14:37:06 CEST] <nonex86> frame type=I [14:37:14 CEST] <nonex86> dump all frames [14:37:20 CEST] <nonex86> grep the certain type [14:38:05 CEST] <nonex86> what kind of stream we are speaking? h264? [14:38:44 CEST] <wildefyr> yes [14:38:52 CEST] <nonex86> not sure about one thing [14:38:59 CEST] <nonex86> when ffprobe report i frame [14:39:01 CEST] <nonex86> does it mean [14:39:03 CEST] <nonex86> idr [14:39:11 CEST] <wildefyr> well what you've said is confusing [14:39:15 CEST] <nonex86> and i frame with SEI recovery point [14:39:33 CEST] <wildefyr> I want to enter a timestamp and get the closest keyframe to the entered timestamp [14:39:40 CEST] <wildefyr> then cut from there [14:40:26 CEST] <nonex86> isnt ffmpeg muxer concat work this way? [14:40:51 CEST] <nonex86> it takes closest to keyframe pts and cut fragment? [14:40:57 CEST] <wildefyr> no [14:41:01 CEST] <wildefyr> well [14:41:05 CEST] <wildefyr> yes for video streams [14:41:10 CEST] <wildefyr> but I have audio as well [14:41:18 CEST] <wildefyr> that it cuts from the entered time and NOT the keyframe [14:41:30 CEST] <wildefyr> so I have audio that is out of sync from video [14:41:53 CEST] <c_14> ffprobe -skip_frame nokey -select_streams v -show_entries frame=pkt_pts_time -of default=nokey=1:nw=1 [14:41:54 CEST] <wildefyr> or frozen video frame until it syncs with audio [14:42:03 CEST] <c_14> Go through the list output from that command and pick the timestamp closest to what you want [14:43:50 CEST] <nonex86> few days ago i cut using ffmpeg from movie (h264 + aac) without any audio issue :/ [14:44:01 CEST] <nonex86> i didnt supply any special parameters [14:45:13 CEST] <nonex86> result fragment had video and audio in sync [14:45:17 CEST] <nonex86> at least it looks like [14:45:44 CEST] <wildefyr> well it depends on the player I think [14:46:08 CEST] <wildefyr> ie importing the cut file into a video editor makes the audio and video become out of sync [14:47:18 CEST] <nonex86> the players cant do anything, if the pts of the streams will be out of sync [14:47:33 CEST] <nonex86> dont think it was a player magic :/ [14:49:45 CEST] <Zucca> Hi. Does ffmpeg have a video filter that could copy parts that have no significant changes from previous frame to next one? I would use this to eliminate the wast of bits to render noice in the background. ImageMagick has the -fuzz switch for creating animations that acts close to this at least. So I could "pipe" the whole video trough IM, but that seems dirty... [14:52:31 CEST] <wildefyr> ah! [14:52:43 CEST] <wildefyr> -ss seeks if it's placed before -i [14:53:17 CEST] <wildefyr> if after it cuts there from the audio, and then the following keyframe for video [14:55:48 CEST] <DHE> Zucca: there's a "decimate" filter that drops frames when they are close enough to the previous frame. technically it produces a variable FPS output which you would then need to compensate for if you can't take that. [15:01:22 CEST] <Zucca> DHE: Ok. I'll see how that works. Thanks. :) [15:46:47 CEST] <Kadigan> VFR is evil. [15:47:12 CEST] <nonex86> variable frame rate? [15:47:22 CEST] <Kadigan> Yes. [15:47:52 CEST] <nonex86> well, if you need to merge several compatible streams [15:48:01 CEST] <nonex86> but with different frame rate [15:48:20 CEST] <nonex86> its the only solution - proper container with vfr support :) [15:48:30 CEST] <Kadigan> I should clarify: VFR is evil when you're trying to use some form of predictive frame stuffing/tweening. [15:48:56 CEST] <Kadigan> Unfortunately, this happens a lot in anime, esp. in the action genre. [15:48:58 CEST] <nonex86> ah :) [15:49:58 CEST] <Kadigan> There's this thing called SVP (Smooth Video Project) for Windows which does that -- it tweens until the frame rate matches the display refresh rate. Each time FPS changes, the buffer needs to be rebuilt, which results in a noticeable jerk. [15:50:03 CEST] <Kadigan> Meh. [15:50:34 CEST] <nonex86> frame tweening sounds like a magic [15:50:47 CEST] <nonex86> i did not say the simple blend from one frame to another [15:50:53 CEST] <nonex86> *did not mean [15:51:11 CEST] <Kadigan> Yeah, they do proper motion vector detection and compensation, along with artifact masking and such. [15:52:03 CEST] <Kadigan> The result, when it works well, is smooth but sharp animation. [15:52:17 CEST] <nonex86> its better than nothing i guess [15:52:29 CEST] <Kadigan> And yes, it works best with high-quality content. [15:52:55 CEST] <Kadigan> Well, I suppose it's the PC's counterpart to what TVs do... Used to hate that, I did. [15:53:05 CEST] <Kadigan> (makes film material look TV-like) [15:53:15 CEST] <nonex86> too much blur? :) [15:53:31 CEST] <Kadigan> No, the main difference is in the time resolution/domain. [15:53:41 CEST] <Kadigan> Film = 24fps, something, [15:53:48 CEST] <Kadigan> while TV is either 50i or 60i [15:54:13 CEST] <Kadigan> Less spatial resolution, more temporal resolution. Film feels a bit unreal, while TV material typically looks like a reality show, [15:54:16 CEST] <Kadigan> or something... [15:54:32 CEST] <nonex86> isnt interlace here is another pain in the @ss? :) [15:55:01 CEST] <Kadigan> You know the difference when you go filming with your cheap-ass consumer camera, and it looks... well, like if YOU filmed it, not a pro film crew? :D [15:55:05 CEST] <Kadigan> That's the feel I mean :D [15:55:51 CEST] <Kadigan> Try filming it in 24p, or 24/1.001 -- it'll feel very different. [15:56:40 CEST] <Kadigan> As for interlacing, I believe YADIF is sufficiently advanced to basically IGNORE interlacing at this point. ffmpeg will crunch it down just as well. [15:57:16 CEST] <Kadigan> (actually, if you ask YADIF to rebuild frames from fields, 1:1 fi:fr, and set PTS accordingly, you'll get slow motion with that "filmic" look!) [15:59:42 CEST] <nonex86> that SVP thing, is it commercial or opensource? [16:02:17 CEST] <nonex86> nevermind, found it :) [16:07:33 CEST] <Kadigan> You want madVR and MPC-HC with that, I suppose ;) [16:42:04 CEST] <termos> My encoded h264 and aac works by itself but when I mux it together in an FLV container everything is messed up. No audio and lagging picture. If I mux into mpegts everything is fine. Any ideas? [16:52:10 CEST] <nonex86> termos: what about your pts? [16:53:42 CEST] <nonex86> termos: how do you provide them? check the pts in both flv and ts files [16:53:53 CEST] <nonex86> termos: be sure they are correct in both case [16:54:18 CEST] <nonex86> termos: enable ffmpeg debug, check for warning/errors when muxing to flv [17:06:51 CEST] <termos> I could double-check my pts, but it's strange how it's working for mpegts isn't it? [19:29:15 CEST] <help> hi [19:29:43 CEST] <Guest30393> is there anyone here currently? [19:35:04 CEST] <maximq> Hi guys. I've been experimenting a bit with the Tesseract based OCR filter, which takes bitmaps/dvb subs input and outputs the text format to the frame metadata property 'lavfi.ocr.text'. What I want is to get this text into WebVTT and segmented into a mpegts. Would I need to write an intermediate filter to extract the frame metadata and use the output to the WebVTT encoder? Or can this be done more dynamically? [19:54:13 CEST] <c_14> maximq: probably [20:09:44 CEST] <DHE> can I specify the number of threads used by the filter_complex pipeline on the commandline like I can for codecs? [20:17:03 CEST] <DHE> I can do it in libav, but I can't find any way to do it in ffmpeg itself. [20:19:17 CEST] <BtbN> Doesn't the global threads option also count for those? [20:20:28 CEST] <DHE> I thought -threads was input/output sensitive [20:23:23 CEST] <DHE> whereas filter_complex is inherently a global thing, leaving the question of where does that work? [20:41:09 CEST] <BtbN> DHE, allocating the filter graph should do some option parsing, which then ends up setting nb_threads on the FilterGraph/AVFilterContext from the threads option [20:42:32 CEST] <BtbN> ffmpeg.c is probably also involved in that [20:42:35 CEST] <DHE> BtbN: that's not what I'm experiencing... [20:42:46 CEST] <DHE> actually, let me try one thing... [20:44:02 CEST] <BtbN> https://github.com/FFmpeg/FFmpeg/blob/master/ffmpeg_filter.c#L1022 [20:44:18 CEST] <DHE> BtbN: wrapped in if(simple) which I assume means filter_complex doesn't apply [20:44:41 CEST] <BtbN> that's indeed odd. Seems like a bug [20:45:23 CEST] <DHE> yeah debugger confirms it's not being used. (test run with 1 input, 3 outputs, 4 instances of "-threads 1" on the commandline) [20:45:47 CEST] <BtbN> Just moving it out of the if(simple) should do the trick? [20:45:59 CEST] <BtbN> At least I'm not aware of a way to set filter graph opts via the commandline [20:47:33 CEST] <DHE> BtbN: https://paste.fedoraproject.org/456337/98918814/ here's the backtrace of a filter_complex being loaded [20:50:09 CEST] <DHE> nb_threads=0 and I don't see any place to set a number of threads [20:50:28 CEST] <BtbN> it should just carry over the encoder threads count imo [20:51:02 CEST] <BtbN> But I wouldn't be surprised if the filter_compley string has some way to pass opts to the filter graph itself [20:51:11 CEST] <DHE> okay that backtrace isn't applicable. that's used only temporarily... [20:51:18 CEST] <DHE> it should, but if there's multiple encoders which one does it use? [20:51:41 CEST] <BtbN> There should be one filter graph per encoder? [20:52:25 CEST] <BtbN> OutputStream *ost = fg->outputs[0]->ost; [20:52:29 CEST] <BtbN> So if there's more: The first one. [20:57:31 CEST] <BtbN> DHE, doesn't settings the threads option on the individual filter work? [21:00:34 CEST] <Parker__> Hey guys! I'm back! [21:01:13 CEST] <Parker__> Question: I have 1000 .ppm images that I want to convert into a movie and all of them have a white background [21:01:48 CEST] <Parker__> is there any tricks I could do to save on filesize yet maintain a lossless representation of the important pixels? [21:02:10 CEST] <klaxa> https://trac.ffmpeg.org/wiki/Create%20a%20video%20slideshow%20from%20images [21:02:19 CEST] <klaxa> if you want lossless compression use a lossless codec [21:02:37 CEST] <Parker__> I'm using x264 [21:02:50 CEST] <Parker__> but for a 1 minute movie, the filesize is about 150 mb... [21:03:08 CEST] <Parker__> -c:v libx264 -preset veryslow -qp 0 [21:03:19 CEST] <klaxa> not gonna get any better than that [21:03:29 CEST] <Parker__> Mmm [21:03:34 CEST] <klaxa> you can *try* -preset placebo [21:03:36 CEST] <Parker__> if my images are mostly white background [21:03:42 CEST] <klaxa> but it will change it by 1-2% [21:03:46 CEST] <Parker__> is there anyway I could cut down on filesize? [21:04:02 CEST] <klaxa> it doesn't matter [21:04:21 CEST] <klaxa> the ppm images get decoded and then encoded with x264 [21:04:31 CEST] <Parker__> Mmmm [21:04:35 CEST] <Parker__> how can I make this smaller... [21:04:37 CEST] <klaxa> whether you save space in the images or not is irrelevant [21:04:49 CEST] <klaxa> use lossy compression [21:05:08 CEST] <Parker__> what would you recommend? [21:05:51 CEST] <klaxa> try -crf 23 and if it satisfies you stick with it, if not decrease by 2 or 3 [21:06:04 CEST] <klaxa> hint: use -t 10 to only encode 10 seconds of material [21:06:17 CEST] <klaxa> to check quality on a small sample [21:12:25 CEST] <DHE> BtbN: ffmpeg -threads -i INPUT -threads 1 -filter_complex "..." [-threads 1 -opts.... output1] [...repeat...] # is there a better syntax? [21:13:43 CEST] <DHE> this command runs ffmpeg with no threads except those produced by the call to configure_filtergraph(), so I think you're right about treating the threads bit there as a bug... [21:14:11 CEST] <DHE> I'm wondering if maybe a -filter_complex_threads option might be appropriate rather than taking -threads from the first output target [21:17:06 CEST] <DHE> or is that overcomplicating it? [21:20:57 CEST] <Parker__> klaxa: what about vp9? [21:21:11 CEST] <Parker__> do you have any experience with that lossy codec? [21:21:18 CEST] <klaxa> you can give it a try [21:25:38 CEST] <Parker__> maybe x265 [21:50:50 CEST] <furq> vp9 and x265 both have lossless modes [21:51:02 CEST] <DHE> BtbN: I'll see if I can make a patch to fix or improve things... the right way to fix it is still a bit of an open question though [23:06:29 CEST] <StrongBad> hi, can I get some help with my command syntax? I have an mp4 video file w/ audio and an m4a edited audio file. I'd like to replace the audio in the existing mp4 with the edited audio that I have. [23:06:54 CEST] <StrongBad> I've googled around for examples of other commands people have used, but I can't seem to find anyone doing exactly what I want to do [23:07:28 CEST] <StrongBad> edited audio is the m4a [23:11:42 CEST] <klaxa> ffmpeg -i video.mp4 -i audio.m4a -map 0:v -map 1:a -c copy replaced_video.mp4 [23:12:33 CEST] <klaxa> that will take the video from the 0th input (-map 0:v) and the audio from the 1st input (-map 1:a), copy the codecs and put it in replaced_video.mp4 [23:16:47 CEST] <StrongBad> this is what I just tried klaxa ffmpeg-3.1.4-win64-static\bin>ffmpeg -i C:\Users\$user\Desktop\ffmpeg-3.1.4-win64-static\files\video_trimmed.mp4 -i C:\Users\$user\Desktop\ffmpeg-3.1.4-win64-static\files\video_audio.m4a -map 0:v -map 1:a -c copy video_newaudio.mp4 [23:17:00 CEST] <StrongBad> it gives me this [mp4 @ 0000000002a25e00] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead. [23:17:21 CEST] <StrongBad> may be more there, lots to read [23:17:26 CEST] <StrongBad> but that was highlighted [23:18:26 CEST] <StrongBad> ah wait [23:18:32 CEST] <kepstin> StrongBad: so, did it work? what's the problem? [23:18:45 CEST] <StrongBad> I think I'm stupid [23:19:01 CEST] <StrongBad> I didn't see anything show up in the working folder [23:19:15 CEST] <StrongBad> but I see I didn't put a path for the output, so I think it went into bin [23:20:06 CEST] <kepstin> yes, if you don't use a full path, it'll write to the current directory [23:21:03 CEST] <StrongBad> wonderful [23:21:09 CEST] <StrongBad> thank you klaxa [23:21:13 CEST] <b6s3d> im am trying to convert a mp4 file in 720p so it will be playable on psp and i get this error [libxvid @ 0x86980c0] Invalid pixel aspect ratio 477/272, limit is 255/255 reducing [23:21:18 CEST] <llogan> StrongBad: you can ignore that codecpar message (or use the git version as recommended, not the release version) [23:21:47 CEST] <b6s3d> i am using this command: for file in *.mp4; do outfile='/media/user/B056-729D/VIDEO/'$file; ffmpeg -i "$file" -b 450k -s 320x240 -vcodec libxvid -ab 128k -ar 48000 -acodec aac -strict -2 "$outfile";done [23:22:29 CEST] <b6s3d> ok [23:24:34 CEST] <b6s3d> llogan: https://pastebin.mozilla.org/8920527 [23:26:32 CEST] <furq> i thought the psp supported h264 these days [23:27:42 CEST] <b6s3d> i think it just ignores the error or changes it to standard? [23:27:58 CEST] <b6s3d> i see a working file on the computer, but don't know yet if it will work on the psp [23:28:51 CEST] <furq> if that was a warning and not an error then yeah, it just reduced it to a valid par [23:29:01 CEST] <furq> but you shouldn't use xvid unless you have a good reason to [23:29:11 CEST] <furq> apparently the psp supports 480p h264 [23:33:10 CEST] <StrongBad> thanks again klaxa , kepstin llogan [23:34:16 CEST] <b6s3d> well, i think i have a good reason? i have a psp. [23:34:32 CEST] <b6s3d> 480p h264 you say furq [23:37:27 CEST] <furq> according to google [23:37:55 CEST] <b6s3d> could you share a link or two? [23:40:32 CEST] <furq> i can but it's in japanese [23:40:34 CEST] <furq> http://wiki.crav-ing.com/index.php?PSP [23:41:24 CEST] <furq> `-s 640x480 -level 3.0 -profile:v main` should be all you need [23:41:37 CEST] <furq> or -s 720x480 -aspect 16:9 for widescreen [23:42:03 CEST] <b6s3d> could you add it to my command that i shared with you in that paste? [23:42:16 CEST] <b6s3d> 720x480 would be great [23:42:36 CEST] <furq> ffmpeg -i "$file" -s 640x480 -level 3.0 -profile:v main -c:a aac -b:a 128k -ar 48000 "$outfile" [23:42:45 CEST] <furq> assuming outfile ends in .mp4 [23:45:06 CEST] <b6s3d> ayy works without error messaging :D [23:45:32 CEST] <furq> actually the native res is 480x272 so you probably just want to use that [23:45:43 CEST] <furq> 480p is only useful if you're plugged into a tv [23:46:29 CEST] <furq> you'll need -level 2.1 for that [23:47:54 CEST] <JEEB> ay [23:53:32 CEST] <b6s3d> and how would that look like furq? [23:58:27 CEST] <b6s3d> bam: ffmpeg -i "$file" -s 480x272 -level 2.1 -profile:v main -c:a aac -strict -2 -b:a 128k -ar 48000 "$outfile" [23:58:38 CEST] <b6s3d> video is supported, but it can't play the file ? [00:00:00 CEST] --- Fri Oct 21 2016

1 0

ffmpeg-devel.log.20161019
by burek 20 Oct '16

20 Oct '16

[01:40:48 CEST] <cone-095> ffmpeg 03James Almer 07master:47ee6f1a59b1: avformat/matroskadec: support parsing Chroma Location elements [02:01:01 CEST] <cone-095> ffmpeg 03James Almer 07n3.0.4:HEAD: avformat/matroskadec: support parsing Chroma Location elements [03:11:44 CEST] <cone-095> ffmpeg 03Aman Gupta 07master:f45d5e07ddbd: lavc/videotoolboxenc: skip SEI allocation when side data is not present [03:29:23 CEST] <atomnuker> alright, how the hell does this work [03:29:33 CEST] <atomnuker> ff_af_queue_remove gets called with an identical frame size [03:30:11 CEST] <atomnuker> yet on the last frame there are samples left (this is okay, that's what's meant to happen) [03:31:18 CEST] <atomnuker> but if I copt exactly what goes on in another encoder I get 0 samples left [03:40:34 CEST] <atomnuker> nevermind, missing avctx->initial_padding [04:10:54 CEST] <cone-095> ffmpeg 03James Almer 07master:ea201ad14cde: avformat/mov: pass the demuxer's AVFormatContext to avpriv_request_sample() [04:58:56 CEST] <cone-095> ffmpeg 03James Almer 07master:f5cfc0cc54ce: avformat/matroskadec: clarify some Colour element defaults using enum values [04:58:57 CEST] <cone-095> ffmpeg 03James Almer 07master:0ad71ed6f98e: avformat/matroskadec: fix BitsPerChannel element's default value [09:10:43 CEST] <cone-500> ffmpeg 03Muhammad Faiz 07master:23b6f880d6d7: avfilter/firequalizer: add cubic_interpolate function on gain [09:14:55 CEST] <Matador> Hi [09:16:10 CEST] <Matador> Anyone else having issues on QSV (libmfx) ? Couple bug reports already on it, hoping something can be done :| [09:16:18 CEST] <Matador> https://trac.ffmpeg.org/ticket/5859 [09:16:25 CEST] <Matador> https://trac.ffmpeg.org/ticket/5848 [09:26:51 CEST] <Chloe> What's the status on the QSV revert & sync with libav? [10:30:54 CEST] <jkqxz> Matador: First one looks like user error; MSS 2017 does work. At a guess, they need to look at their dynamic linker setup. [10:31:10 CEST] <jkqxz> Don't know about the second one, I haven't used it on Windows. [10:34:18 CEST] <BtbN> I also just wouldn't bother with QSV on Linux. It's bad and pointless. [11:00:31 CEST] <cone-500> ffmpeg 03Matthieu Bouron 07master:b8c158a4eddb: lavc/mediacodec_wrapper: do not discard codecs reporting they do not support any profile [11:00:32 CEST] <cone-500> ffmpeg 03Matthieu Bouron 07master:0f7fce87ea98: lavc: add vp8/vp9 mediacodec decoders [11:00:33 CEST] <cone-500> ffmpeg 03Matthieu Bouron 07master:f62c54456db0: lavc: add mpeg4 mediacodec decoder [11:00:34 CEST] <cone-500> ffmpeg 03Matthieu Bouron 07master:d5082a2ce776: lavc/mediacodec: use more meaningful filenames [13:02:51 CEST] <cone-500> ffmpeg 03Sven C. Dack 07master:da4d0fa86b48: avcodec/nvenc: add test for Temporal AQ support [13:02:52 CEST] <cone-500> ffmpeg 03Sven C. Dack 07master:aebbcb2706ab: avcodec/nvenc_hevc: Added missing option -temporal_aq [13:02:53 CEST] <cone-500> ffmpeg 03Sven C. Dack 07master:1aa8fa418a82: avfilter/vf_scale_npp: select cubic and lanczos as alternative where super-sampling is not supported [13:04:46 CEST] <thegeek_> I'm using libavcodec and I have a out-of-bounds access in ffv1enc.c, choose_rct_params() [13:05:24 CEST] <thegeek_> I've enabled version 4 (experimental), which is why that code path is used [13:07:20 CEST] <thegeek_> this is the offending statement: [13:07:21 CEST] <thegeek_> unsigned v = *((const uint32_t*)(src[0] + x*4 + stride[0]*y)) [13:07:53 CEST] <thegeek_> I'm using 16 slices on a 640x480 image, which means that each slice is 160x120 [13:08:32 CEST] <thegeek_> this function iterates through all pixels in the picture (y = 0; y<h) (x=0;x<w) [13:09:06 CEST] <thegeek_> it always crashes when y==119 and x == 120 [13:10:04 CEST] <thegeek_> the buffers it operates on are not allocated by me as I do not pass in a reference-counted AVFrame [13:10:27 CEST] <thegeek_> it seems to me that the code reads out of bounds on memory that I have not allocated [13:10:37 CEST] <thegeek_> which could be a bug [13:27:45 CEST] <cone-500> ffmpeg 03Jean Caille 07master:6498549fd75f: lavf/mov: support gopro hero moments udta tag [13:27:46 CEST] <cone-500> ffmpeg 03Michael Niedermayer 07master:18ad44d14548: avformat/movenc: Skip duration check for discontinuous fragments [13:27:47 CEST] <cone-500> ffmpeg 03Derek Buitenhuis 07master:dfe7e5501d4a: FATE: Add test for muxing discontinuous MP4 fragments with large timestamps [13:31:21 CEST] <michaelni> thegeek_, how can this be reproduced? [13:40:57 CEST] <thegeek_> michaelni: my current repro is tied to my custom code/application [13:41:12 CEST] <thegeek_> I shall try to create a repro with ffmpeg [13:41:35 CEST] <thegeek_> but I was wondering if this code is sound or very experimental [13:41:50 CEST] <thegeek_> I'm using v4 as it allocates much less memory [17:41:38 CEST] <jacereda> Looks like avienc.c doesn't write the "strf" tag for AVMEDIA_TYPE_DATA... any reason for that? if that tag is skipped, bad things will happend for data streams in avidec.c (like not setting codec_type). Or am I missing something? [17:56:18 CEST] <jacereda> OTOH, the MKTAG('s', 't', 'r', 'f') handler in avidec.c has an 'if (!size) break;' that will also avoid setting codec_type. [18:01:08 CEST] <jacereda> The commit message for that bit says "Do not try to parse empty strf tags.", but I fail to see why the empty strf should be problematic. [20:56:07 CEST] <Matador> qsv is a huge fail now on Windows + Ubuntu [20:56:10 CEST] <Matador> it worked months ago [21:57:22 CEST] <Compn> Matador : use months ago ffmpeg [21:57:30 CEST] <Compn> i heard the merges may have borked it ? [21:59:07 CEST] <JEEB> not sure if it was the merges that borked it, but IIRC the current state of it @ Libav was better than FFmpeg's (except for the nobody-cares MJPEG decoder) [21:59:14 CEST] <JEEB> or was it encoder as well? [21:59:54 CEST] <JEEB> in any case it IIRC was decided that since the maintainer didn't pop up here we'd reset to Libav's version during the next merges [22:00:06 CEST] <JEEB> that's just my understanding of things, though [22:03:24 CEST] <Chloe> JEEB: I asked about the status of that yesterday but no response. [22:04:04 CEST] <rcombs> the MJPEG decoder might be useful if lavc didn't fall apart on moderately large frame sizes [22:04:35 CEST] <JEEB> right [22:07:49 CEST] <rcombs> lol ints [23:15:00 CEST] <cone-736> ffmpeg 03Andreas Cadhalpun 07master:a5b4476a602f: avformat: prevent triggering request_probe assert in ff_read_packet [23:25:19 CEST] <tmm1> is there a filter that does the opposite of h264_mp4toannexb [23:26:02 CEST] <nevcairiel> no [23:26:09 CEST] <JEEB> I think the mov muxer did something like that [23:26:19 CEST] <nevcairiel> the opposite is trivial [23:26:30 CEST] <nevcairiel> which is why muxers that need it usually just do it internally [23:28:47 CEST] <nevcairiel> i think w e have some helper function they all call [23:29:46 CEST] <tmm1> cool i'll check it out, thanks [23:40:29 CEST] <cone-736> ffmpeg 03Carl Eugen Hoyos 07master:d8716864f967: lavf/riff: Document the interleaving of VSM4 video. [00:00:00 CEST] --- Thu Oct 20 2016

1 0

ffmpeg.log.20161019
by burek 20 Oct '16

20 Oct '16

[00:00:25 CEST] <t4nk834> i need help [00:00:34 CEST] <t4nk834> Invalid file index 1 in filtergraph description [0:v]yadif,scale=hd1080,setsar=1[b];[b][1:v]overlay=main_w-overlay_w-10:main_h-overlay_h-10. [00:00:48 CEST] <t4nk834> without yadif i have logo [01:17:07 CEST] <hseg> How do I replace the last second of a video by a black screen? [01:17:29 CEST] <hseg> i.e. I'd like the audio to keep playing, but no video data to be provided. [01:23:12 CEST] <furq> hseg: -vf "drawbox=t=iw:enable='between(t,999,1000)'" [01:23:17 CEST] <furq> assuming the video is 1000 seconds long [01:26:25 CEST] <hseg> Nice. t=iw makes the thickness of the box edges the entire width of the frame? [01:26:44 CEST] <furq> it sets the thickness to whatever the video width is [01:26:49 CEST] <hseg> and enable controls when the filter runs. [01:26:57 CEST] <furq> if you have vertical videos you'll probably want to make that more robust [01:27:25 CEST] <hseg> In what way isn't that robust? [01:27:48 CEST] <furq> never mind i'm stupid [01:28:08 CEST] <hseg> The left and right edges of the box will cover the entire video, right? [01:28:13 CEST] <furq> yeah [01:28:25 CEST] <hseg> Nice, thank you. [01:28:46 CEST] <hseg> Will only be able to test this tomorrow, as it's getting quite late here. [01:29:08 CEST] <hseg> BTW, any idea why inputs don't support -to? [01:30:18 CEST] <furq> nothing that springs to mind [01:30:29 CEST] <hseg> (also, I suppose I could put that filter within whatever filter_complex I'm using the video in) [01:30:45 CEST] <furq> yeah -vf and -filter_complex will overwrite each other [01:31:02 CEST] <hseg> OK [01:33:52 CEST] <hseg> Missed your answer re -to if you answered it. [01:34:01 CEST] <furq> 00:30:17 ( furq) nothing that springs to mind [01:34:42 CEST] <hseg> Ah. It's annoying, since it means that whenever I'm clipping I need to compute durations instead of giving start/stop timestamps. [01:35:28 CEST] <furq> if you're reencoding you can use -vf trim [01:36:08 CEST] <furq> it doesn't take duration specs though so you'd need to calculate seconds [01:36:45 CEST] <furq> actually it says it does but i remember people complaining that it either doesn't actually or it's impossible to escape correctly [01:38:52 CEST] <hseg> Hm. That looks useful, thanks. I'm reencoding because the concat filter is the only way I found that robustly concated stuff, but recent experiments suggest that I could have just used the demuxer instead to save time. [01:39:14 CEST] <hseg> Eh. It's an overnight job anyway, so I don't care as much. [01:46:55 CEST] <hseg> Thanks, and goodnight! [02:14:52 CEST] <jpsharp> okay, next question. I'm using ffmpeg to play an rtmp stream out to /dev/fb0. Everything bops along until what looks like the rtmp server changes files and the DTS from the server drops to 0. ffmpeq subsequently freaks out and stops playing video. [02:15:16 CEST] <jpsharp> I have +igndts in my -fflags, but ffmpeg still complains and hangs. [10:28:34 CEST] <k_sze[work]> Is there a way to get some meaningful error message in case sws_getContext fails? [10:28:36 CEST] <k_sze[work]> https://www.ffmpeg.org/doxygen/3.1/group__libsws.html#gaf360d1a9e0e60f906f7… [10:31:25 CEST] <nonex86> enable debug in ffmpeg api [10:31:32 CEST] <nonex86> verbose level for example [10:32:04 CEST] <k_sze[work]> But that's going straight to stderr, right? [10:32:26 CEST] <nonex86> you can redirect it [10:32:40 CEST] <nonex86> by providing own log function [10:32:50 CEST] <nonex86> if you hate stderr [10:35:04 CEST] <k_sze[work]> This? https://www.ffmpeg.org/doxygen/3.1/group__lavu__log.html#ga14034761faf581a8… [10:42:16 CEST] <nonex86> k_sze[work]: exactly [10:47:44 CEST] <k_sze[work]> There's no description of what the arguments are supposed to be. :( [10:48:00 CEST] <nonex86> ? [10:48:16 CEST] <nonex86> as a last resort you can use source code of ffmpeg [10:48:24 CEST] <nonex86> this helps ;) [10:49:14 CEST] <nonex86> static void log_callback(void*, int, const char*, va_list); [10:49:29 CEST] <nonex86> av_log_set_callback(FFMPEGLib::log_callback); [10:49:45 CEST] <nonex86> void FFMPEGLib::log_callback(void *avcl, int level, const char *fmt, va_list vl) [10:49:50 CEST] <nonex86> just an example [10:50:07 CEST] <nonex86> _vsnprintf_s(buf, kBufSize, _TRUNCATE, fmt, vl); [10:50:07 CEST] <nonex86> OutputDebugStringf("[0x%08x][FFMPEG] %s", GetCurrentThreadId(), buf); [10:51:18 CEST] <nonex86> and pass the message to ffmpeg default callback - av_log_default_callback(avcl, level, fmt, vl); [10:51:23 CEST] <nonex86> if you need [10:51:25 CEST] <nonex86> thats all [11:21:20 CEST] <k_sze[work]> Is it generally ok if I call avcodec_close or avcodec_free_context with an AVCodecContext that hasn't been flushed yet? [14:12:29 CEST] <ootje> what is the -filter_complex equivalent of -loop 1 -i image.png -t 2 ? tried -filter_complex "movie=image.png:loop=0, setpts=PTS-STARTPTS" -t 2 but it just renders on and on and on without stopping. tried adding ", trim=0:2" also but nope [14:38:00 CEST] <Kadigan_KSB> Hey. Quick question: when I encode webm using ffmpeg, I invariably end up with a opsnr.stt file. What flag do I need to use to disable this behavior? (from what I can see on the web, it's PSNR packet data?) [14:39:13 CEST] <Kadigan_KSB> I can work around the issue by deleting it afterwards, but can I disable this? [15:04:52 CEST] Action: Kadigan_KSB sighs [15:08:02 CEST] <Kadigan_KSB> http://pastebin.com/raw/7FKArVAX -- here you go [15:09:08 CEST] <furq> Kadigan_KSB: looks like your libvpx was built with --enable-psnr [15:09:16 CEST] <Kadigan_KSB> (The AVOption error is in reference to -b:a, since the input.mov file does not have an audio channel) [15:09:20 CEST] <Kadigan_KSB> Yeah, I was afraid of that. [15:09:29 CEST] <furq> what distro is that [15:10:01 CEST] <Kadigan_KSB> As in version? [15:10:11 CEST] <furq> if you want [15:10:18 CEST] <furq> i mostly just wanted to make a mental note to avoid it [15:10:41 CEST] <Kadigan_KSB> http://pastebin.com/raw/1hFLpvKh [15:10:55 CEST] <furq> i meant your linux distro [15:10:59 CEST] <Kadigan_KSB> Ah, no. [15:11:07 CEST] <Kadigan_KSB> It's OS X 10.9.5 [15:11:08 CEST] <furq> oh right, llvm [15:11:10 CEST] <furq> yeah [15:11:16 CEST] <furq> is this ffmpeg from homebrew or something [15:11:22 CEST] <Kadigan_KSB> Yeah, MacPorts. [15:11:46 CEST] <furq> if ports means like freebsd ports then it shouldn't be too hard to rebuild libvpx [15:12:03 CEST] <furq> but yeah you should report that upstream, it's definitely wrong for that to be default [15:12:04 CEST] <Kadigan_KSB> Well, it works perfectly well for my needs. The leftover file is merely a nuisance. [15:13:36 CEST] <Kadigan_KSB> Anyway, thanks for the assist. :) [16:05:15 CEST] <Kadigan_KSB> Hm... I have a second issue. :/ [16:05:26 CEST] <Kadigan_KSB> This one is much more about color grading that actual compression, but here goes. [16:05:48 CEST] <Kadigan_KSB> I check my videos at each step of the way, and they all look fine -- until they get to YouTube, or some other online player, whatever. [16:06:00 CEST] <Kadigan_KSB> For some reason I'm seeing a HUGE shift towards magenta. [16:06:13 CEST] <furq> what's the source pixel format [16:06:15 CEST] <Kadigan_KSB> Now, from what I can google up, this may be related to color profiles. [16:06:40 CEST] <furq> if it's not yuv420p then youtube will convert to that, and evidently it's not very good at it [16:07:09 CEST] <furq> if it's already yuv then it's probably not that [16:07:24 CEST] <Kadigan_KSB> It's most likely yuv422p10le [16:07:37 CEST] <Kadigan_KSB> (Prores source from a Canon cam) [16:07:42 CEST] <furq> it never hurts to do the format conversion yourself [16:07:44 CEST] <Kadigan_KSB> (well, Prores-encoded) [16:08:08 CEST] <Kadigan_KSB> Oh, wait - [16:08:12 CEST] <furq> assuming you're already transcoding [16:08:14 CEST] <Kadigan_KSB> for youtube I encode to .mkv [16:08:19 CEST] <Kadigan_KSB> and I do pix_fmt yuv420p [16:08:24 CEST] <furq> never mind then [16:08:52 CEST] <Kadigan_KSB> I also check the .mkv, at least using VLC (I'm on a Mac, it's hard to use the wonderful myriad of WINDOWS tools :D) [16:09:05 CEST] <Kadigan_KSB> and it checks out. But then on uploading, it doesn't. [16:09:08 CEST] <furq> check it with mpv in opengl mode if you can [16:09:21 CEST] <furq> it does sound like some kind of colourspace issue though [16:09:25 CEST] <Kadigan_KSB> What is MPV? [16:09:29 CEST] <furq> which isn't something i've ever had to deal with [16:09:38 CEST] <furq> https://mpv.io/ [16:09:57 CEST] <Kadigan_KSB> I'm working on calibrated displays, so there's a big chance the ICC profile gets saved to the video at SOME point. [16:10:11 CEST] <furq> i didn't know icc profiles in video was even a thing [16:10:34 CEST] <furq> but if it is then it's probably safe to assume youtube wants nothing to do with it [16:11:01 CEST] <Kadigan_KSB> Actually, from what I'm lead to believe by folk at CreativeCow, [16:11:15 CEST] <Kadigan_KSB> it's ffmpeg and other tools that don't, and YouTube/Vimeo apparently RESPECT the profiles. [16:12:03 CEST] <furq> if the transcoded version plays back ok then that shouldn't be the issue [16:12:09 CEST] <furq> unless it's being passed through as metadata [16:13:17 CEST] <Kadigan_KSB> I thought I had that killed via -map_metadata -1 [16:14:41 CEST] <furq> maybe vlc is using your monitor's default icc and youtube is using a different one [16:14:50 CEST] <Kadigan_KSB> It's possible. [16:15:34 CEST] <wildefyr> best way to compile ffmpeg on windows under msys? configure complains heavily about not to use msys, but instead use mingw, but I thought msys was built on mingw? [16:16:45 CEST] <Kadigan_KSB> wildefyr: there's an msys-based autobuilder for building under win32 and win64 [16:16:51 CEST] <Kadigan_KSB> -- what I know [16:16:53 CEST] <wildefyr> link? [16:17:19 CEST] <Kadigan_KSB> https://github.com/jb-alvarado/media-autobuild_suite [16:17:31 CEST] <wildefyr> that'd be great, but I'm attempting to build ffmpeg with the nvidia sdk to enable the nvenc encoder [16:18:08 CEST] <Kadigan_KSB> furq: I have mpv -- what do I do? [16:18:36 CEST] <furq> mpv --vo=opengl filename [16:18:50 CEST] <nonex86> i build ffmpeg on msys with ms vc++ without any problems [16:18:59 CEST] <furq> there are some options for using icc profiles as well [16:19:37 CEST] <furq> https://mpv.io/manual/stable/#video-output-drivers-target-prim [16:19:41 CEST] <furq> that and a few below it [16:19:55 CEST] <wildefyr> nonex86, it's odd, configure refuses to 'make' a MakeFile [16:20:06 CEST] <wildefyr> latest git version here [16:20:09 CEST] <Kadigan_KSB> Well, I don't know if it ran in OpenGL (seeing as Mac OS X has only OpenCL?) but it looked fine. [16:21:58 CEST] <nonex86> ffmpeg version 3.1.4 Copyright (c) 2000-2016 the FFmpeg developers [16:21:58 CEST] <nonex86> built with Microsoft (R) C/C++ Optimizing Compiler Version 18.00.40629 for x64 [16:21:58 CEST] <nonex86> configuration: --toolchain=msvc --enable-static --disable-shared --enable-avresample --enable-gpl --enable-libx264 --enable-libx265 --enable-libvpx --enable-libmp3lame --enable-libmfx --enable-nvenc --enable-nonfree --enable-libkvazaar --enable-libvorbis --enable-libopus --enable-libxvid [16:22:04 CEST] <nonex86> something like that... [16:23:26 CEST] <wildefyr> okay I'll --toolchain=msvc [16:24:57 CEST] <nonex86> ah.. forgot one thing, 1) i patched configure script myself 2) i did some modifications to result makefile :) [16:25:43 CEST] <nonex86> ...two things to be correct, yeah :) [16:25:49 CEST] <wildefyr> yeah that's probably what I am going to have to do myself.. [16:25:51 CEST] <wildefyr> what a pain [16:28:43 CEST] <termos> Does anyone know why the sound is not playing on this stream? rtmp://cp353594.live.edgefcs.net/live/1003_1003_1364k@316197 [17:04:52 CEST] <Kadigan_KSB> http://s10.postimg.org/u6nkqoj61/Screen_Shot_2016_10_19_at_17_03_13.png this is the kind of difference I see between (left) HTML5 video and (right) local playback of the file after downloading it from the HTML5 video element. [17:04:56 CEST] <wildefyr> is there really any benefit to using cbr over vbr? [17:04:58 CEST] <Kadigan_KSB> That's just sad. [17:05:13 CEST] <wildefyr> cvf* [17:05:29 CEST] <Kadigan_KSB> cvf? [17:05:38 CEST] <furq> cfr? [17:05:41 CEST] <wildefyr> nah I did mean cbr [17:05:50 CEST] <furq> not really [17:06:11 CEST] <Kadigan_KSB> There is, if you're streaming or using a device that has fixed buffer requirements. [17:06:13 CEST] <furq> some devices require cbr but those are few and far between [17:06:24 CEST] <furq> it's pretty much worthless for streaming [17:06:28 CEST] <furq> just use vbv for that [17:06:46 CEST] <wildefyr> i thought as much [17:06:50 CEST] <furq> assuming we're talking about x264 here [17:06:57 CEST] <wildefyr> well x265 [17:07:20 CEST] <furq> that's more or less the same afaik [17:07:39 CEST] <furq> x264 doesn't have a true cbr mode, the closest you can get is vbv and padding undersized packets [17:07:56 CEST] <furq> which is required for bluray iirc [17:08:05 CEST] <furq> but if you're not specifically targeting something along those lines, it's pointless [17:10:19 CEST] <ritsuka> Kadigan_KSB: which player? probably one is doing color correction and the other YOLO [17:16:27 CEST] <Kadigan_KSB> ritsuka: OS X preview app (should be using QT) [17:16:50 CEST] <ritsuka> and the browser? [17:19:48 CEST] <Parker__> Hey ffmpeg, I'm trying to convert a series of rendered .ppm frames into a lossless movie [17:19:51 CEST] <Parker__> ffmpeg -i untitled.%05d.ppm -c:v libx264 -preset veryslow -qp 0 out.mkv [17:20:03 CEST] <Parker__> seems to work, but the resulting movie is very blurry compared to the ppm files going in. [17:20:40 CEST] <BtbN> make sure you use a lossless pixel format as well [17:20:49 CEST] <DHE> -pix_fmt yuv444 maybe? [17:21:01 CEST] <DHE> note that some players, especially hardware players, can't play these back [17:21:43 CEST] <Parker__> Mmm [17:22:14 CEST] <nonex86> btw, about qp=0, will libx264 convert it to qpmin? or 0 will be used? [17:22:30 CEST] <kepstin> Parker__: is the input rgb images? if so, you probably want to use '-c:v libx264rgb' instead [17:22:42 CEST] <kepstin> Parker__: of course, that'll have even less compatibility :) [17:23:13 CEST] <kepstin> nonex86: qp=0 is a special value that switches the encoder to lossless mode. [17:23:19 CEST] <Parker__> I just want to make clean videos of my molecular dynamics movies :( [17:23:31 CEST] <Parker__> not sure if still RGB [17:23:32 CEST] <nonex86> kepstin: i see, thanks [17:24:22 CEST] <DHE> nonex86: qpmin and qpmax are for quality (ie: target bitrate) mode to put a cap on how high or low it will go [17:25:12 CEST] <nonex86> DHE: roger, thanks :) [17:25:25 CEST] <Parker__> Mmm, ffmpeg -r 25 -pix_fmt yuv420p -i untitled.%05d.ppm -c:v libx264rgb -preset veryslow -qp 0 out.mkv [17:25:28 CEST] <Parker__> and ffmpeg -r 25 -pix_fmt yuv420p -i untitled.%05d.ppm -c:v libx264rgb -preset veryslow -qp 0 out.mkv [17:25:31 CEST] <Parker__> and ffmpeg -r 25 -pix_fmt yuv420p -i untitled.%05d.ppm -c:v libx264 -preset veryslow -qp 0 out.mkv [17:25:36 CEST] <Parker__> still look very blurry [17:26:22 CEST] <DHE> Parker__: try this: ffmpeg -r 25 -i untitled.%05d.ppm -pix_fmt yuv444 -c:v libx264 -preset slow -qp 0 out.mkv [17:26:48 CEST] <kepstin> the only way the libx264rgb one could look blurry is if your player is rendering them incorrectly, that will exactly preserve the pixels from your input as-is. [17:26:58 CEST] <Parker__> I'm on linux using VLC [17:27:17 CEST] <kepstin> DHE: I'm pretty sure most versions of ffmpeg will use yuv444 by default in this case? (and print out a warning about it) [17:27:34 CEST] <kepstin> Parker__: it would be helpful if you pastebin the ffmpeg full output somewhere. [17:27:53 CEST] <Parker__> Yes [17:27:58 CEST] <Parker__> okay also this is really weird [17:28:03 CEST] <DHE> kepstin: i'm not 100% sure honestly. yuv420p (unless overridden) sure sounds like a sane default for x264 [17:28:09 CEST] <Parker__> watching the video for the first time on VLC, it is blury [17:28:11 CEST] <Parker__> letting it loop [17:28:14 CEST] <Parker__> it becomes clear... [17:28:22 CEST] <kepstin> Parker__: sounds like a vlc bug then :/ [17:28:28 CEST] <furq> Parker__: try with mpv [17:29:00 CEST] <DHE> .. what if you play the .ppm files directly with ffplay? [17:29:20 CEST] <furq> and yeah x264 defaults to yuv444p with an rgb source [17:30:16 CEST] <Parker__> Installing MPV now [17:30:20 CEST] <Parker__> how do you play with ffplay DHE? [17:30:45 CEST] <furq> i expect `ffplay img%05d.ppm` works [17:31:03 CEST] <Parker__> ffply looks great [17:31:49 CEST] <Parker__> kepstin: furq it is a VLC bug [17:31:52 CEST] <Parker__> MPV looks great [17:31:55 CEST] <Parker__> so does ffplay [17:32:00 CEST] <Parker__> thanks for the help everyone! [17:35:05 CEST] <wildefyr> morale of the lesson: mpv is god [17:50:03 CEST] <bryan_> hello everybody [17:53:25 CEST] <bryan_> i am trying to add dri2 api calls in ffmpeg gstreamer plugin...it is getting compiled but at runtime it shows undefined reference to dri2 library...how can i link dri2 lib inside ffmpeg plugin? [17:54:53 CEST] <bryan_> i want to resolve dri2 symbols at compile time itself [18:00:09 CEST] <jkqxz> bryan_: You'd need to add libxcb-dri2 (presumably this is using XCB?) to the ffmpeg configure script. See the other X extensions which are already there (libxcb-shm, say). [18:00:40 CEST] <jkqxz> Also I'm rather curious what you're going to do with it... [18:02:09 CEST] <bryan_> jkqxz: thank you...will try that.. [18:03:41 CEST] <bryan_> i am using libdri2.so library.. [18:10:39 CEST] <jkqxz> Huh, legacy libX11. Can I suggest using xcb instead? [18:13:13 CEST] <jkqxz> If you really want to use it libX11-style, you need to add a line like the one at configure:5978 for shm. It can probably just go at the top level. [18:20:29 CEST] <jacereda> hi [19:36:54 CEST] <Kentoo> Hi, I am using libav, is it possible to open an output file without truncating it? I mean I want to edit it, for example adding a subtitle stream [19:37:36 CEST] <kepstin> Kentoo: no, you'll generally have to read the file as input, and remux to a new file (which can be renamed back) [19:38:14 CEST] <Kentoo> kepstin: alright, thank u [19:38:15 CEST] <kepstin> none of the video container formats are designed for in-place editing, it would be really complicated and require moving a lot of data around, just writing to a new file is easier (and probably faster too) [19:38:30 CEST] <Kentoo> kepstin: so all video editing software use the method u said? [19:39:00 CEST] <kepstin> pretty much yeah. There's a few specialized tools for specific formats that allow limited in-place changes, but not anything like adding a new stream. [20:03:30 CEST] <ac_slater> hey guys. Does anyone have a diagram for 10bit YUV422? [22:18:36 CEST] <kuroro_> hello. im thinking about using static build of ffmpeg for android via https://johnvansickle.com/ffmpeg/ [22:18:52 CEST] <kuroro_> but im not sure which one to choose (armhf or armel) version [22:20:43 CEST] <furq> you probably want armhf [22:21:17 CEST] <kuroro> is it for more device compatibility? (more devices use armhf?) [22:21:58 CEST] <furq> armhf normally refers to armv7 [22:22:11 CEST] <furq> some older phones will need armel [22:22:23 CEST] <furq> and some very new phones will use aarch64, which i guess you'll need to build yourself [22:23:27 CEST] <furq> i'm not sure if those builds will work on android though [22:24:18 CEST] <kuroro> "i'm not sure if those builds will work on android though" - ah, so there might something different in android that might make them not work then.. [22:24:26 CEST] <furq> different libc [22:24:50 CEST] <furq> i don't know whether that'll actually break anything though [22:25:27 CEST] <kuroro> i see [00:00:00 CEST] --- Thu Oct 20 2016

1 0

ffmpeg-devel.log.20161018
by burek 19 Oct '16

19 Oct '16

[00:30:00 CEST] <cone-441> ffmpeg 03Carl Eugen Hoyos 07master:a20f3238be93: lavf/avidec: Do not fail for very large idx1 tags. [01:13:46 CEST] <atomnuker> jamrial_: to convert e.g. abs_pow34 to avx2 I just need to do "vinsertf128 m0, m0, [inq+sizeq], 1" to load the upper part of the register, right? [01:24:42 CEST] <atomnuker> (also would cvtsi2ss m3, dword maxvalm work on non-windows non-unix64 systems?) [01:27:34 CEST] <jamrial_> atomnuker: no, movaps with ymm registers will load 32 bytes instead of 16. vinsertf128 is when you want to duplicate 16 bytes in a ymm register [01:28:35 CEST] <jamrial_> and regarding cvtsi2ss, only unix64 has maxval already in a gpr regardless of how you init the function. x86_32 (any os) and win64 have it on stack [01:29:56 CEST] <jamrial_> so to answer the question, yes :p [01:34:14 CEST] <atomnuker> why does every single avx2 asm use vinsertf128 then? [01:34:34 CEST] <atomnuker> I thought it was some magic to load the upper part of the register [01:36:51 CEST] <jamrial_> it's used to insert 16 bytes of data in the upper half of the register, but it's not the only way to achieve that [01:39:08 CEST] <jamrial_> "movaps ymm0, [mem]" loads 32 bytes from memory, "movaps ymm0, ymm1" moves all 32 bytes from ymm1 to ymm0, etc [01:39:11 CEST] <atomnuker> so movaps will load 32 bytes but e.g. movu/mova won't? [01:39:44 CEST] <jamrial_> mova/movu are macros that expand to mov[au]ps or movdq[au] [01:40:25 CEST] <jamrial_> all of them can load 32 bytes if you use ymm regs [01:43:31 CEST] <atomnuker> well, if all I had to do was INIT_YMM avx2 then I think maybe something else isn't aligned since I segfault [01:44:16 CEST] <jamrial_> probably the in and out buffers. they should be 32 bytes aligned, and padded [01:45:25 CEST] <jamrial_> also, the splatd/shufps stuff wont work just like that with ymm regs. for those you'll probably need to use vinserft128 to fill the upper 16 bytes [01:46:43 CEST] <jamrial_> or vbroadcastss [01:50:17 CEST] <atomnuker> as for fixing the ARM build, just putting if (ARCH_X86) ff_aac_dsp_init_x86() should work, shouldn't it? [01:50:27 CEST] <atomnuker> it's what aacpsdsp does [01:55:55 CEST] <jamrial_> atomnuker: yeah [02:02:32 CEST] <atomnuker> disappointing, no real performance improvements switching abs_pow34 to avx2 [02:07:58 CEST] <jamrial_> is the loop really processing eight floats at a time instead of the four from the sse/sse2 version? [03:39:28 CEST] <cone-134> ffmpeg 03Zhao Zhili 07master:7853d838a6e4: avformat/tests/gitignore: add fifo_muxer entry [04:16:36 CEST] <kierank> 1:02 AM <"atomnuker> disappointing, no real performance improvements switching abs_pow34 to avx2 [04:16:40 CEST] <kierank> avx2 is integer, no [04:16:45 CEST] <kierank> isn't it just normal AVX [04:35:36 CEST] <Zeranoe> It looks like '-Wl,--image-base,0x140000000' is causing a compiler error with GCC 6.2.0 when snappy is included. This seems specific to 64-bit. It looks like that ldflag was introduced with a58c22d61260941fc651add73836882d5b112fdb [04:38:37 CEST] <rcombs> Zeranoe: snappy? [04:38:45 CEST] <rcombs> and what error? [04:39:22 CEST] <Zeranoe> rcombs: libstdc++-v3/src/c++11/cow-stdexcept.cc:236:(.text$_Z35_txnal_cow_string_C1_for_exceptionsPvPKcS_+0x2c): relocation truncated to fit: R_X86_64_PC32 against undefined symbol `_ITM_RU1' [04:40:03 CEST] <Zeranoe> rcombs: https://github.com/google/snappy [10:06:11 CEST] <cone-762> ffmpeg 03Muhammad Faiz 07master:2c1be03cb38f: fate: add swr-convertaudio test [15:02:19 CEST] <cone-095> ffmpeg 03Carl Eugen Hoyos 07master:31a0a8421658: lavf/avidec: Be more verbose when ignoring very large tag size. [15:21:34 CEST] <cone-095> ffmpeg 03Michael Niedermayer 07master:2bd99564540a: doc/examples/demuxing_decoding: Drop AVFrame->pts use [15:32:44 CEST] <mateo`> michaelni: should I push your mediacodec patch ? Or should I ? [15:38:14 CEST] <michaelni> mateo`, sure feel free to push, its still on my todo but i half forgotten [15:41:35 CEST] <cone-095> ffmpeg 03Michael Niedermayer 07master:9545ff3ec391: avcodec/mediacodec: Factor duplicate include [16:26:02 CEST] <wm4> ffmpeg demuxers now output broken DTS timestamps [16:26:11 CEST] <wm4> because of that shitty google edit list patch [16:26:14 CEST] <wm4> ... [16:26:35 CEST] <wm4> I guess I'm supposed to fix them somehow, because the finder of a bug gets to keep the bug [16:32:04 CEST] <nevcairiel> for mov only then, i assume? [16:35:39 CEST] <wm4> lol the patch adding AV_PKT_FLAG_DISCARD didn't even bump any library versions [16:36:23 CEST] <wm4> only the AVFrame change has a bump [16:37:11 CEST] <wm4> god, what an idiotic patch [16:37:25 CEST] <wm4> looking forward to breaking it shit all over the place [17:38:49 CEST] <cone-095> ffmpeg 03Muhammad Faiz 07master:acd74f92009d: swresample/resample: fix return value of build_filter [17:44:12 CEST] <cone-095> ffmpeg 03Muhammad Faiz 07master:d3be186ed1bc: avfilter/firequalizer: add dumpfile and dumpscale option [17:56:10 CEST] <atomnuker> jamrial: what's wrong with if (ARCH_X86) ff_aac_dsp_init_x86(s);? [18:10:01 CEST] <nevcairiel> That should be fine, assuming the prototype always exists as well [18:16:12 CEST] <atomnuker> it's the way it's done in every init function out there so it should be fine [18:23:52 CEST] <BtbN> Isn't stuff like that the reason ffmpeg needs at least O1? [18:24:02 CEST] <BtbN> So the symbol gets optimized out [18:25:24 CEST] <atomnuker> yep [18:27:30 CEST] <jamrial> atomnuker: nothing wrong with it. why do you ask? [18:29:03 CEST] <atomnuker> you put a "[...]" below that change [18:32:41 CEST] <kierank> that means [snip] [18:34:46 CEST] <atomnuker> makes sense now [18:47:10 CEST] <Chloe> timestamps are broken? [18:48:59 CEST] <Chloe> ugh. libav changing file names of hevc asm [19:18:28 CEST] <Chloe> https://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/utils.c;h=b0345b… This comment about demuxer_skip_samples sucks, why should the discard frame flag, ignore skip_samples set by the decoder? [19:20:40 CEST] <ubitux> "comment"? [19:20:48 CEST] <ubitux> the only comment i see is related to pkt copy [19:21:17 CEST] <Chloe> line 2358 [19:21:32 CEST] <Chloe> I just set it to start on the first relevant line [19:27:59 CEST] <Chloe> does anyone actually know how the edit list patches work [20:16:17 CEST] <BBB> wbs: ping [20:21:05 CEST] <wbs> BBB: pong [20:21:30 CEST] <BBB> wbs: how large are the aarch64 registers? 16byte (like sse) or 32byte (like avx)? [20:21:57 CEST] <wbs> BBB: 16 byte registers, 32 such registers [20:22:04 CEST] <BBB> yuck :( [20:22:05 CEST] <BBB> ok [20:22:41 CEST] <BBB> thats the opposite of the avx problem (32byte registers, but only 16 of em) [20:23:00 CEST] <BBB> the simd numbers look really good, nice job [20:23:33 CEST] <wbs> it's a step up from 32 bit arm though, where you have 16 x 16 bytes registers, or 32 x 8 (you can treat them interchangably like either of those) [20:23:54 CEST] <BBB> hm... [20:24:03 CEST] <BBB> so can you treat them like 16 32-byte registers on aarch64? [20:24:29 CEST] <wbs> no [20:24:58 CEST] <BBB> <- arm n00b [20:25:46 CEST] <cone-095> ffmpeg 03Jon Toohill 07master:81f4f789de7c: lavc/libmp3lame: send encoder delay/padding in packet side data [20:25:47 CEST] <ubitux> using them as 32x8 can be a bitch though [20:25:47 CEST] <cone-095> ffmpeg 03Jon Toohill 07master:3b02f6dd7be8: lavf/mp3enc: write encoder delay/padding upon closing [20:25:56 CEST] <ubitux> because the high part is not as accessible [20:26:11 CEST] <ubitux> it's often use 8 lower or all 16 [20:26:26 CEST] <wbs> ubitux: no, I meant on 32 bit arm, where you have d0-d31 [20:26:42 CEST] <ubitux> yeah i was talking about aarch64 [20:26:53 CEST] <wbs> on aarch64, using the high half as a separate register (so you'd have 64 registers) isn't really an option no [20:27:11 CEST] <wbs> or it's kinda inconvenient at least [20:28:18 CEST] <ubitux> btw, there are a bunch of places in your macro where you could pass the suffix in the param [20:28:30 CEST] <ubitux> to avoid the annoying \() [20:29:07 CEST] <wbs> can you point out which one? in some cases I've chosen that to keep the macro invocation a bit more readable, but I'm wildly inconsistent about it [20:29:34 CEST] <ubitux> my eyes were on the idct [20:29:52 CEST] <ubitux> dunno if you did it elsewhere [20:30:16 CEST] <wbs> right; those are mostly intentional [20:30:19 CEST] <philipl> BtbN: what's next for dynlink cuda? [20:30:35 CEST] <BtbN> sending it to the ML. [20:31:00 CEST] <wbs> because the idct/iadst transforms are absolutely mindnumbing anyway, even without the extra boilerplate of those suffixes there; I much rather keep the macros a bit bulkier [20:31:51 CEST] <wbs> BBB: anyway, once the patches are merged, I'll post full benchmarks of pre/post speedups and such. I still occasionally get a few fps more when finding more things to optimize [20:32:23 CEST] <BBB> cool [20:32:44 CEST] <BBB> these new aarch64 devices are all multicore, right? [20:33:01 CEST] <wbs> yes, most are quad at least [20:33:04 CEST] <wbs> my phone has got 8 cores [20:33:22 CEST] <wbs> (4 low power ones and 4 high power ones, but all of them have got simd) [20:33:41 CEST] <wbs> iOS devices usually have got much fewer cores, but higher performance per core [20:34:16 CEST] <BBB> 4, nice [20:34:34 CEST] <wbs> yup, it's super nice for things like this [20:35:58 CEST] <wbs> (although the phone clocks down pretty soon if you push it too much) [20:36:20 CEST] <BBB> hm :( thats a little sad [20:36:37 CEST] <BBB> but even then, 4 cores, with simd should probably give you 1080p realtime right? [20:36:54 CEST] <wbs> earlier I've benched 208 fps for sintel, 93 fps for tos and 50 fps for etv, out of your sample collection [20:37:15 CEST] <ubitux> i'd assume most of these new aarch64 devices have a vp9 hw decoder anyway [20:37:27 CEST] <ubitux> which is likely much more interesting to use most of the time [20:37:33 CEST] <ubitux> i wonder where it would be useful [20:39:14 CEST] <wbs> ubitux: at least my phone only have got hevc, no vp9 [20:39:34 CEST] <ubitux> ah? interesting; mine seems to have vp9 [20:39:45 CEST] <ubitux> that's some pretty decent speed btw [20:40:24 CEST] <wbs> yeah, if the decoder itself can decode on like 100 fps, it probably won't get insanely hot when decoding realtime, and probably won't clock down all too much [20:41:37 CEST] <wbs> hmm, my phone has got a snapdragon 810, and at least wikipedia claims it has got even vp9 encoding (so I'd assume decoding as well). but perhaps sony didn't care to include the driver for that [20:43:32 CEST] <BBB> 50 for etv is pretty good, that clip is pretty hard and high bitrate [20:44:22 CEST] <wbs> yeah. and let's see what numbers I get once I get it merged. at least for loop filter (which affects sintel the most) I think I've got a bit of more speedup coming [20:44:36 CEST] <wbs> (but I don't benchmark on the phone all too often because it's a bit more annoying than directly on a devboard) [21:33:55 CEST] <mateo`> ubitux: software decoding is always interesting, testing, fallback in case the hw api is busy with other things [22:33:19 CEST] <Zeranoe> Seems pretty quiet on the Snappy project end... [22:33:32 CEST] <Zeranoe> But hopefully they have something to say https://groups.google.com/forum/#!topic/snappy-compression/4QzFeWFaTlU [22:35:48 CEST] <BtbN> Well, there's Zstandard now. [22:40:40 CEST] <Zeranoe> Are there any usage statistics for FFmpeg versions? I'm considering building releases from different release braches (3.1, 3.0), but I'm curious if there's even a demand [22:41:34 CEST] <cone-095> ffmpeg 03Rostislav Pehlivanov 07master:d2ae5f77c61a: aacenc: add SIMD optimizations for abs_pow34 and quantization [23:16:11 CEST] <Compn> Zeranoe : for what kind of user ? e.g. anyone using a release is just using that release because of distro package [23:16:23 CEST] <Compn> anyone using git is because of using git or one of your nightly builds [23:25:39 CEST] <cone-095> ffmpeg 03Michael Niedermayer 07release/3.0:5771a0c8237d: doc/examples/demuxing_decoding: Drop AVFrame->pts use [00:00:00 CEST] --- Wed Oct 19 2016

1 0

ffmpeg.log.20161018
by burek 19 Oct '16

19 Oct '16

[00:37:09 CEST] <Phi> The libx264 still isn't taking any options [00:37:58 CEST] <c_14> Try convincing it that if it invest now the stock price will only rise in the future. [00:38:35 CEST] <c_14> (I'm going to need more information if I'm going to try and help you) [00:40:22 CEST] <Phi> mm [00:40:34 CEST] <Phi> but I thought everyone here was psychic! [00:40:50 CEST] <c_14> Not on Monday.s [00:41:12 CEST] <Phi> fair enough [00:41:14 CEST] <Phi> http://stackoverflow.com/questions/40053873/ffmpeg-rtsp-stream-to-mpeg4-h26… [00:41:36 CEST] <Phi> basically, using AVFormatContext to chuck a H264 stream into a MPEG4 container, using libx264 encoder [00:41:50 CEST] <Phi> I can't change any of the options, the profile is always High [00:42:14 CEST] <Phi> av_opt_set(video_file_codec_context->priv_data, "profile", "main", AV_OPT_SEARCH_CHILDREN); [00:42:20 CEST] <Phi> seems to be the main suggestion [00:43:52 CEST] <Phi> I've checked the examples and I've even proxied libx264 Codec's Init function pointers, but they're not even called [00:46:56 CEST] <Phi> what's weirder is I check the X264Context, and the encoder variable isn't even set [00:47:07 CEST] <Phi> yet a mp4 file is produced, which is very confusing [00:51:35 CEST] <Phi> any idea c_14? [00:56:03 CEST] <c_14> If anything, you should be calling av_opt_set directly before marker3 [00:59:32 CEST] <Phi> should I be calling avcodec_open2? [00:59:35 CEST] <Phi> it doesn't seem used [00:59:56 CEST] <c_14> It's in the examples, so probably [01:00:34 CEST] <c_14> https://git.ffmpeg.org/gitweb/ffmpeg.git/blob_plain/HEAD:/doc/examples/deco… [01:08:44 CEST] <Phi> I get the feeling I'm doing something weird [01:10:00 CEST] <Phi> because I use AVIOContext, AVFormatContext, AVCodecContext... [01:10:18 CEST] <Phi> that example isn't very accurate, it's just input -> images, which I can do [01:10:44 CEST] <Phi> I need output video file [01:11:17 CEST] <Phi> I mostly pulled my code from http://stackoverflow.com/questions/9251747/record-rtsp-stream-with-ffmpeg-l… [01:47:05 CEST] <blue_misfit> I'm seeing something crazy.. if I made an H.264 proxy with no b-frames of an editing codec like ProRes or DNxHD then I can drop it on top of the source in Premiere or Resolve and it's absolutely perfect... but if I enable b-frames it's immediately out of sync [01:47:10 CEST] <blue_misfit> has anyone else observed this?? [01:48:37 CEST] <DHE> graphical glitches, or just audio drift? [01:49:40 CEST] <blue_misfit> immediate audio sync offset equal to the number of bframes [01:49:50 CEST] <blue_misfit> sorry not audio sync offset [01:51:17 CEST] <blue_misfit> the version with b-frames is slipped back by a few frames [01:51:27 CEST] <blue_misfit> equal to the number of consecutive b-frames the proxy is encoded with [01:56:21 CEST] <blue_misfit> so like if a video normally starts at 00:00:00:00 [01:56:26 CEST] <blue_misfit> it will start at 00:00:00:02 [01:56:34 CEST] <blue_misfit> if you encode with 3 consecutive b-frames [01:58:03 CEST] <blue_misfit> hmm actually looks like it's just always a 2 frame offset.. using 3 bframes or 10 bframes always shows the same result [02:09:02 CEST] <lindylex> Can I get ffmpeg to get the title of the youtube video file and not download the video? [02:10:07 CEST] <furq> lindylex: use youtube-dl [02:10:21 CEST] <lindylex> Thanks [02:10:25 CEST] <furq> http://vpaste.net/6IHoZ [02:10:48 CEST] <lindylex> furq: you are the best! [02:15:45 CEST] <blue_misfit> DHE, looks like it actually works with 1 b-frame but any more than that and you immediately have a 2 frame offset in Premiere [02:15:51 CEST] <blue_misfit> very odd. I'm chatting with their support about this [02:19:20 CEST] <DHE> b-frames are decoded and displayed out of order. poor players or buggy muxers may screw it up resulting in that... [02:19:34 CEST] <DHE> but I don't know what app you're using so I'm just going to stop here [02:20:00 CEST] <furq> he said it's premiere [02:20:07 CEST] <blue_misfit> yeah - Premeiere Pro 2015.4 [02:20:09 CEST] <blue_misfit> it's fun stuff [02:20:25 CEST] <furq> doesn't premiere use the system (dshow/quicktime) codecs [02:20:26 CEST] <DHE> still havn't used it... [02:20:37 CEST] <blue_misfit> maybe - it also might use its own internal codecs [02:20:59 CEST] <blue_misfit> per support "I believe this is by Design with THE GOP size(in frames) , Its a general rule that when I frames is set to 1 then all frames are I frmaes, but when its largesr than 1 its then when the action takes place. That is why its coming as 00:00:00:02" [02:21:00 CEST] <blue_misfit> LOL [02:21:26 CEST] <furq> what about p frames [02:21:34 CEST] <furq> or frmaes, rather [02:22:11 CEST] <DHE> uhhh... [02:22:39 CEST] <DHE> while all I-frames are technically valid and recognized as a profile, nobody does that except for special situations [02:22:54 CEST] <furq> yeah no bframes has nothing to do with gop size [02:24:11 CEST] <furq> if it's happening in resolve as well then i'd suspect it's an issue with the system decoders [02:24:21 CEST] <furq> which i'm sure will be fun to, uh, resolve [02:24:56 CEST] <DHE> I would guess something is reading the frames one at a time from the file, and presenting them one at a time as they come out of the decoder without really understanding the need to buffer them.. [02:25:14 CEST] <blue_misfit> yeah agreed [02:25:26 CEST] <blue_misfit> this is crazy though! It's Premiere!! It's widely known and trusted for editing video! [02:25:49 CEST] <DHE> still, b-frames are business as usual for H1264 and h264 has been around for a while now. this is a rookie sort of error... [02:26:02 CEST] <blue_misfit> of course [02:26:23 CEST] <blue_misfit> there's definitely nothing wrong with these files - decoding them back to y4m returns the same number of bytes so it's not like frames are being lost [02:26:28 CEST] <blue_misfit> they're also totally there in Elecard [02:52:50 CEST] <Phi> I might join later, cheers for your help c_14 [03:35:54 CEST] <beauty> Is there a avformat_copy_context funtion like "avcodec_copy_context" [03:37:54 CEST] <DHE> like avcodec_parameters_from_context ? [03:38:07 CEST] <DHE> part of the codecpar system [03:38:17 CEST] <k_sze[work]> When I use av_interleaved_write_frame, is any of the errors recoverable? [03:42:02 CEST] <beauty> DHE: year [03:42:35 CEST] <beauty> I want to store the state of avformatcontext [03:43:20 CEST] <DHE> ... do you mean save the state of the encoder? [03:43:52 CEST] <beauty> year [03:43:55 CEST] <beauty> decoder [03:44:23 CEST] <DHE> year? [03:45:08 CEST] <beauty> I want to open ffmpeg,decode ,close ffmpeg. Then open again,av_seek to the last position, decode ,close ffmpeg. repeat [03:46:11 CEST] <DHE> that's not quite how it works. internal codec.... stuff... isn't just in the avcodeccontext. this strategy could work only if you make sure to bail on a keyframe or are ready to deal with exact seeking without keyframes. [03:46:50 CEST] <DHE> but using that method, saving the avcodeccontext isn't going to help. you're saving the frame number or exact timestamp, etc [03:48:00 CEST] <beauty> so,how to realise my requirement? [03:50:22 CEST] <beauty> I want use ffmpeg to decode 4000 streamed videos. [03:55:54 CEST] <beauty> how to use av_probe_input_format get avinputformat? [06:03:10 CEST] <Phantom_X5> hello, anyone interested in helping me? [06:03:39 CEST] <qmr> I'm interested in you not wasting our disk space with useless questions [06:03:50 CEST] <qmr> Do not ask to ask, just ask. If someone is willing and able to help, they will [06:04:13 CEST] <qmr> the other big thing is ask then WAIT. so many people join, ask, leave in 2 minutes when they don't get immediate help [06:06:08 CEST] <Phantom_X5> I have an issue with ffmpeg moving an output file to a different folder on the webspace. File uploads, moves to temp folder, ffmpeg converts successfully, file does not move to completed folder. Not sure if it is a permissions problem or something else [06:08:47 CEST] <qmr> that's all out of the context of ffmpeg as far as I know [06:08:55 CEST] <qmr> what web app are you using? [06:09:49 CEST] <qmr> I was often considered the smartest admin in the room at one of my old jobs. really I'm old and senile and confused most of the time, however, you can solve pretty much all problems with ........... [06:10:39 CEST] <qmr> logs, verbose output, google, documentation, strace/dtrace, tcpdump, source code [06:11:02 CEST] <qmr> in particular you should have logs of some kind for whatever web thing you are running [06:14:27 CEST] <Phantom_X5> I'm starting out with testing clipbucket [06:15:55 CEST] <Zeranoe> Phantom_X5: A bit of your server side code would go a long way. [06:17:41 CEST] <Phantom_X5> can you be more specific? [06:18:17 CEST] <Zeranoe> Phantom_X5: How are you calling FFmpeg [06:18:44 CEST] <Phantom_X5> not sure.. I'll see what I can find [06:22:57 CEST] <Phantom_X5> there is a ffmpeg.class.php file [06:27:52 CEST] <Phantom_X5> I will have to dig through this to find exactly what it is asking ffmpeg to do. It might be asking apache to deal with the finished file instead of ffmpeg. If I find the source, I'll come back [06:27:57 CEST] <Phantom_X5> thanks for the help tho [10:48:01 CEST] <i11ega1Code> hi i have a problem: https://pastebin.mozilla.org/8919776 [11:09:18 CEST] <i11ega1Code> how do i make mp4 playable for psp on ubuntu? i need something like oneline command that will convert the files of a directory and delete them when converted [11:20:05 CEST] <hero_biz> hi guys. [11:21:09 CEST] <JEEB> i11ega1Code: http://x264.fushizen.eu/random/psp_encoding.txt [11:21:34 CEST] <JEEB> (I should really update that to x264-params instead of x264opts [11:21:49 CEST] <JEEB> (that file hasn't been updated in like five years by now) [11:21:59 CEST] <hero_biz> guys, I'm trying to sharp a bad blurry source a little while encoding with ffmpeg. I'm using unsharp=3:3:1.5 for this reason. I wonder if there is any better sharpner, or better params for unsharp. [11:24:09 CEST] <hero_biz> what happens is I increase luma_msize_x and luma_msize_y? [11:30:00 CEST] <jubalh> Hi, I am thinking about converting all my avi + prop. codec into ogg theora format. Is there anything that speaks against this? I dont see a reason but it seems like most people dont do this, so I wonder if I am missing something. [11:31:13 CEST] <JEEB> a) do you have requirements of a colorspace that is not 4:2:0 YCbCr b) do you understand the limitations of theora (eats bits for breakfast) [11:32:10 CEST] <jubalh> b) no i dont. I read that it should be a major alternative to the prop codecs. I hoped it can help me to not have to install all the prop codecs and thus not add those inofficial repos [11:32:43 CEST] <jubalh> eating bits sounds like i will miss something regarding quality? [11:32:55 CEST] <JEEB> uhh, many distros have normal lavc in their normal repos? just maybe not installed by default [11:33:00 CEST] <JEEB> (like debian/ubuntu) [11:33:54 CEST] <hero_biz> here is a small snapshot from source: http://oi64.tinypic.com/2qby5pg.jpg which is a disaster quality. is there a better option than unsharp=3:3:1.5 for sharpening this video? [11:35:01 CEST] <jubalh> JEEB: I am not so familiar with these video things. i wonder if there is something to read which will make me better understand the options i have and the consequences it will have to change to another format [11:35:20 CEST] <jubalh> so far i totally rely on opinion on others via google. [11:42:27 CEST] <i11ega1Code> ehm... [11:42:30 CEST] <i11ega1Code> are you guys here [11:42:35 CEST] <i11ega1Code> why aren't you answering [11:43:03 CEST] <hero_biz> i11ega1Code: JEEB gave your answer. [11:43:24 CEST] <i11ega1Code> i don't see it, perhaps i have disconnected ? [11:43:37 CEST] <i11ega1Code> could you give me the answer he gave me hero_biz? [11:44:57 CEST] <i11ega1Code> hero_biz [11:45:46 CEST] <hero_biz> <JEEB> i11ega1Code: http://x264.fushizen.eu/random/psp_encoding.txt [11:45:55 CEST] <hero_biz> <JEEB> (I should really update that to x264-params instead of x264opts [11:46:05 CEST] <hero_biz> <JEEB> (that file hasn't been updated in like five years by now) [11:50:06 CEST] <i11ega1Code> ok how do i update? [11:50:45 CEST] <hero_biz> it says, use x264-params rather than x264opts in command line. [11:51:20 CEST] <i11ega1Code> ok is there something wrong with this command: [11:51:21 CEST] <i11ega1Code> for file in *.mp4; do outfile=CONVERTED/$file; ffmpeg -y -i "$file" -flags +bitexact -s 400x192 -r 29.97 -b:v 512k -c:a aac -b 672k -ab 96k -ar 24000 -f psp -strict -2 "$outfile"; done [11:51:51 CEST] <i11ega1Code> i tried that ^ though i need to use: [11:51:51 CEST] <i11ega1Code> for file in *.mp4; do outfile=CONVERTED/$file; ffmpeg -y -i "$file" -flags +bitexact -s 400x192 -r 29.97 -b:v 512k -acodec libfaac -b 672k -ab 96k -ar 24000 -f psp -strict -2 "$outfile"; done [11:52:52 CEST] <i11ega1Code> if anyone could help, it would be awesome, i need to go to a trip soon and i would love to watch some episodes on my psp [11:55:54 CEST] Action: hero_biz smartphones are much more great for watching episodes, I personally use galast s 5 with mx player :) [11:57:13 CEST] <jubalh> I am reading the ffmpeg manpage and ocumentation. in the command at http://ffmpeg.org/ffmpeg-all.html#Audible-AAX i dont see what '-c:a copy' does? [11:58:28 CEST] <i11ega1Code> for file in *.mp4; do outfile=CONVERTED/$file; ffmpeg -i "$file" -b 300 -s 320x240 -vcodec xvid -^C 32 -ar 24000 -acodec aac "$outfile"; done [11:58:28 CEST] <i11ega1Code> gives me : Unknown encoder 'xvid' [11:58:48 CEST] <iamtakingiteasy> hi, one wild question regarding mp4 container with aac audio payload in mdat boxes; should aac frames be raw or prefixed by ADTS-header? [11:59:38 CEST] <hero_biz> jubalh: means copy audio untouched. [12:00:07 CEST] <hero_biz> i11ega1Code: means your ffmpeg is not build with xvid support. [12:03:02 CEST] <i11ega1Code> i needed to change it to libxvid encoder [12:03:07 CEST] <i11ega1Code> :) [12:04:21 CEST] <jubalh> hero_biz: I see. Is there any pro in doing this to an aax file? i thought about converting to ogg vorbis. But maybe this will make me loose the chapter feature? [12:06:29 CEST] <fling> How to convert iso to flac? [12:06:31 CEST] <hero_biz> that option is not related to chapter (if you mean mkv chapter) as i know. [12:06:34 CEST] <i11ega1Code> YAY WORKED [12:06:34 CEST] <i11ega1Code> for file in *.mp4; do outfile=CONVERTED/$file; ffmpeg -i "$file" -b 300 -s 320x240 -vcodec libxvid -ab 32 -ar 24000 -acodec aac -strict -2 "$outfile"; don [12:07:14 CEST] <hero_biz> i'n not expert in encoding though. [12:07:21 CEST] <jubalh> okay [12:07:23 CEST] <jubalh> thanks [12:09:13 CEST] <oleg_mo> hello all, i got problem, i try to install this app http://pspvc.sourceforge.net/ from source, during instalation i got error [12:09:14 CEST] <oleg_mo> ERROR: XviD not found [12:09:14 CEST] <oleg_mo> If you think configure made a mistake, make sure you are using the latest [12:09:14 CEST] <oleg_mo> version from SVN. If the latest version fails, report the problem to the [12:09:14 CEST] <oleg_mo> ffmpeg-devel(a)mplayerhq.hu mailing list or IRC #ffmpeg on irc.freenode.net. [12:09:14 CEST] <oleg_mo> Include the log file "config.err" produced by configure as this will help [12:09:14 CEST] <oleg_mo> solving the problem. [12:09:15 CEST] <oleg_mo> -e \E[01;31mERROR during configure FFMPEG [12:12:15 CEST] <oleg_mo> sorry, i fix the problem https://help.ubuntu.com/community/PSP. [12:13:14 CEST] <nonex86> did i miss something? today the day of psp? [12:17:23 CEST] <oleg_mo> :) [12:17:43 CEST] <oleg_mo> i just have miss the lib to install [12:18:20 CEST] <oleg_mo> https://gist.github.com/xdamman/e4f713c8cd1a389a5917#file-install_ffmpeg_ub… [12:18:20 CEST] <oleg_mo> i reinstall with this sh all deps libs [12:18:20 CEST] <nonex86> check the config.log in ffmpeg folder [12:18:46 CEST] <nonex86> last records give you an idea why ffmpeg failed to build [12:19:02 CEST] <nonex86> usually, yes, its some dependencies [12:36:28 CEST] <fling> How to convert from bin/cue? [12:40:58 CEST] <fling> ok it is bchunk [12:45:06 CEST] <ozette> vcodex.com is an interesting website [12:45:23 CEST] <ozette> found it recently [12:46:37 CEST] <jubalh> do i see it right that mp4, m4a and m4b are all the same container format. the only difference is in how players treat them? but in theory i could just change filename and there is no conversion needed? [13:41:56 CEST] <acidflash> Hi [13:42:06 CEST] <acidflash> Can we discuss ffserver related issues here aswell ? [13:51:29 CEST] <acidflash> I was wondering if the ffm format is able to "receive" more than just audio and video streams, ie: a subtitle stream [13:52:40 CEST] <hseg> Hi. I have a ~2h long mkv file. I need to remove the part 01:02:00.5-01:02:02. Is there a way to do this without a massive reencoding job? [13:53:12 CEST] <hseg> (i.e. I need 00:00:00-01:02:00.5 followed by 01:02:02-the end) [13:53:19 CEST] <nonex86> hseg: what video codec? [13:53:52 CEST] <hseg> h264+vorbis [13:54:23 CEST] <nonex86> well, not sure about ffmpeg itself, but h264 can easily be cut from idr to idr [13:54:39 CEST] <hseg> Oh? [13:55:34 CEST] <nonex86> also usually its possible cut at frame basis with some reencoding 1st and last gop's [13:55:37 CEST] <hseg> (Reason this happened - I concatenated a bunch of episodes, and typo'd the end timestamp for one of them. It is really jarring) [13:55:50 CEST] <hseg> OK - how do I do it? [13:56:02 CEST] <nonex86> i guess ffmpeg can cut at idr basis (key frames) [13:56:40 CEST] <hseg> Not sure the keyframes will align with the gap I want. [13:57:18 CEST] <nonex86> -ss? [13:57:35 CEST] <hseg> You're thinking of the inverse problem I'm trying to solve. [13:57:49 CEST] <hseg> I need to remove a segment of the file, not extract it. [13:58:09 CEST] <nonex86> :) [13:58:18 CEST] <hseg> Yeah. [13:58:35 CEST] <hseg> It would suffice for me if I could blank out that segment. [13:58:49 CEST] <nonex86> actually this dosnt change the solution :) under the hood [13:58:58 CEST] <hseg> i.e. replace the frames there by black screen with null audio. [13:59:04 CEST] <hseg> What do you mean? [13:59:39 CEST] <nonex86> nevermind, you just asking a functinality i created in my previouse project :) [14:00:02 CEST] <nonex86> cut/merge h264 streams minimizing stream reencoding [14:00:25 CEST] <nonex86> so i am looking to your problem from developer perspective :) [14:01:20 CEST] <hseg> ... Does vorbis also support no-reencoding cuts? [14:01:40 CEST] <hseg> Because otherwise this is a no-go, as my mkv has h264+vorbis. [14:02:24 CEST] <nonex86> not sure about vorbis [14:02:25 CEST] <hseg> I *could* just reencode everything, it just would take ~4h [14:02:45 CEST] <nonex86> but even if it still have frame basis encoding [14:02:46 CEST] <hseg> BTW, any reason input streams don't support -to? [14:02:59 CEST] <hseg> Hm. [14:03:15 CEST] <nonex86> you can still cut from the first frame [14:03:25 CEST] <nonex86> your audio stream can start earlier [14:03:28 CEST] <nonex86> or later [14:03:32 CEST] <nonex86> compared to video [14:03:43 CEST] <nonex86> usually that gap is really small [14:03:49 CEST] <nonex86> but in your case [14:04:02 CEST] <nonex86> when you want delete from the middle of the file [14:04:09 CEST] <nonex86> and then merge two parts [14:04:14 CEST] <nonex86> i guess you need to reencode [14:04:26 CEST] <nonex86> well, audio encoding is quite fast [14:04:37 CEST] <nonex86> even if you need to resample stream [14:05:31 CEST] <nonex86> even if you have several audio streams :) [14:06:19 CEST] <nonex86> btw, by saying 'frame' here, i mean audio frame [14:06:21 CEST] <hseg> Right, I noted that e.g. when youtube-dl uses ffmpeg to reencode audio or to merge multiple streams into one container, it's much faster than my current case of concating videos. [14:07:36 CEST] <nonex86> in your case the good thing is you cut/merge the same video stream [14:08:35 CEST] <nonex86> again, if you can tolerate idr/recovery point (key frame) accuracy [14:08:46 CEST] <nonex86> you can just merge h264 NALU's [14:09:04 CEST] <nonex86> if you need frame accuracy... [14:09:18 CEST] <nonex86> that depends on video stream itself [14:09:25 CEST] <hseg> OK, so I'm running ffprobe to get the keyframe positions. [14:10:05 CEST] <hseg> So I guess my suggestion of blanking out that segment isn't useful? [14:10:22 CEST] <nonex86> depends on you [14:10:30 CEST] <nonex86> can you tolerate holes inside the stream? :) [14:10:43 CEST] <nonex86> or you need to real cut/merge? [14:10:56 CEST] <nonex86> cut 1st part using ffmpeg [14:11:03 CEST] <nonex86> cut 2nd part using ffmpeg [14:11:08 CEST] <nonex86> merge them using ffmpeg [14:11:30 CEST] <nonex86> or maybe ffmpeg can do this thing in atomic operation [14:11:32 CEST] <nonex86> dont know [14:11:35 CEST] <hseg> I can tolerate image being black and sound being null at that point - it occurs right after a blackout. [14:11:53 CEST] <nonex86> better ask cli adepts here :) [14:11:56 CEST] <hseg> The only question is whether it will incur a long rendering time. [14:14:02 CEST] <nonex86> https://trac.ffmpeg.org/wiki/Concatenate [14:16:51 CEST] <hseg> I could cut and merge. However, doesn't that take a long time? [14:17:07 CEST] <nonex86> depends [14:17:19 CEST] <nonex86> dont forget about "copy" parameter [14:18:31 CEST] <hseg> I've been using the concat filter... That may have caused my bad experience. [14:19:29 CEST] <nonex86> :) [14:22:42 CEST] <hseg> Yeah... So you're suggesting using -ss/-t to pick the parts before/after the split, passing -c copy to avoid reencoding, then concating via the concat protocol with -c copy? [15:01:35 CEST] CTCP AVStream: from beauty (beauty!~user@2001:cc0:2026:4002:1423:827f:2a58:320a) to #ffmpeg [15:01:35 CEST] <beauty> AVFormatContextAVCodecContext, which costs memory largely? [15:06:39 CEST] <DHE> those data structures are not that big in the grand scheme of things. an active codec (decoder or encoder) may be huge though because the codec itself will store reference frames, etc. [15:08:14 CEST] <beauty> oh. [15:30:53 CEST] <fling> How do I add a covertart picture to ogg or mka properly? Should I set AV_DISPOSITION_ATTACHED_PIC somehow? [15:42:43 CEST] Action: fling found vorbistools [15:45:48 CEST] <Kirito> Many TV's now support interpolating (that's the correct term, right?) videos on the fly, making them appear as if they were recorded at higher frame rates by comparing two frames and generating a new one between them. This process is apparently inexpensive enough that it can be done in real-time, but it makes me wonder this: [15:45:50 CEST] <Kirito> Could you gain much of a benefit by pre-encoding source videos with interpolation? For example, could you improve the quality of the interpolation much by comparing more than just two frames? Looking further ahead and farther behind when generating the middle frame [15:46:42 CEST] <Kirito> and if so, are there any practical applications of this, specifically in ffmpeg filters (i.e. not proprietary) [15:49:56 CEST] <DHE> TVs have DSP chips in them designed to do just that. it is fairly expensive in software, but you can build a chip to do it far quicker using custom circuitry [15:50:13 CEST] <Kirito> Ahhh, interesting, that explains a lot [15:50:27 CEST] <Kirito> As I was really surprised that TV's could do it so easily [15:53:07 CEST] <Kirito> With that in mind still, could you still produce much better results in software? Or are DSP chips pretty much good enough that it wouldn't be worth the effort/extra storage [15:57:44 CEST] <retard> this is the first thing to turn off [15:57:51 CEST] <retard> when you get a new tv [15:58:24 CEST] <Kirito> (Personal preference, I actually love it) [15:59:36 CEST] <DHE> when playing video games you turn it off. no exceptions. giving the DSP frame look-ahead opportunities costs latency. [16:00:04 CEST] <Kirito> Oh, that's fair definitely, I mean for regular TV though. [16:46:21 CEST] <SpeakerToMeat> Good day. [16:47:46 CEST] <SpeakerToMeat> Question, I'm mixing a video file and an audio file in a single mux (.mov) with ffmpeg using copy for both codecs (no recode), but ffmpeg and ffprobe are showing the wrong fps for the origin video file (showing 25 fps when I know for a fact the file is 24 fps), is there any way to change the fps (via constant frame count, that is reinterpeting) while muxing them? [16:48:51 CEST] <SpeakerToMeat> I don't want a recode or processing of the frames, I just want ffmpeg to say "oh this is really 24fps not 25". [16:50:31 CEST] <SpeakerToMeat> The input video file is DNxHD raw (extracted from an MXF with bmx) [16:51:15 CEST] <nonex86> why are you care about fps? [16:51:20 CEST] <nonex86> as long as pts is correct [16:51:29 CEST] <nonex86> guess nothing to worrie about [16:51:42 CEST] <nonex86> isnt it? [16:53:18 CEST] <hidsbv> while joining two videos, I receive message- "Requested output format'concat' is not a suitable output format". whats wrong and how to correct it ? [16:53:46 CEST] <c_14> concat isn't an output format [16:53:49 CEST] <c_14> It's an input format [16:54:31 CEST] <hidsbv> command - ffmpeg.exe -f concat -i lst.txt -c copy out.mp4 [16:55:05 CEST] <SpeakerToMeat> nonex86: pts? [16:55:32 CEST] <nonex86> SpeakerToMeat: presentation timestamp [16:55:50 CEST] <hidsbv> c_14: ? [16:58:00 CEST] <hidsbv> while joining two videos, I receive message- "Requested output format'concat' is not a suitable output format". whats wrong and how to correct it ? [16:58:02 CEST] <hidsbv> command - ffmpeg.exe -f concat -i lst.txt -c copy out.mp4 [16:59:18 CEST] <SpeakerToMeat> nonex86: Because I like my audio and video to be in sync [17:03:51 CEST] <nonex86> SpeakerToMeat: your audio and video will be in sync if pts correct for both streams [17:04:03 CEST] <nonex86> SpeakerToMeat: its not related to fps value you see in header :) [17:04:11 CEST] <DHE> Kinda off-topic, I need to do some bitstream hacking to H264. Is there documentation or an ffmpeg module that can do SEI splitting and reassembly? [17:04:35 CEST] <nonex86> SpeakerToMeat: ill ask again - why are you care about fps value in header? you have any problems with result file? [17:14:00 CEST] <CorvusCorax> Hi. I want to extract individual frames from a video (as images) - but only the frames that are actually stored. For example, video has a timebase of 60 fps but only stores 20 fps, with possible gaps. I only want the frames that are actually there, if possible with the correct pts from the filename. is that possible? [17:14:54 CEST] <CorvusCorax> I tried just recoding using the image2 codec, but that creates lots of extra images for all the "missing" frames, too [17:22:53 CEST] <CorvusCorax> (which are all identical) [17:35:00 CEST] <SpeakerToMeat> nonex86: There's no header, it's raw [17:35:44 CEST] <SpeakerToMeat> nonex86: Yes, result file a) says it's 25 fps, we could load it on premiere and reinterpret it as 24 fps in a 24fps project there if it comes to that b) audio and video desync [17:38:48 CEST] <nonex86> you have two separate streams [17:38:52 CEST] <nonex86> audio and video [17:39:01 CEST] <nonex86> and try to mux them in one file, correct? [17:39:16 CEST] <SpeakerToMeat> correct [17:39:48 CEST] <nonex86> i doubt your desync problem related to "wrong" fps you see in the output file [17:40:05 CEST] <SpeakerToMeat> then what is it related to? [17:40:17 CEST] <nonex86> start time in both streams are the same? [17:40:25 CEST] <nonex86> or you provided time offsets somehow? [17:41:01 CEST] <SpeakerToMeat> start time is the same, both files where synchronized inside the mxf when the video "knew" it was 24fps [17:41:09 CEST] <SpeakerToMeat> and have the same count [17:41:43 CEST] <SpeakerToMeat> DOn't worry I'll find this on my own, or fidn a tool that can do it. thanks anyhow [17:41:53 CEST] <nonex86> ok :) [17:44:49 CEST] <SpeakerToMeat> aaand the pts is wrong. it's at 25 fps [17:51:36 CEST] <SpeakerToMeat> -r 24 on the input gives me a right pts, but it's not stored in the output file. [17:54:06 CEST] <SpeakerToMeat> I'll just convert to discrete dpx frames and use that. [18:03:10 CEST] <crunchybit> need to clean up some audio of a speech from 1968 of some hippy screaming in a megaphone. Sounds really compressed and distorted. Anything that ffmpeg and do? [18:03:38 CEST] <crunchybit> if not any recommendations? [18:20:41 CEST] <Amitari2> Anyone who can help me extract frames from a RAW AVI? (BGR) [18:21:37 CEST] <Amitari2> This is what happens when I try. [18:21:38 CEST] <Amitari2> http://pastie.org/10945514 [19:45:26 CEST] <CorvusCorax> @Amitari: No decoder for stream #0:0, filtering impossible [19:46:57 CEST] <CorvusCorax> that's odd, i had no issues with raw bgr24 data... [19:47:04 CEST] <Amitari> CorvusCorax: Ouch. [19:47:57 CEST] <CorvusCorax> and earlier: [avi @ 0x55719b365f20] Could not find codec parameters for stream 0 (Video: none (BGR[24] / 0x18524742), none, 2160x2160): unknown codec [19:48:18 CEST] <JEEB> it just doesn't know what that format is supposed to be [19:48:24 CEST] <CorvusCorax> maybe if you force the codec to raw, but I don't remember the syntax to enforce the decoding codec [19:48:34 CEST] <JEEB> before -i you set -c:v rawvideo [19:48:45 CEST] <JEEB> and then -pix_fmt BGR (also before -i) [19:48:55 CEST] <JEEB> before -i = decoding parameters [19:49:00 CEST] <JEEB> after -i = encoding parameters [19:49:01 CEST] <CorvusCorax> thanks :) [19:49:56 CEST] <JEEB> also you could just add that fourcc to work as raw video, BGR [20:01:38 CEST] <Amitari> JEEB: I'll try! [20:03:11 CEST] <Amitari> Nope, didn't work. :( [20:03:15 CEST] <Amitari> http://pastie.org/10945577 [20:04:32 CEST] <JEEB> ouch [20:06:01 CEST] <DHE> Does anyone know where to find code (or documentation, I can write my own if needed) on how to break an H264 stream down into its SEI components? I need to do a little bit of bitstream hacking [20:07:25 CEST] <CorvusCorax> Amitari, do ffmpeg -pix_fmts [20:07:54 CEST] <CorvusCorax> i think the one you need is "bgr24" lower case [20:08:00 CEST] <Amitari> Should that go before the -i, and what should it replace? [20:08:06 CEST] <CorvusCorax> but that depends on how your raw video is encoded [20:08:34 CEST] <CorvusCorax> its a separate call, with no other arguments, it just prints outr all supported pixel formats of your ffmpeg version :) [20:08:46 CEST] <Amitari> Shit, something happened! [20:09:27 CEST] <JEEB> well the error looks like pix_fmt isn't something that works with decoding (?) although that is still weird since you should be able to set pix_fmt with at least raw video input [20:09:39 CEST] <Amitari> http://pastie.org/10945581 [20:09:58 CEST] <JEEB> yeah, it just parsed the pix_fmts option [20:10:04 CEST] <JEEB> and printed you the available pix_fmts [20:10:17 CEST] <CorvusCorax> thats what it was supposed to. those are the valid formats to be given with -pix_fmt [20:10:22 CEST] <JEEB> yes [20:10:31 CEST] <JEEB> he just gave it more parameters so he probably expected more than that from it [20:10:33 CEST] <CorvusCorax> you need bgr24 I think [20:10:51 CEST] <Amitari> So, how should the command go then? [20:11:02 CEST] <Amitari> I sorta tried that myself, but I dunno what commands from before should be removed and whatnot. [20:11:11 CEST] <JEEB> -pix_fmt bgr24 before -i [20:11:13 CEST] <CorvusCorax> ffmpeg -c:v rawvideo -pix_fmt bgr24 -i video.avi /out/%08d.png [20:11:43 CEST] <Amitari> Didn't work. :( [20:11:52 CEST] <Amitari> http://pastie.org/10945583 [20:12:33 CEST] <JEEB> yeah, so it doesn't work for input weird enough :/ [20:12:57 CEST] <CorvusCorax> odd. but ffmpeg treats bgr24 as default for rawvideo avis anyway, just skip it [20:14:33 CEST] <Amitari> You want me to try that? [20:15:28 CEST] <CorvusCorax> ffmpeg -c:v rawvideo -i video.avi /out/%08d.png [20:17:03 CEST] <Amitari> Well, something happened, but the conversion failed. :( http://pastie.org/10945586 [20:17:30 CEST] <Amitari> Wait! [20:17:30 CEST] <Amitari> No! [20:17:36 CEST] <Amitari> Now it works, I just had to change the filepath a litt. [20:17:39 CEST] <Amitari> Thanks! :D [20:19:33 CEST] <CorvusCorax> are the images good? or are they filled with pixel-puke? [20:19:40 CEST] <BinaryBench> Hey, I'm on windows and do not seem to have a "screen-capture-device" as mentioned in https://trac.ffmpeg.org/wiki/Capture/Desktop [20:19:48 CEST] <CorvusCorax> if the pix_fmt is wrong you can end up with weird results [20:19:57 CEST] <Amitari> Nope, they look good! :D [20:19:59 CEST] <furq> BinaryBench: https://github.com/rdp/screen-capture-recorder-to-video-windows-free [20:20:18 CEST] <furq> that's the device referred to on the wiki, there are probably others you can use [20:20:27 CEST] <furq> or use -f gdigrab which is slower but doesn't need anything external [20:21:09 CEST] <JEEB> Amitari: that just happens to happen because the input pix_fmt is being defaulted to bgr24, so if it wasn't that it would probably be more problematic [20:21:12 CEST] <BinaryBench> furq: speed is all I care about, however the phrase "java" gives me concerns about that.... [20:21:36 CEST] <furq> i've never used it but dshow should be much faster than gdi [20:21:39 CEST] <JEEB> that said, there's a lot of semi-standardized fourccs for various raw things [20:22:01 CEST] <BinaryBench> furq: I suppose I'll give it a go... [20:22:09 CEST] <JEEB> (libavformat/riff.c has a whole long list of them) [20:22:13 CEST] <CorvusCorax> While we are at converting videos to images, does anyone know a way to extract only existing frames from a video? Assume a camera stores an avi and claims 60fps but there's only 10 fps in it, if i do ffmpeg -i input.avi out%04d.png, I get lots of duplicate images where the missing frames get automatically duplicated [20:22:16 CEST] <furq> it's probably worth trying gdigrab first [20:22:19 CEST] <furq> it might be fast enough [20:22:57 CEST] <CorvusCorax> ffmpegs -r option to change framerate is only an option if the actually achieved rate is constant, but its not [20:26:57 CEST] <BinaryBench> furq: with gdigrab, what would be the command to record? [20:31:59 CEST] <bryan_> Hello everybody [20:33:48 CEST] <bryan_> i am using ffmpeg interface API in my shared library...but when i run the third party application which uses my API...says undefined symbols for ffmpeg API...how can i resolve this issue? [20:34:56 CEST] <JEEB> sounds like the library isn't in your search path? [20:35:38 CEST] <bryan_> i have exported the ffmpeg library path in LD_LIBRARY_PATH on console but still it is not able to resolve it [20:36:23 CEST] <JEEB> is it the same you linked against? [20:36:27 CEST] <CorvusCorax> are you linking in your library dynamically or statically into the 3rd party app? [20:36:28 CEST] <bryan_> i think one way to do is "dlopen" but does it really required? [20:36:40 CEST] <JEEB> no, dlopen is not required [20:36:40 CEST] <bryan_> dynamically [20:37:09 CEST] <CorvusCorax> nah, dlopen you only need to load libraries fully dynamicaly, for exmaple to load plugins which aren't known at compiletime yet [20:37:22 CEST] <JEEB> it just sounds like the version of the libraries you linked against aren't available [20:37:55 CEST] <JEEB> I've been doing shared builds in nonstandard locations for quite a while and WorksForMe is my experience so far [20:38:19 CEST] <JEEB> so either it's something really dumbly simple or something very specific to what you're trying to do in some specific way [20:38:20 CEST] <bryan_> ok [20:39:32 CEST] <CorvusCorax> the systems dynmic linker should load the correct ffmpeg libs when your api library is being loaded. as long as they are installed [20:40:29 CEST] <bryan_> hmm..will check it..might be doing something wrong in my setup.. [20:40:44 CEST] <bryan_> thank you [20:54:39 CEST] <bencc> I'm doing several concurrent live transcoding tasks. what cpu metric tells me I doing to much and might drop frames? [20:56:37 CEST] <BtbN> none, the stream fps do. [20:57:21 CEST] <bencc> BtbN: ok [20:58:14 CEST] <BtbN> you can probably guess that when all cores are at 100%, you are likely also dropping frames. But it might be just allright then as well. Or it might be dropping earlier, because it's not perfectly distributed accorss all cores. [21:53:03 CEST] <DHE> bencc: that's something you should measure ahead of time and test run. BtbN pretty much nails it - only an actual measurement from ffmpeg will provide the information. [21:53:57 CEST] <BtbN> Can probably also take a look at the outgoing bandwidth [21:54:10 CEST] <BtbN> if you are doing strict CBR, you are able to calculate the expected bandwidth quite accurately [21:54:15 CEST] <DHE> I find that at 95% usage you might run into a CPU cap. operating systems don't migrate threads real-time so you can end up with 0 active jobs on one core and 2 on another. you're effectively bottlenecked [22:16:44 CEST] <teratorn> is anyone very familiar with RTSP over UDP? how does the stream / player / receiver recover from udp packet loss, and how does that translate in to loss of video and audio frames? is there any retransmit logic in the protocols? we're transcoding from RTSP to HLS and the HLS files are getting corrupted due to lost packets in the incoming RTSP/UDP stream... [22:20:28 CEST] <BtbN> It will be fixed on the next I frame. [22:22:50 CEST] <DHE> I'd suggest checking the packet spacing on either the receiver or (preferably) the sender side. make sure the packets are spaced and not in sudden bursts [22:23:38 CEST] <DHE> I've had a lot of problems with that all across the network. [22:26:28 CEST] <teratorn> DHE: well I don't think I have any control over the sender, it's a little boxy encoder thing [22:26:43 CEST] <teratorn> BtbN: could you elaborate? [22:27:09 CEST] <BtbN> that's how encoding works. If a frame is missed/broken, the stream is broken until the next I frame comes by [22:33:22 CEST] <DHE> key frames (I frames) are full frames, other frames are based on the differences to other frames [22:41:10 CEST] <SchrodingersScat> is there a format that's better than others for getting clean cuts? like not having it go to a keyframe or etc? [22:41:38 CEST] <DHE> well, if you're willing to crank the bitrate you could make every frame a keyframe... [22:41:46 CEST] <SchrodingersScat> tell me more [22:42:11 CEST] <DHE> with ffmpeg using -g 1 uses nothing but keyframes [22:42:17 CEST] <SchrodingersScat> is very short, so size isn't really an issue [22:43:11 CEST] <teratorn> BtbN: OK I guess that makes sense [22:44:11 CEST] <SchrodingersScat> ahahahaha, I may have spoken too early, this is gonna be huge, but should be fine. DHE thanks for the tip. [22:44:30 CEST] <DHE> SchrodingersScat: you were warned. :) [22:44:49 CEST] <SchrodingersScat> quite [22:45:43 CEST] <DHE> SchrodingersScat: might compromise on a small number, like -g 5 depending how accurate you need [22:46:01 CEST] <furq> you could just use an intra-only codec [22:46:13 CEST] <DHE> like mjpeg? [22:46:16 CEST] <furq> that's probably better if you're going to be cutting it up later [22:46:23 CEST] <furq> i was thinking more like ffv1 but sure [22:46:28 CEST] <DHE> after teratorn's issue I had keyframes on the brain... [22:46:52 CEST] <DHE> ffv1 is all keyframes? good to know... [22:47:01 CEST] <furq> it is by default [22:47:09 CEST] <furq> you can increase the gop size but i don't know how that actually works [22:47:22 CEST] <furq> it doesn't seem to reduce the bitrate very much [22:49:42 CEST] <SchrodingersScat> k, think that took care of it [22:56:27 CEST] <teratorn> BtbN: so what is the trick to not corrupting the HLS stream that I'm transcoding to? I guess you MUST NOT encoding any frames until you get another valid iframe again? [22:56:58 CEST] <BtbN> Don't lose any packets I guess [22:56:58 CEST] <DHE> if you're getting packet loss, it means a network problem. your hardware should be able to tell you where it's happening [23:03:56 CEST] <teratorn> U D P [23:04:04 CEST] <teratorn> rtsp supports streaming over U D P [23:04:13 CEST] <teratorn> you will get packet loss [23:04:26 CEST] <teratorn> I'm not sure why I *have* to use UDP instead of TCP, but there yo ugo [23:04:43 CEST] <teratorn> I think the packet loss (wifi) was causing the tcp streams to hang, so we switched to udp as a quick fix (i'm sorry) [23:04:49 CEST] <teratorn> for making the internet worse [23:04:52 CEST] <BtbN> could try to use a container with solid error correction. But I don't think ffmpeg supports muxing that. [23:05:08 CEST] <BtbN> UDP is the better choice for streams usually [23:05:17 CEST] <teratorn> i don't have any choice, it's either rtsp over tcp or rtsp over udp from this boxy encoder gadget [23:05:19 CEST] <BtbN> With a short gop interval, you loose a few seconds at max [23:05:41 CEST] <teratorn> i'm not sure how configurable the boxy thing is, or even if we are allowed to configure it. i doubt it. [23:06:19 CEST] <DHE> ... wifi? [23:06:29 CEST] <teratorn> it's a matter of disregarding all the necessary data from the corruption caused to the rtsp stream due to packet loss, and encoding a valid hls stream just with placeholder frames [23:07:22 CEST] <teratorn> DHE: yes, hanging tcp rtsp streams due to wifi packet loss. now having switched to udp, our transcoding is borked because we're not actually handling hte packet loss correctly for some reasons. [23:07:23 CEST] <DHE> it's not quite that easy. from the first corrupted frame, some kind of image smear will persist until the next keyframe [23:07:41 CEST] <teratorn> nothing I can do about that rtp doesn't have any retransmit logic built in [23:07:54 CEST] <teratorn> i don't control the sender. black box rtsp stream over tcp or udp... [23:08:06 CEST] <teratorn> so you either use tcp [23:08:11 CEST] <DHE> can you make it more reliable? plug something into the same switch as this encoder thing? [23:08:13 CEST] <TD-Linux> rtcp supports retransmissions. you should be able to negotiate that with rtsp [23:08:15 CEST] <teratorn> or you use udp and recover from packet loss *somehow* [23:08:18 CEST] <furq> is the sender using wifi [23:08:38 CEST] <teratorn> TD-Linux: thank you I will have to look in to rtcp to see if that is supported [23:09:12 CEST] <teratorn> and how we can use it to recover lost frames... but keep in mind this is live streaming, so the device would have to maintain a buffer to retransmit frames from [23:09:14 CEST] <BtbN> it's not trivial, if possible at all, to detect that kind of corruption [23:09:19 CEST] <teratorn> s/frames/packets/ [23:09:49 CEST] <TD-Linux> BtbN, yes it is, RTP has sequence numbers [23:09:59 CEST] <teratorn> yes, the sequence numbers, but! [23:09:59 CEST] <DHE> I'm suggesting you prevent the loss rather than recover from it. why are you using wifi here? [23:10:04 CEST] <teratorn> udp may be delivered out of order [23:10:07 CEST] <BtbN> That's if a whole frame gets lost? [23:10:11 CEST] <TD-Linux> teratorn, there are a lot of ways to do it, and yeah you need to keep them around for retransmission [23:10:14 CEST] <teratorn> DHE: no choice [23:10:26 CEST] <TD-Linux> there are other things you can do with RTCP, like request an Iframe on packet loss rather than a retransmission [23:11:00 CEST] <teratorn> DHE: I can't control all the variables. I just have to make a solution that lets us "recover" from udp packet loss of the rtsp stream, either by inducing retransmission via an rtcp control channel, or by disregarding data and only doing the transcode with valid data [23:11:15 CEST] <teratorn> TD-Linux: OK i definiately have homework to do, thanks [23:11:40 CEST] <teratorn> TD-Linux: its a little boxy encoder thing, I'm not sure how much control I have [23:11:47 CEST] <TD-Linux> there is even really fancy stuff like request a frame predicted only from frames you have, though your options might be limited as you don't have much control over one side [23:12:12 CEST] <teratorn> basically, if I can do it with well supported rtsp/rtcp features, I'm ok [23:13:02 CEST] <teratorn> or, again, just ignore the frames that are corrupted or have references to missing frames, and once enough time has elapsed, I can go ahead and pretend those packets will never come in, mark them as lost, and continuing the transcode with placeholder frames [23:16:09 CEST] <TD-Linux> teratorn, yup, that will work too (waiting for next keyframe). if it helps to google, "AVPF" is what you need in the rtsp sdp [23:16:18 CEST] <TD-Linux> (for rtcp) [23:20:40 CEST] <teratorn> TD-Linux: thank you [23:40:44 CEST] <SchrodingersScat> yay, i figured it out, thanks guys [23:41:40 CEST] <DHE> Does anyone know where to find code (or documentation, I can write my own if needed) on how to break an H264 stream down into its SEI components? I need to do a little bit of bitstream hacking. [23:51:55 CEST] <emilsp> what am I doing wrong if when I parse my avfilter (I need to crop a frame), avfilter_graph_parse2() returns a nonzero code and spouts in stderr [AVFilterGraph @ 0x7ff801e2c320] No such filter: 'buffer' [23:52:47 CEST] <emilsp> http://sprunge.us/ZJBb?c on line 48 [23:53:56 CEST] <DHE> you don't need to provide buffer and buffersink in the pipeline. you create them yourself as endpoints to the pipeline and attach them manually [23:53:58 CEST] <jpsharp> I'm feeding ffmpeg with -i file.gif with an output to an mp4 file. I have it set for infinite loop. That all works, but I'd like to be able to just change the gif while the process is running and have the resultant mp4 file change. ffmpeg seems to read the file once and then use that over and over, how can I tell ffmpeg to reload the file either on command or occasionally. [23:54:35 CEST] <emilsp> DHE, so the 'arg' string is wrong ? [23:54:59 CEST] <DHE> emilsp: there's some examples in the doc directory, including an avfilter [23:55:09 CEST] <emilsp> woohoo, thanks :) [23:55:26 CEST] <DHE> doc/examples/filter*.c [23:56:30 CEST] <emilsp> yup, am reading them already, thank you very much [23:58:47 CEST] <t4nk834> hi all [00:00:00 CEST] --- Wed Oct 19 2016

1 0

ffmpeg-devel.log.20161017
by burek 18 Oct '16

18 Oct '16

[00:59:05 CEST] <philipl> BtbN: remind me how you force the framerate so that deinterlaced transcoding works [00:59:18 CEST] <BtbN> -r as input option [00:59:38 CEST] <BtbN> the deinterlaced framerate, that is. [00:59:43 CEST] <BtbN> For 25i, you'd pass 50 [01:00:13 CEST] <philipl> cool. that works [01:01:42 CEST] <philipl> So, I poked at the avi seeking thing more. The best solution I've got so far is (specifically for mpv) to turn on genpts. [04:40:41 CEST] <cone-222> ffmpeg 03Shivraj Patil 07release/3.1:d89979e86b32: avutil/mips/generic_macros_msa: rename macro variable which causes segfault for mips r6 [04:40:41 CEST] <cone-222> ffmpeg 03Shivraj Patil 07release/3.1:d2566b124af8: Support for MIPS cpu P6600 [04:40:41 CEST] <cone-222> ffmpeg 03Hendrik Leppkes 07release/3.1:263add4462a4: ffmpeg: remove unused and errorneous AVFrame timestamp check [04:40:41 CEST] <cone-222> ffmpeg 03Moritz Barsnick 07release/3.1:fc36e692c4ac: tools: fix grammar error [04:40:41 CEST] <cone-222> ffmpeg 03Moritz Barsnick 07release/3.1:30c80e81d2a5: lavc: fix typos [04:40:41 CEST] <cone-222> ffmpeg 03Moritz Barsnick 07release/3.1:f12c0da09b04: lavfi: fix typos [04:40:41 CEST] <cone-222> ffmpeg 03Moritz Barsnick 07release/3.1:7fefd776682a: cmdutils: fix typos [04:40:42 CEST] <cone-222> ffmpeg 03Michael Niedermayer 07release/3.1:08eef74a39d7: avformat/utils: Update codec_id before using it in the parser init [04:40:43 CEST] <cone-222> ffmpeg 03Moritz Barsnick 07release/3.1:6109c10b81d6: doc: fix various typos and grammar errors [04:48:18 CEST] <cone-222> ffmpeg 03Michael Niedermayer 07release/3.1:675258764dce: Update for 3.1.5 [05:02:58 CEST] <cone-222> ffmpeg 03Shivraj Patil 07release/3.0:ef5b120e2dd6: avutil/mips/generic_macros_msa: rename macro variable which causes segfault for mips r6 [05:02:59 CEST] <cone-222> ffmpeg 03Shivraj Patil 07release/3.0:c993a11e56ab: Support for MIPS cpu P6600 [05:03:00 CEST] <cone-222> ffmpeg 03Hendrik Leppkes 07release/3.0:27f1fb125143: ffmpeg: remove unused and errorneous AVFrame timestamp check [05:03:01 CEST] <cone-222> ffmpeg 03Moritz Barsnick 07release/3.0:6457346e6a15: tools: fix grammar error [05:03:02 CEST] <cone-222> ffmpeg 03Moritz Barsnick 07release/3.0:17b8e7799f64: lavc: fix typos [05:03:03 CEST] <cone-222> ffmpeg 03Moritz Barsnick 07release/3.0:8baf2d8fadc1: lavfi: fix typos [05:03:04 CEST] <cone-222> ffmpeg 03Moritz Barsnick 07release/3.0:88f52f2f8f21: cmdutils: fix typos [05:03:05 CEST] <cone-222> ffmpeg 03Michael Niedermayer 07release/3.0:047e0049d1cc: Changelog: update for recent commits [07:43:53 CEST] <atana> michaelni: hi [07:45:15 CEST] <atana> I am not sure how to use fft function of ffmpeg. I tried on a test code https://dpaste.de/es8K. could you point me to an example or something [12:48:35 CEST] <michaelni> atana, example code for all the ffts should be in libavcodec/tests/fft.c, (ignore all code in it that doesnt apply to avfft) [12:50:41 CEST] <cone-192> ffmpeg 03Hendrik Leppkes 07master:656feb641de3: mov: move stsd finalization to an appropriate place [13:23:37 CEST] <atana> michaelni: I have made changes to `libavfilter/avfilter.c` and libavfilter/avfilter.h in the ffmpeg repo cloned from github. after configuration I installed it using make install. Now, I want to test the filter by running a driver program. How do I compile the program? I ran `gcc -g pp.c -o pfilter -Iinclude -Llib` but it says undefine reference to function peakPointsFilter however I have included it in libavfilter/avfilter.h file [13:23:47 CEST] <atana> am I missing some flag? [13:32:03 CEST] <michaelni> atana, the easiest way to test a filter is with ffplay or ffmpeg command line programs [13:32:36 CEST] <michaelni> but either way maybe you are missing a line for the new filter in libavfilter/Makefile [13:33:24 CEST] <michaelni> ./ffplay somerandomvideo.mp4 -vf myfilter=myfiltersarguments [13:34:09 CEST] <atana> I haven't done any changes in libavfilter/Makefile. I will check [13:35:02 CEST] <michaelni> better ffplay example: (now with audio inszead of video filter) ./ffplay random.mp3 -af myfilter=myfiltersarguments [13:36:55 CEST] <atana> are myfiltersarguments separated by comma? [13:39:19 CEST] <ubitux> atana: myfilter=arg0:arg1:arg23=val23:arg57=val57 [14:22:58 CEST] <atana> michaelni: I could't figure out what to include in Makefile. This is the git diff of what I have done https://dpaste.de/fReB . function `peakPointsFilter` is included in avfilter.h and is defined in avfilter.c but still on compilation `gcc -g pp.c -o pfilter -Iinclude -Llib` it says undefine reference to function peakPointsFilter [15:00:38 CEST] <cone-192> ffmpeg 03Aman Gupta 07master:9ea91e4114e8: lavc/videotoolboxenc: implement a53cc [15:00:39 CEST] <cone-192> ffmpeg 03Rick Kern 07master:aa413b810a7f: lavc/videotoolboxenc: flush/free frames on close [15:00:40 CEST] <cone-192> ffmpeg 03Rick Kern 07master:9875695e2cee: lavc/videotoolboxenc: Update a53cc handling [15:00:41 CEST] <cone-192> ffmpeg 03Rick Kern 07master:d3874b74f35b: lavc/videotoolboxenc: Error log formatting. [15:03:36 CEST] <barb_> Hello, I'd like to apply for outreachy but the deadline is today and I won't have the time to finish my qualification task. Does anybody know if maybe I could start contributing today(and finish something consistent by the end of the week), could I still apply and mention that in my application? I'm asking because since the decisions are only final in November 08, so there would be still time to evaluate me. [15:06:02 CEST] <Compn> barb_ : sure, but we dont have qual tasks set up [15:06:10 CEST] <Compn> i dont think [15:06:47 CEST] <Compn> but yeah it cant hurt to submit an application [15:07:06 CEST] <Compn> we dont mind if you do the qual task after the application [15:07:13 CEST] <Compn> to answer your question :) [15:09:52 CEST] <barb_> Oh, great, so.. I talked to Paul Mahol about the XPM decoder project and he mentioned something about extending bitdepth support to some filter [15:10:19 CEST] <barb_> but then mentioned that I could pick another project myself, like creating a QR generator [15:10:45 CEST] <barb_> do you have any other ideas for tasks? [15:11:10 CEST] <barb_> maybe more appropriate to the QR generator project [15:13:29 CEST] <barb_> Or maybe you guys have any particular filter(s) in min for this addition [15:13:55 CEST] <barb_> *in mind [15:55:02 CEST] <cone-192> ffmpeg 03Andreas Cadhalpun 07master:eb751f06db9f: matroskadec: fix NULL pointer dereference [16:31:42 CEST] <Compn> barb_ : ah, no idea :) [16:31:55 CEST] <Compn> qr generator you say? [16:31:58 CEST] <Compn> hmmm [16:32:06 CEST] <Compn> never used a qr code myself :) [16:32:26 CEST] <Compn> i am upset they dont just print the website under the qr code, its quicker for me to type it in :P [16:32:46 CEST] <Compn> if we are talking about the same QR thing... :D [16:35:13 CEST] <Compn> barb_ : there was someone who wanted to pay a developer to work on 4k visual perception filter [16:35:44 CEST] <Compn> e.g. to cut encoding bits by discarding changes that the human eye cannot really tell the difference of [16:35:59 CEST] <wm4> lolwut [16:36:23 CEST] <Compn> like the audio encoding does, audio ... whatever its called [16:37:03 CEST] <Compn> wm4 : comcast guy is hiring [16:38:24 CEST] <Compn> https://en.wikipedia.org/wiki/Perceptual_audio_coder [16:38:31 CEST] <Compn> that , except for video [16:41:04 CEST] <wm4> that should be within an encoder, not in a separate libavfilter filter (which is what your statement sounded like to me) [16:42:03 CEST] <barb_> hahaha yes, we're talking about the same QR code hahaha [16:42:19 CEST] <barb_> I agree, there are link shorteners for a reason [16:42:45 CEST] <barb_> but they are quite popular nowadays so... hahaha [16:43:02 CEST] <barb_> and it seems to have a pretty cool algorithm behind it [16:45:08 CEST] <barb_> hmmmm, interesting (the discarding thing) [16:46:53 CEST] <Compn> wm4 : you dont think its possible to want a filter to introduce as a step in between raw source and a standard video codec that does not have perceptual feature ? [16:47:15 CEST] <Compn> or whatever hardware encoder that does not have that* [16:47:27 CEST] <Compn> who knows, i dont [16:47:34 CEST] <Compn> but if you want a job i have his contact info :D [16:48:13 CEST] <kierank> What has a qr generator got to do with FFmpeg [16:48:17 CEST] <kierank> Or multimedia [16:49:26 CEST] <barb_> Compn : I'm not sure I'm fit for the job, I've seem some stuff about it in college, but very superficially, but who knows [16:50:16 CEST] <barb_> Compn : So, yes, I'd like to get his contact, maybe we can arrange something [16:51:22 CEST] <barb_> kierank : Maybe because it's a data encoding process that has a media as an output? [16:52:30 CEST] <Compn> its text input though ? [16:52:42 CEST] <Compn> >forgot how qr works :D [16:52:42 CEST] <barb_> I believe so [16:52:54 CEST] <Compn> we dont usually deal with text inputs [16:53:12 CEST] <Compn> some subtitles stuff [16:53:33 CEST] <barb_> mmm [16:54:38 CEST] <barb_> so, do you think it would be better to drop this idea? [16:56:20 CEST] <Compn> kierank doesnt like any of the project ideas in gsoc or opw.... :P [16:56:32 CEST] <barb_> hahaha really? why? [16:56:50 CEST] <Compn> we are an ornery bunch here. [16:57:06 CEST] <funman> can swresample do S32 -> S16 without going through FLTP as intermediate format? [16:57:32 CEST] <Compn> funman : rgb ? yuv ? [16:57:42 CEST] <Compn> err audio [16:57:51 CEST] <Compn> my brain is frozen today. [16:58:30 CEST] <funman> Compn: palettized audio [16:58:57 CEST] <funman> problem is it does 48000 fps right now (floats per second) [17:00:48 CEST] <funman> it's not clear whether swr can run without an intermediate sample format [17:01:04 CEST] <barb_> Compn : haha did you guys try to suggest other ideas? Or it was not up to you? [17:01:37 CEST] <barb_> Compn : Like, there are not many of them in the official page [17:02:26 CEST] <Compn> anyone can suggest ideas, its difficult to think up ideas [17:03:02 CEST] <Compn> that can be done in that timeframe [17:03:41 CEST] <barb_> mmmm, got it [17:05:25 CEST] <Compn> by someone who is new to the code, as there are some advanced concepts in multimedia [17:06:56 CEST] <barb_> sure [17:07:13 CEST] <barb_> Paul also suggested a new audio denoising filter using wavelets [17:07:40 CEST] <barb_> would it be more useful to the purpose of the organization? [17:08:46 CEST] <barb_> (I keep mentioning him and not asking him directly because I only have his email, so it's a bit complicated to get this kind of insight in such short notice) [17:09:16 CEST] <Compn> better ask the admins, i'm just here to take up space [17:09:35 CEST] <Compn> michaelni or durandal_170 (paul's nickname, not here atm) [17:09:44 CEST] <barb_> haha ok, but thank you anyway :) [17:09:45 CEST] <Compn> or whoever else is opw admin / mentor [17:09:51 CEST] <michaelni> atana, you should add a avfilter as in a new file with AVFilter struct and implemenattion not adding code directly to avfilter.c [17:09:52 CEST] <Compn> no problem [17:10:04 CEST] <michaelni> atana, see git log of libavfilter for examples on how to do that [17:10:22 CEST] <wm4> "palettized audio" [17:10:23 CEST] <wm4> what and why [17:12:32 CEST] <funman> michaelni: should swr be able to run without needing an intermediate sample format? [17:14:39 CEST] <wm4> funman: what for [17:14:47 CEST] <wm4> this is just going to make everything more complex and slower [17:15:01 CEST] <funman> wm4: for s32 -> s16 [17:15:58 CEST] <michaelni> for simple sample convert (no resamping) it should not need an intermediate sample format [17:16:15 CEST] <michaelni> as in int32 -> int16 [17:16:50 CEST] <funman> michaelni: ok thanks [17:17:09 CEST] <kierank> funman: try with avresample =p [17:17:29 CEST] <funman> kierank: obviously i thought of it [17:18:05 CEST] <wm4> I think I've seen code that makes swr use floats only if resampling or rematrixing is involved [17:22:15 CEST] <funman> yeah seems i'm not setting channel layouts correctly, that's why it goes through float [17:30:21 CEST] <barb_> michaelni : any input on the QR Code discussion? Or any suggestion on a new denoising filter? [17:36:01 CEST] <michaelni> barb_, QR generator (and reader maybe too?) could be interresting projects if pauls (or someone else) is ok with mentoring these [17:37:26 CEST] <michaelni> about audio denoising i am not upto date with that area so i cannot comment [17:39:37 CEST] <funman> michaelni: works perfectly now, no more intermediate format [17:41:42 CEST] <barb_> michaelni: Oh, I see, so I think I'll commit to the QR generator/reader. Thank you! :) [17:44:03 CEST] <wm4> why does ffmpeg need to be able to read/write Q codes [17:44:05 CEST] <wm4> *QR [17:47:38 CEST] <kierank> ^ this [17:47:52 CEST] <kierank> hey look michaelni, this is how we ended up with a web server last time [17:48:09 CEST] <kierank> but no we will shoot ourselves in the foot again [17:49:07 CEST] <kierank> barb_: I'm sure you mean well but I can't see how a QR code reader and writer fits in ffmpeg [17:49:12 CEST] <wm4> do we still have a web server? [17:50:00 CEST] <barb_> =S [17:50:20 CEST] <barb_> I understand, but do you have anything else in mind? [17:50:31 CEST] <barb_> Like a new feature? [17:51:02 CEST] <cone-192> ffmpeg 03Carl Eugen Hoyos 07master:f04c27fe7ce9: lavc/videotoolboxenc: Enable a53cc by default. [17:51:37 CEST] <barb_> Since I couldn't finish a qualification task on time, I had kind of given up, then today I decided to check with you guys. [17:51:45 CEST] <barb_> And try it anyway [17:52:31 CEST] <barb_> So I needed something new, cause I suppose the suggested projects already have enough submissions [17:52:59 CEST] <iive> are qr codes in question these qube barcodes? [17:53:19 CEST] <michaelni> I see noting wrong with a QR reader or generator, it fits in the many diverse filters we have in libavfilter [17:53:33 CEST] <wm4> I don't know about the project status, but everyone is welcome to contribute any time [17:53:54 CEST] <nevcairiel> its not much of a filter, is it? cant exactly take video and make a QR code out of it, or audio for that matter [17:54:17 CEST] <wm4> libavfilter is a really weird framework for such arbitrary things, and you'll end up with an awkward user interface for low quality implementations [17:54:18 CEST] <michaelni> you can scan a video for QR codes and extract metadata [17:54:54 CEST] <michaelni> also QR codes can get a overlaid human readable text by such filter for example [17:55:27 CEST] <kierank> wm4: demand a vote [17:55:52 CEST] <wm4> who cares [17:55:53 CEST] <michaelni> id consider it fun if i could on the flick of a button see QR codes in a video in human readable form [17:56:06 CEST] <wm4> "flick of a button" [17:56:13 CEST] <wm4> more like writing a 10KB command line [17:56:14 CEST] <michaelni> switch [17:56:32 CEST] <wm4> lol [18:16:23 CEST] <cone-192> ffmpeg 03Andreas Cadhalpun 07release/3.1:1a43626fdf30: configure: fix detection of libopenjpeg [18:16:24 CEST] <cone-192> ffmpeg 03Andreas Cadhalpun 07release/3.1:a22155dacd93: libopenjpegenc: stop reusing image data buffer for openjpeg 2 [18:16:25 CEST] <cone-192> ffmpeg 03Andreas Cadhalpun 07release/3.1:d391719be19b: libopenjpegenc: fix out-of-bounds reads when filling the edges [18:16:26 CEST] <cone-192> ffmpeg 03Andreas Cadhalpun 07release/3.1:a2d3e7392d2d: Changelog: update for recent commits [18:17:22 CEST] <cone-192> ffmpeg 03Andreas Cadhalpun 07release/3.0:ada229e66fea: configure: fix detection of libopenjpeg [18:17:23 CEST] <cone-192> ffmpeg 03Andreas Cadhalpun 07release/3.0:de42af2beef7: libopenjpegenc: stop reusing image data buffer for openjpeg 2 [18:17:24 CEST] <cone-192> ffmpeg 03Andreas Cadhalpun 07release/3.0:0887f1835d69: libopenjpegenc: fix out-of-bounds reads when filling the edges [18:17:25 CEST] <cone-192> ffmpeg 03Andreas Cadhalpun 07release/3.0:f309b698e6b7: Changelog: update for recent commits [18:26:58 CEST] <cone-192> ffmpeg 03Andreas Cadhalpun 07master:c143a9c96ff9: aiffdec: fix division by zero [18:41:21 CEST] <kierank> wm4: are we getting a qr code filter then [18:42:01 CEST] <JEEB> it didn't go anywhere [18:42:07 CEST] <wm4> yeah, there was no further discussion [18:42:10 CEST] <JEEB> as in, the discussion stopped after the /part [18:42:33 CEST] <JEEB> or more like, had finished before that but anyways [18:43:28 CEST] <JEEB> also is the way to go with XML parsing now libxml2 or libexpat? [18:43:42 CEST] <JEEB> if someone would start working on a DASH demuxer [18:44:52 CEST] <wm4> if I do the latter, I'll use libexpat [18:45:01 CEST] <JEEB> ok [19:10:05 CEST] <RiCON> probably better to wait/decide with hds demuxer before starting to work on that? [19:11:08 CEST] <RiCON> if the hds author goes with libxml2 and dash uses expat one would end with two separate xml libs in ffmpeg? [19:12:11 CEST] <cone-192> ffmpeg 03Carl Eugen Hoyos 07master:df9b151a1bdc: doc: Update x264 option a53cc, forgotten in 93e04102. [19:15:35 CEST] <Chloe> We obviously just need to write our own xml lib within libavutil [19:17:14 CEST] Action: JEEB gently stabs stabs Chloe [19:17:17 CEST] <BtbN> If there is already an xml parser ready to be used via configure, used by other parts of ffmpeg, using it seems a reasonable requirement. [19:19:22 CEST] <RiCON> since people can choose between 4 tls libs why not choose between 2/3 xml libs too, one might say [19:19:52 CEST] <Chloe> barb_: I see you were interested in the XPM encoder/decoder, why don't you just do that? [19:20:07 CEST] <JEEB> BtbN: yes - that's why I was wondering which way the whole thing was heading [19:20:19 CEST] <JEEB> because I sure as hell wouldn't want to rewrite my XML parsing code [19:21:52 CEST] <barb_> Chloe : I'll be applying for that project as well, I was just going for a plan B, just in case somebody else wants to do it too. [19:24:52 CEST] <nevcairiel> afaik we currently have funding for only one outreachy applicant either way, so your best chances would likely be to put all the energy into the most promising project [19:25:21 CEST] <nevcairiel> (unless some more funds appeared when I wasnt paying attention) [19:25:36 CEST] <wm4> <RiCON> if the hds author goes with libxml2 and dash uses expat one would end with two separate xml libs in ffmpeg? <- not like I care [19:25:53 CEST] <Chloe> I care, pls no [19:26:23 CEST] <wm4> I encouraged that contributor to use libexpat, and whether others agree or not or whether they agree the patch in whatever form is not up to me [19:27:55 CEST] <Chloe> XML isn't like TLS, there is no benefit for using multiple libraries, also 3/4 of the TLS libraries are API compatible (afaik). There is little reason to use yet another XML library [19:29:32 CEST] <nevcairiel> the only real benefit of supporting both openssl and gnutls is licensing and system availability - which is the same argument one might make for any kind of library [19:29:44 CEST] <nevcairiel> not that it makes it any better [19:29:57 CEST] <BtbN> So libavxml it is [19:31:30 CEST] <barb_> nevcairiel : mmmm, I see... [19:33:14 CEST] <Plorkyeran> the only unusual thing about TLS libraries is that none of them have a reasonable license [19:33:28 CEST] <Plorkyeran> so it's easier to argue for giving the user the choice of which unreasonable license they want [19:34:48 CEST] <nevcairiel> gnutls is lgpl, is that so bad? its the mostl iberal license you can build ffmpeg under as well [19:35:36 CEST] <Plorkyeran> oh right it is [19:36:02 CEST] <Plorkyeran> I have gnutls mentally lumped under "awkward license" because it was an issue for using it in ios apps [19:36:03 CEST] <wm4> more importantly all tls libs suck [19:36:07 CEST] <Plorkyeran> but that's an unusual case [19:36:47 CEST] <JEEB> btw, was the internal representation of subtitles in lavc ASS? [19:36:50 CEST] <Compn> i thought openssl sucks and was full of buggy holes, and thats why people were moving to polarssl and gnutls :P [19:37:06 CEST] <JEEB> so if you want to support various styled text related things you have to parse ASS tags? [19:37:13 CEST] <JEEB> (if you encode subtitles) [19:37:38 CEST] <Compn> i think ffmpeg originally converted all srt and subtitle tags to ass internally, yes [19:37:46 CEST] <Compn> but i think that has since been changed. [19:38:14 CEST] <nevcairiel> it should still do that if you use a subtitle "decoder" [19:38:22 CEST] <JEEB> ok, I'm just thinking of implementing a subtitle format and the first thing of course that comes to mind is the internal representation [19:38:33 CEST] <Compn> ah [19:38:34 CEST] <JEEB> from which I would then create the output format [19:38:48 CEST] <nevcairiel> ubitux knows all the details of course, but from what I understand internally all text subtitles are "decoded" to ass syntax [19:38:55 CEST] <JEEB> ok [19:39:36 CEST] Action: Compn members the old days when people complained about the "ass" name [19:44:00 CEST] <ubitux> JEEB: yes, internal decoded representation of text subtitles is ASS [19:44:13 CEST] <ubitux> text subtitles decoders output ASS [19:44:25 CEST] <wm4> "decoders" [19:44:32 CEST] <ubitux> (and demuxers just split the original format) [19:44:55 CEST] <ubitux> encoders have access to an ASS parser which allows them to define a bunch of callbacks [19:45:08 CEST] <ubitux> for handling different styles [19:48:05 CEST] <JEEB> ok [19:48:19 CEST] <JEEB> like italic and crap like that I guess [19:48:27 CEST] <Compn> yep [19:48:43 CEST] <Compn> italic, underline, bold... [19:48:53 CEST] <Compn> colors [19:49:04 CEST] <JEEB> alright [19:55:18 CEST] <cone-192> ffmpeg 03Muhammad Faiz 07master:92de89d14acb: avfilter/firequalizer: optimize gain/gain_entry command [19:55:19 CEST] <cone-192> ffmpeg 03Muhammad Faiz 07master:923488bdaaab: avfilter/showcqt: add font option [20:46:45 CEST] <cone-192> ffmpeg 03Andreas Cadhalpun 07master:9959a52b14bc: astdec: fix division by zero [20:46:46 CEST] <cone-192> ffmpeg 03Andreas Cadhalpun 07master:bc7e128a6e8e: westwood_aud: prevent division by zero [23:31:18 CEST] <atomnuker> michaelni: could you test the v3 of the AAC SIMD for crashes, should be up on the ML soon? [23:36:52 CEST] <rcombs> ubitux: might be a good time to make the internal format """pseudo-ass""", wherein backslash-escapes are Actually A Thing [23:37:34 CEST] <ubitux> rcombs: we'll probably do that when assv5 is out ;) [23:43:51 CEST] <michaelni> atomnuker, will do [00:00:00 CEST] --- Tue Oct 18 2016

1 0

ffmpeg.log.20161017
by burek 18 Oct '16

18 Oct '16

[00:01:19 CEST] <emilsp> I'll check for sure just now [00:01:45 CEST] <emilsp> but I don't know whether the ffmpeg binary will be part of ffmpeg (the package) or the new and shiny ffmpeg2.8 (the package) [00:02:16 CEST] <emilsp> ah, the main swscale version is 262500 [00:05:17 CEST] <furq> the ffmpeg package in arch is 3.1 [00:05:36 CEST] <furq> 2.8 will be the legacy package for stuff which hasn't updated to the new api [00:06:04 CEST] <emilsp> oh, nice [00:06:18 CEST] <emilsp> so then, in 3.1 SwsContext isn't typedeffed by default ? [00:08:17 CEST] <c_14> It's not typedeffed [00:08:22 CEST] <c_14> Was it in the earlier releases? [00:08:38 CEST] <emilsp> I'm judging by the example code I was looking at [00:08:49 CEST] <c_14> The functions all take and return a struct SwSContext [00:08:51 CEST] <emilsp> https://github.com/filippobrizzi/raw_rgb_straming/blob/master/client/x264de… [00:08:52 CEST] <c_14> What example code? [00:09:10 CEST] <emilsp> maybe C++ does stupid things [00:10:33 CEST] <emilsp> it does indeed seem that C++ might be doing incredibly stupid things :( [00:10:36 CEST] <c_14> yep [00:10:38 CEST] <c_14> That's C++ [00:10:43 CEST] <emilsp> FUCK ME [00:11:21 CEST] <emilsp> sorry about that [00:11:35 CEST] <emilsp> and the spam; again, thank you both very much for the help [01:50:50 CEST] <MrMonkey31> uhm wow, so trim feature is a gyp... I tried to achieve a simple "crop out" of the middle of a video and I got a file which plays 10 minutes of nothing before any data begins [01:53:59 CEST] <MrMonkey31> looking at the docs it waits till the end to say none too helpfully insert a setpts filter. this is the kind of stuff that melts ordinary people's BrAiNS!!..... [01:55:16 CEST] <MrMonkey31> I saw 'setpts=PTS-STARTPTS ' elsewhere, which I presume is the equation needed to 'fix' it to act like a simple trim, but can anyone confirm this before I start another 20 min encode? [01:55:41 CEST] <furq> that's correct [01:56:36 CEST] <furq> you'll need the same thing with asetpts if you're using atrim [01:57:03 CEST] <MrMonkey31> yeah, thx dood. skullmelt averted! [01:58:03 CEST] <furq> it would be nice if those filters had examples with setpts considering that's probably how they're usually used [01:59:39 CEST] <MrMonkey31> nah! I'm not prepared to let my audio be handled automatically. instinctively I just forebode all manner of desync, under-run and god-only-knows what other stream "problems". I can't wrap my mind around stamps controlling a sound wave, man. that's just too out there [02:09:43 CEST] <klaxa> shouldn't pts (and dts) even be "filterable" at (de)muxing? there would be no need to re-encode in that case, but i'm not sure if it's in ffmpeg [06:58:51 CEST] <bencc> I'm doing screen capture with "-vcodec libx264 -pix_fmt yuv420p -preset:v ultrafast -crf 0" [06:59:23 CEST] <bencc> what parameters can I change to make the output file smaller without increasing cpu too much? [10:34:13 CEST] <maarhart> hi, I want to do the same as http://stackoverflow.com/questions/21510521/ffmpeg-move-a-slider-image-over… but removing the upper figure, so keeping just the waveform and the overlayed picture. how can I do this? [10:37:56 CEST] <maarhart> sorry, disregard my question [10:44:32 CEST] <pihpah> Anyone has ever tried to use Intel Quick Sync Video with ffmpeg? [10:50:06 CEST] <emilsp> is there a perror equivalent for ffmpeg ? [10:50:51 CEST] <BtbN> av_strerror? [10:51:08 CEST] <BtbN> just look at error.h, there are several functions and macros. [10:52:23 CEST] <emilsp> thanks :) [11:01:24 CEST] <emilsp> this might be a bit of a stupid question, but what's the coordinate system that is used for ffmpeg ? [11:01:38 CEST] <emilsp> is y=0 at the top of the frame ? or at the bottom ? [11:06:57 CEST] <BtbN> as ffmpeg usually uses offset=x+linesize*y I'd say it's safe to assume y=0 is the top row. [12:21:36 CEST] <termos> I'm transcoding using libfdk_aac with sample_fmt s16, but ffprobe keeps telling me my output is fltp format and I get no audio. Why is it detecting it wrongly, or is there something I'm not setting? [12:21:46 CEST] <termos> It only happens when I transcode audio from ac3 to aac [12:22:17 CEST] <BtbN> the sample_fmt only matter when writing raw pcm. [12:23:40 CEST] <termos> hm interesting, but I'm setting it also in my filter graph with the filter aformat=sample_fmts=s16. is it not really doing anything? [12:24:08 CEST] <BtbN> it is, it will convert to that format, and if the aac encoder does not support it as input, it will be automatically converted to something it supports again. [12:24:34 CEST] <BtbN> And the output ffprobe shows of the aac stream depends entirely on the output of the aac decoder it uses. [12:43:29 CEST] <termos> ok so the aac encoder will convert it for me anyway [12:44:20 CEST] <termos> it's just very curious that if I transcode from aac -> aac and mp3 -> aac it's fine, but ac3 -> aac seems to cause issues with the output being interpreted as fltp [12:44:38 CEST] <termos> I must have forgotten something, just can't figure out what [12:45:26 CEST] <BtbN> It will not convert anything [12:45:33 CEST] <BtbN> it's aac it writes, not some sample format [12:49:36 CEST] <iive> is fltp just float point? [12:50:03 CEST] <iive> most audio decoders work with floats internally, so that's what they output as native. [12:50:12 CEST] <iive> decoder->codecs [12:50:31 CEST] <BtbN> fltp should just be 32 bit floats, yes [12:51:17 CEST] <BtbN> https://github.com/FFmpeg/FFmpeg/blob/master/libavutil/samplefmt.h#L69 [12:53:49 CEST] <termos> the audio plays for maybe 1s when I start my stream, then it stops [12:54:19 CEST] <termos> rtmp://cp353594.live.edgefcs.net/live/udp_sound_133377_1364k@316196 this is an example stream that I set up [13:30:55 CEST] <s0126h> why so many people obsessed with 10 video encoding [13:31:01 CEST] <s0126h> why so many people obsessed with 10 bit video encoding [13:35:37 CEST] <qmr> mo bits mo problems I always say [13:36:26 CEST] <termos> bitses? [14:19:16 CEST] <klaxa> s0126h: with 8-bit per channel banding is quite an issue, especially for animated content [14:19:58 CEST] <s0126h> what is "banding"? [14:20:15 CEST] <DHE> when you can see the colours as distinct lines [14:20:29 CEST] <s0126h> do you have example of a picture [14:20:51 CEST] <DHE> 256 shades from black to while when you have 1920 pixels in one direction and 1080 in the other (or more!) starts to become noticeable [14:22:30 CEST] <s0126h> do you have a example [14:25:30 CEST] <DHE> me personally, no [14:27:14 CEST] <s0126h> anybody have an example of this banding issue ? picture or something [14:27:43 CEST] <klaxa> https://en.wikipedia.org/wiki/Colour_banding [14:28:36 CEST] <s0126h> i see a photograph there [14:28:40 CEST] <s0126h> what's wrong with it [14:31:28 CEST] <klaxa> in the sky there are bands of different shades of blue [14:31:49 CEST] <klaxa> zoom in to see it better [14:45:24 CEST] <s0126h> klaxa i see [14:45:35 CEST] <s0126h> klaxa why si it doing that [14:46:02 CEST] <s0126h> and which video encoder is that [14:46:56 CEST] <BtbN> it is doing that because there are only that many shades of blue in 8 bit color. [14:47:19 CEST] <s0126h> typo? [14:47:34 CEST] <s0126h> that sentence didn't make sense [14:47:37 CEST] <klaxa> it does [14:47:55 CEST] <klaxa> you have 24 bits for all of rgb [14:48:12 CEST] <s0126h> so it's only noticable in blue? [14:48:15 CEST] <klaxa> but in this picture it's only a change of a few bits [14:48:40 CEST] <BtbN> The color doesn't matter [14:48:42 CEST] <klaxa> in the gradient the color changes by less than 1 bit per pixel [14:48:43 CEST] <BtbN> there are 8 bits per color [14:48:55 CEST] <s0126h> btbn my bad, your sentence do make sense [14:49:45 CEST] <s0126h> so it doesn't matter what video encoder is used? all 8bit encoding will do that in the sky like the picture? [14:54:15 CEST] <BtbN> It's not an encoding artifact. [14:54:25 CEST] <BtbN> It would happen with plain lossless images as well. [14:54:38 CEST] <s0126h> are you serious [14:55:02 CEST] <s0126h> why would it do that on lossless encoding? if original didn't do that [14:55:14 CEST] <s0126h> unless original did it too [15:00:41 CEST] <BtbN> Becaue 8 bit colors are 8 bit colors [15:01:28 CEST] <furq> the original would do that by definition [15:01:55 CEST] <furq> if you reduce the bit depth then it's not a lossless conversion [15:03:51 CEST] <s0126h> furq but what if original is also 8bit [15:04:02 CEST] <furq> then it will have those banding artifacts [15:52:39 CEST] <beauty> hello [15:54:30 CEST] <klaxa> hi [15:57:52 CEST] <beauty> how to use a ffmpeg process to surpport multi channel stream data? [16:00:39 CEST] <klaxa> what [16:00:48 CEST] <klaxa> can you explain that a bit more specifically? [16:18:35 CEST] <beauty> klaxa: how to make ffmpeg support four thousands videos to decode at the same time? [16:18:51 CEST] <beauty> the videos is streamed data. [16:21:15 CEST] <klaxa> just use a lot -i [16:21:34 CEST] <klaxa> make sure you increase your filedescriptor limit [16:23:46 CEST] <beauty> ?? [16:24:01 CEST] <beauty> I want to use ffmpeg code. [16:24:07 CEST] <beauty> not ffmpeg command. [16:24:57 CEST] <retard> you're not really describing what you're trying to do at all [16:27:35 CEST] <beauty> oh, sorry [16:28:08 CEST] <beauty> I want to use ffmpeg decode multi streamed video file at the same time. [16:28:18 CEST] <beauty> I want to use ffmpeg decode multi streamed video files at the same time. [16:53:31 CEST] <Bogas> Hello all i have to convert this code from ffmped to ffmbc what i need to change ? [16:53:33 CEST] <Bogas> C:\utils\ffmpeg.exe -i "S:\pharos\sandbox_cubix\Production\CubixIn\AETN\ATN-29.mxf" -y -s 1024x576 -c:v libx264 -profile:v baseline -level 3 -b:v 1117k -maxrate 1200k -threads 0 -vf "[in]yadif=0:1,drawtext=fontfile=/windows/fonts/arial.ttf: timecode='00\:00\:00\:00': r=25.00: \x=(w-tw)/2: y=(1*lh): fontcolor=white:fontsize=80: box=1: boxcolor=0x00000000@1[out]" -aspect 1.78 -pix_fmt yuv420p -acodec libvo_aacenc -ar 48000 -ac 2 -ab 25 [16:54:57 CEST] <Bogas> any one can help me im getting a error related to the 1 frame and i read to try to change to ffmbc but i do not know how to do this [16:58:01 CEST] <Bogas> Hello any one here ? [16:58:39 CEST] <BtbN> This is #ffmpeg. I don't even have any idea what ffmbc is. [17:00:11 CEST] <BtbN> https://github.com/bcoudurier/FFmbc "This branch is 441 commits ahead, 50371 commits behind FFmpeg:master. " oh my. Well, it's an ancient ffmpeg fork. Something you definitely don't want to use. [18:00:27 CEST] <acamargo> hi. I'm segmenting a live feed using ffmpeg to produce 60 seconds duration time files. but when I import those files in adobe premiere the duration is lesser than ffprobe/mediainfo show. any tip about this issue? I'm producing keyframes at 30 frames interval and capturing at 29.97fps [18:01:40 CEST] <acamargo> one more thing, when I concat the files using ffmpeg the result duration is fine. [18:03:26 CEST] <acamargo> but when I concat the segments on premiere, the duration is lesser than ffmpeg's concated file [18:05:22 CEST] <acamargo> it seems like premiere is losing/discarding some frames [18:09:55 CEST] <acamargo> here's my ffmpeg command http://pastebin.com/3QLuSMcA [18:45:45 CEST] <ferdna> how do i tell ffmpeg to keep recording to the same file...? [18:46:04 CEST] <ferdna> i dont want to overwrite the already created file [18:46:10 CEST] <ferdna> ( option -y ) [18:57:22 CEST] <c_14> Not supported internally at all (though you can hack it by using shell redirection for mpegts and similar formats) [19:06:04 CEST] <michael_> Hi, I have a question, born out of this sharex github issue: https://github.com/ShareX/ShareX/issues/205 [19:06:23 CEST] <michael_> developer said, in december 2015, that root cause is ffmpeg, not supporting pause [19:06:36 CEST] <michael_> is this situation the same as of today? [19:07:01 CEST] <michael_> (forgive me for avoiding to delve into ffmpeg documentation, i'm not the expert type of guy) [19:19:37 CEST] <c_14> Well, ffmpeg supports pause in the same way that every posix process supports pause. [19:19:39 CEST] <c_14> SIGSTOP [19:19:44 CEST] <c_14> Other than that, no [19:21:45 CEST] <JEEB> also how do you pause, for example, live streaming to youtube (most probably done through RTMP?)? do you just push null packets to the FLV? [19:22:34 CEST] <JEEB> pretty sure all those things would have to be defined and then APIs defined around such functionality in specific output protocols [19:22:38 CEST] <furq> pause/enter works here, but obviously you need a terminal up to do that [19:22:38 CEST] <furq> and yeah i wouldn't expect it to work reliably [19:23:16 CEST] <furq> pausing while capturing or broadcasting is probably going to break something [19:23:44 CEST] <JEEB> or in other cases I think the "pause" functionality starts pushing some pre-defined overlay or something to the end point [19:33:22 CEST] <furq> i just tested with lavfi and rtmp and i can confirm that it does sort of work and it does break something [19:59:25 CEST] <michael_> so from what i get, it means pausing is not a natural evolution for ffmpeg, or easy to implement, hence i rather not insist on sharex issue. thanks for the info. it appears the only way is to make many screen recordings, and then glue them together, afterwards [20:02:03 CEST] <JEEB> nah, it just isn't clear what exactly is meant by "pause" [20:02:59 CEST] <JEEB> because in most cases it's just not as simple as "pause the whole process from input to pushing packets over the internets" [20:03:20 CEST] <JEEB> people and esp. protocols tend to want different things [20:03:42 CEST] <JEEB> or more specifically protocols can just kick you out if you do nothing with the connection [20:05:30 CEST] <furq> michael_: if you're recording to a file then the current behaviour might be what you want [20:05:53 CEST] <furq> you'd need some way to send keys to the process, though [20:07:11 CEST] <furq> if you're broadcasting then you could probably hack something together with sendcmd [20:07:35 CEST] <michael_> i didn't realize the implications, i was just focusing on sharex feature, screen recording which creates a video file, no internet, no broadcast or anything [20:08:06 CEST] <JEEB> the ticket mentions streaming etc [20:08:27 CEST] <JEEB> and thus I of course started going off of that since that's what many people nowadays do with their recordings [20:09:48 CEST] <michael_> i have to admit, i read the issue fast, it wasn't created by me. developer closed my feature request, and sent me to this issue. indeed, if you get streaming into the picture, or shared screen, it gets nasty [20:11:01 CEST] <michael_> but, for simple screen recording, which is creating a video file, does ffmpeg (current version) have such a feature as pause? [20:11:06 CEST] <JEEB> and then with the screen capture modules it depends how you can "pause" them or "destroy" them [20:11:46 CEST] <JEEB> well, furq seemed to note that something might be available even if you just call the ffmpeg cli (as opposed to utilizing the APIs within your application) [20:14:02 CEST] <michael_> sharex most probably uses ffmpeg cli (in options, it shows command line preview, like: -y -rtbufsize 100M -f gdigrab -framerate 20 -offset_x 0 -offset_y 0 -video_size 1920x1200 -draw_mouse 1 -i desktop -f dshow -i audio="Microphone (Realtek High Defini" -c:v libx264 -r 20 -preset veryfast -tune zerolatency -crf 30 -pix_fmt yuv420p -c:a aac -strict -2 -ac 2 -b:a 96k "output.mp4") [20:16:40 CEST] <JEEB> yeah, that's the integration most things begin with [20:16:50 CEST] <JEEB> and then they notice that they can do X,Y,Z better by utilizing the API [20:20:19 CEST] <DrSlony> Hello, what is the best automated video stabilization method available in the latest ffmpeg? Has anything changed on this front in the last 3 years? [20:26:09 CEST] <michael_> ok, thank you guys, thank you JEEB, thank you furq. i appreciate your help, now it's clearer [20:29:04 CEST] <furq> michael_: you could just use ffmpeg directly [20:29:57 CEST] <michael_> directly, for screen recording? :D i wasn't aware of it [20:30:16 CEST] <furq> yeah [20:30:30 CEST] <furq> the options you just pasted from sharex will capture your entire desktop to output.mp4 [20:30:44 CEST] <michael_> i will look into it, sounds interesting [20:30:58 CEST] <furq> it looks like sharex does some fancy stuff but if you just want to capture your desktop or a window then ffmpeg will do the job [20:31:34 CEST] <furq> those options also look really bad for recording to a file [20:31:42 CEST] <furq> -tune zerolatency in particular [20:33:09 CEST] <michael_> my issue with sharex is that i would like to pause the recording, and then continue afterwards. other than that, even with imperfect options, it does a good job, small file, enough quality [20:34:45 CEST] <furq> the other options are questionable but down to taste [20:34:51 CEST] <furq> but -tune zerolatency is just wrong [20:35:31 CEST] <michael_> i don't know what this option should do. why is it wrong? i can open an issue on their github if i understand [20:36:06 CEST] <furq> it makes the file much bigger and disables frame multithreading to reduce the encoder latency [20:36:21 CEST] <furq> which is potentially an issue if you're live streaming [20:36:23 CEST] <DrSlony> I use: ffmpeg -y -f x11grab -show_region 1 -s 1920x1080 -i :0.0+0,0 -an -c:v libx264 -preset ultrafast -qp 0 -threads 0 /ram/drive/screencast.mp4 [20:36:25 CEST] <furq> otherwise it's just a waste [20:37:42 CEST] <michael_> good point then, i will take this to their github [20:38:21 CEST] <furq> DrSlony: how much lossless 1080p screencast can you fit in that ramdisk [20:38:27 CEST] <DrSlony> enough for my needs [20:38:38 CEST] <DrSlony> never reached the limit so can't tell '] [20:38:39 CEST] <DrSlony> ;] [20:42:37 CEST] <DrSlony> Is there a better codec for *intermediate* files if I want either lossless or close to it lossy? something which won't bog down the CPU while recording but which also wont take jiggawatts of space. I transcode all video after recording. [20:46:20 CEST] <bencc> makes sense to do transcoding over tmpfs? [20:46:37 CEST] <DHE> probably not. if you don't have the disk space, you probably won't have the RAM [20:47:25 CEST] <DHE> x264 with -qp 0 will be lossless, so long as you're okay with the 4:2:2 colourspace loss (there are more options for fixing that though) [20:47:42 CEST] <furq> the fastest lossless codec in ffmpeg is ffvhuff afaik [20:47:43 CEST] <DHE> other options include ffv1, huffyuv or the ffmpeg variant... [20:47:48 CEST] <DHE> that's the one [20:47:58 CEST] <furq> but it'll be bigger than ffv1 and x264 lossless [20:48:03 CEST] <bencc> DHE: I have the disk space and enough RAM. trying to avoid iowait [20:48:13 CEST] <bencc> DHE: when transcoding several files at the same time [20:48:16 CEST] <DHE> so you have to choose between CPU or disk space [20:48:22 CEST] <DHE> oh, then you need an SSD [20:48:24 CEST] <Jaex> furq: https://trac.ffmpeg.org/wiki/StreamingGuide suggests -tune zerolatency [20:48:45 CEST] <furq> it also says "Streaming" right at the top of the page [20:49:04 CEST] <furq> like i said, it's potentially useful for streaming [20:49:05 CEST] <Jaex> yes? [20:49:13 CEST] <bencc> DHE: tmpfs increase RAM usage? [20:49:14 CEST] <Jaex> <furq> which is potentially an issue if you're live streaming [20:49:21 CEST] <Jaex> you told it is issue for live streaming [20:49:42 CEST] <DHE> bencc: tmpfs is a linux filesystem that operates as a ramdisk [20:49:42 CEST] <furq> i meant that encoder latency is potentially an issue if you're live streaming [20:49:53 CEST] <DHE> about the only exception is that tmpfs does support being swapped out [20:49:54 CEST] <bencc> DHE: sorry, I meant does tmpfs increase CPU usage? [20:49:55 CEST] <Jaex> it is also issue for recording [20:50:00 CEST] <Jaex> otherwise fps will drop alot [20:50:04 CEST] <furq> no [20:50:20 CEST] <bencc> DHE: you said I have to choose between CPU and disk space. [20:50:25 CEST] <furq> that's what the buffer is for [20:50:42 CEST] <Jaex> buffer is not solution it will nonstop fill [20:50:45 CEST] <Jaex> you cant rely on buffer [20:51:03 CEST] <furq> you can absolutely rely on the buffer if you're not filling it faster than you can encode [20:51:14 CEST] <DrSlony> I don't mind lossy as long as it's easy on the CPU. Is there a lossless or (not-very-)lossy codec which is easy on the CPU and doesn't require chroma subsampling? [20:51:26 CEST] <bencc> DHE: I have 2TB HDD and 16GB RAM. using tmpfs get iowait of the way. I'm saving files to s3 anyway so I don't mind losing them [20:51:27 CEST] <Jaex> ofc it is filled faster than we can encode [20:51:38 CEST] <Jaex> this is why using -tune zerolatency [20:51:56 CEST] <DHE> bencc: ramdisk is only going to hold ~15 GB of space, max then. is that enough space? I'm guessing No if you have multiple projects going at once [20:52:02 CEST] <furq> what [20:52:54 CEST] <bencc> DHE: I'm cpu bound anyway. can do about 4 concurrent transcodings. each file is about 1-2 GB so RAM is enough [20:54:27 CEST] <bencc> DHE: transcoding jobs will start and stop randomly so not all will use max ram at the same time [21:52:17 CEST] <iamtakingiteasy> hi i am not strictly on ffmpeg topic (though using it extensively), but rather with general question: what readings or basic concepts can you recommend/provide for audio/video synchronization? i am implementing a realtime rtmp -> mp4 muxer and confused a lot by audio bitrates, samplerates and bit depths, how they are generally mapped on video frames? [21:54:11 CEST] <iamtakingiteasy> until considering audio all was greate with h264 payload -- simply use timescale equal to FPS and incriment counter each full frame by 1, but now i am not so sure what can i do about mp3/aac payloads [21:54:57 CEST] <iamtakingiteasy> they have a lot of samples in single frame and i am not sure if it could be splitted without decoding in order to match video frames [21:57:07 CEST] <DHE> the demuxer will provide you with a time base and pts values for all packets. typically you only decode a little bit of the video/audio to determine codec parameters - you don't need to decode the whole thing for simple remuxing [21:57:13 CEST] <DHE> at least, most formats do that [21:57:38 CEST] <iamtakingiteasy> yeah, that what i do. but how can i match 1/nth audio frames to 1/FPSths frames of video? [21:58:25 CEST] <iamtakingiteasy> is it even possible without complete decoding of audio samples and re-encoding them back in matching chunks? [21:59:24 CEST] <iamtakingiteasy> i am worried about timings: timescale and frametick parameters [22:00:25 CEST] <iamtakingiteasy> or should i rather adjust video frametick/timescale to match audio instead? [22:01:32 CEST] <iamtakingiteasy> is simple 1/FPS frametick interval okay? [22:01:54 CEST] <JEEB> usually audio and video are going with different tick rates anyways [22:02:22 CEST] <JEEB> and with any sane muxing library you should be able to just feed it packets for the streams as they come [22:02:28 CEST] <JEEB> without thinking too much how it's doing the interleaving [22:03:18 CEST] <JEEB> basically in mp4 you have a DTS, CTS and duration [22:03:26 CEST] <JEEB> (CTS being very similar to PTS) [22:04:50 CEST] <iamtakingiteasy> hou. i am using [ftyp moov] header with zero defaults and [moof mdat]+ sequences each with own [tfdt] box providing the time offset expressed in number of frames transmitted [22:05:24 CEST] <iamtakingiteasy> not sure what DTS, CTS and PTS is [22:05:51 CEST] <JEEB> Decoding Time Stamp, whatever the C was Time Stamp, and Presentation Time Stamp [22:06:01 CEST] <iamtakingiteasy> aha [22:06:09 CEST] <JEEB> so as long as you have timestamps for all samples the decoding entity can match things up [22:06:49 CEST] <iamtakingiteasy> so i shouldn't worry much about keeping them in sync at muxer side? [22:07:13 CEST] <JEEB> you should worry about having the timestamps correct :P [22:07:17 CEST] <iamtakingiteasy> aha [22:07:18 CEST] <JEEB> and that's it [22:07:50 CEST] <iamtakingiteasy> okay, thanks, i was under impression that i had to match each video frame with related audio frame having them exactly the same duration long [22:07:57 CEST] <JEEB> no [22:08:07 CEST] <JEEB> I mean, that happens very very rarely [22:08:15 CEST] <JEEB> just look at any normal mp4 file with L-SMASH's boxdumper :P [22:08:22 CEST] <JEEB> `boxdumper --box file` [22:08:36 CEST] <JEEB> (you probably want to either redirect to file or to less or something) [22:09:04 CEST] <iamtakingiteasy> i am currently using mp4dump from bento tools and mp4file --dump from whatever it comes from, thanks for another tool reference [22:09:34 CEST] <iamtakingiteasy> thanks for the hints [22:09:48 CEST] <JEEB> yeah, you can never have too few tools for stuff like that :) [22:09:54 CEST] <iamtakingiteasy> indeed [22:10:06 CEST] <JEEB> I should really get my smooth streaming things upstreamed to it [22:10:25 CEST] <JEEB> have some patches in my fork that I utilized while debugging some legacy crap [00:00:00 CEST] --- Tue Oct 18 2016

1 0

ffmpeg-devel.log.20161016
by burek 17 Oct '16

17 Oct '16

[00:44:06 CEST] <cone-020> ffmpeg 03Muhammad Faiz 07master:a11757d7cba9: avfilter/showcqt: add csp option [00:53:46 CEST] <cone-020> ffmpeg 03Muhammad Faiz 07master:7c5fca15bbc6: avfilter/firequalizer: change WFUNC_MIN/MAX to NB_WFUNC [00:53:47 CEST] <cone-020> ffmpeg 03Muhammad Faiz 07master:01ab60201be9: avfilter/firequalizer: add tukey window [00:58:37 CEST] <cone-020> ffmpeg 03Muhammad Faiz 07master:8e53c1567c7a: doc/filters: document tukey window on firequalizer [01:53:53 CEST] <cone-020> ffmpeg 03James Almer 07master:bad8bbc26aa3: avformat/matroskadec: set aspect ratio only when DisplayWidth and DisplayHeight are in pixels [01:53:54 CEST] <cone-020> ffmpeg 03James Almer 07master:90826031a205: avformat/matroskaenc: write a DisplayUnit element when aspect ratio is unknown [17:20:11 CEST] <cone-168> ffmpeg 03James Almer 07master:4b0f37dadb74: avcodec/utils: print Chroma Location string in verbose log level [18:59:03 CEST] <ubitux> is this avpacket fate test really useful for testing anything? [19:22:07 CEST] <cone-168> ffmpeg 03James Almer 07master:8b59ce034268: Partially revert "avformat/matroskadec: set aspect ratio only when DisplayWidth and DisplayHeight are in pixels" [19:22:55 CEST] <cone-168> ffmpeg 03Clément BSsch 07master:ad2d2ebd4ee9: ffmpeg: remove unused ist parameter in do_subtitle_out() [19:23:47 CEST] <ubitux> matroska is getting standardized? [19:30:48 CEST] <jamrial> yeah, alongside ffv1 and flac it seems [19:38:35 CEST] <ubitux> nice [19:54:21 CEST] <ubitux> damn the madness of do_video_out()... with half indented dup loop (to i suppose reduce conflict solving with libav) [19:55:51 CEST] <wm4> having shit code is part of ffmpeg culture [19:57:14 CEST] <wm4> (can we change this please) [19:58:15 CEST] <ubitux> if we don't have to merge libav anymore i can reindent if you want [19:58:18 CEST] <ubitux> ;) [19:58:48 CEST] <ubitux> speaking of this, i hope to get done with subtitles asap, it really helps cleaning things up [19:58:54 CEST] <ubitux> then i'll be available again for merges [19:59:06 CEST] <ubitux> i hope we won't be 1000+ when i'm available again [19:59:29 CEST] <ubitux> (423 currently) [20:00:14 CEST] <ubitux> ah, we're at an avconv merge window [20:00:20 CEST] <ubitux> annoying stuff [20:12:23 CEST] <nevcairiel> i was going to mvoe forward but it would break qsv transcoding [20:12:29 CEST] <nevcairiel> we've been discussing what to do about that [20:13:32 CEST] <nevcairiel> tried contacting our qsv guy to port the required changes, but afaik jamrial and michaelni both didnt receive any response [20:16:54 CEST] <ubitux> tell him we trash his work in 3days? [20:20:30 CEST] <wm4> +1 [20:23:26 CEST] <nevcairiel> the positive thing is that after browsing the upcoming changes, after this avconv batch it should be quite a bit easier for a long w hile [20:23:57 CEST] <ubitux> i mean, we're not going to put the project into jeopardy just because an external guy doesn't felt like maintaining his niche code [20:24:15 CEST] <ubitux> so better stress him out and ditch the problem asap [20:24:56 CEST] <nevcairiel> jamrial suggested reverting to libav state and just porting some of the improvements individually [20:25:05 CEST] <nevcairiel> not sure if he plans to push that further [20:28:30 CEST] <ubitux> unrelated question: we always have filters for a/v? [20:28:37 CEST] <ubitux> (in a transcode process) [20:31:11 CEST] <ubitux> like, the only place where do_{audio,video}_out is called is in reap_filters() [20:31:26 CEST] <ubitux> and the only way these functions are called is if there is a filter [20:34:17 CEST] <nevcairiel> probably yes then? =p [20:34:24 CEST] <nevcairiel> i dont know the dataflow yet [20:35:10 CEST] <ubitux> i wonder if i'll need to add a dummy lavfi subtitles support in order to keep things clean in ffmpeg :P [20:35:11 CEST] <BtbN> there should allways be an auto-inserted scaler and stuff [20:39:15 CEST] <jamrial> nevcairiel: yeah, since they didn't reply lets just revert things back to libav state so qsv merges become conflict free [20:39:38 CEST] <jamrial> once they show up again, they can reimplement their custom changes [20:40:34 CEST] <jamrial> this has blocked further merges long enough [20:47:30 CEST] <wm4> their boss can pay them extra to redo the work [20:48:19 CEST] <JEEB> esp. if it's just the MJPEG stuff :V [20:54:39 CEST] <nevcairiel> mjpeg and some options in the encoder [20:54:48 CEST] <nevcairiel> and maybe the CC stuff [21:06:55 CEST] <JEEB> right [21:07:08 CEST] <JEEB> still probably worth it to just override and re-apply [22:22:10 CEST] <cone-168> ffmpeg 03Muhammad Faiz 07master:1a9513bfbc60: avfilter/firequalizer: add scale option [23:25:18 CEST] <wbs> fyi, the patch recently about "support rtsps"; support for rtsps already exists since 2 years or so, this just tweaks it. kinda relevant for the commit naming [00:00:00 CEST] --- Mon Oct 17 2016

1 0

ffmpeg.log.20161016
by burek 17 Oct '16

17 Oct '16

[00:00:05 CEST] <Ana_> Let me try it, hold on! Thank you so much! [00:00:08 CEST] <c_14> (largest resolution, highest audio bitrate) afair [00:01:18 CEST] <Ana_> I just want to confirm, I should run this? ffmpeg -i input.mkv -c copy output.mp4 [00:02:10 CEST] <DHE> yeah. I mean, worst case output.mp4 isn't what you want and you fix it [00:02:28 CEST] <Ana_> What do you mean "isn't what I want"? [00:03:07 CEST] <DHE> if you do have multiple audio or video streams, maybe they don't all make it in or not in the order you want, etc. check it when it's done and make sure it's what you want [00:04:11 CEST] <Ana_> Okay, I will! When I open files that I've re-encoded on FFMPEG into Sony Vegas there are always like four or five audio tracks, so that might be my problem. This code will fix that? [00:05:08 CEST] <DHE> no. but that's interesting. is that one audio stream per track? eg: surround sound presented as 5 or 6 mono tracks? [00:07:49 CEST] <Ana_> Uh I'm not sure. Sorry I'm not very good with all this but when I open the file, it has one track for video, and then audio is split into a bunch of different audio tracks. Some sound-effects and stuff were isolated and dialogue was on a separate track? Maybe they were 5 or 6 mono tracks, I had never considered that. [00:08:51 CEST] <DHE> when you run ffmpeg there's a little info dump on all the tracks. streams are named 0:0, 0:1, 0:2, ... [00:09:26 CEST] <DHE> and just above the progress report is a table showing how input and output streams are mapped [00:10:19 CEST] <Ana_> Actually I had to same problem when I used Wondershare video converter (wasted $50 for it to just double everything in size too) and when I opened it the converted MP4 in Vegas I got split audio tracks too, so apparently it's not only a changing-container issue it happens with conversion too? [00:10:22 CEST] <kerio> Ana_: what made this file? [00:10:39 CEST] <kerio> fam this doesn't happen by magic [00:10:48 CEST] <SchrodingersScat> tell us who did this to you [00:11:00 CEST] <DHE> Ana_: run that ffmpeg command I gave you, and pastebin the entire output of the ffmpeg run when it's done [00:11:00 CEST] <kerio> if the original mkv has multiple audio tracks you're gonna get multiple audio tracks [00:11:46 CEST] <furq> not without -map 0 [00:12:30 CEST] <Ana_> What do you mean? It's a torrented MKV file haha all I want to do is make shitty fanvideos of The Walking Dead [00:12:43 CEST] <Ana_> DHE okay!! [00:14:12 CEST] <kerio> surely that means that you have multiple audio tracks [00:14:25 CEST] <DHE> I still want the output to be sure [00:14:39 CEST] <Ana_> But why is the original file size 500MB and the output is 1GB? [00:14:45 CEST] <furq> because you're reencoding the video [00:14:47 CEST] <Ana_> Okay, one second! Thank you! [00:14:57 CEST] <DHE> yeah, my command shouldn't take more than 10 seconds to run [00:15:25 CEST] <furq> also yeah if you don't specify -map and the output has more than one audio track then something weird is happening [00:15:56 CEST] <kerio> are you sure you're not confusing multiple audio tracks with multiple audio channels? [00:16:33 CEST] <furq> maybe they show up separately in vegas or something [00:17:05 CEST] <Ana_> That might be possible! [00:17:10 CEST] <Ana_> I don't know [00:17:12 CEST] <DHE> let's see what ffmpeg actually says [00:17:20 CEST] <furq> ^ [00:17:32 CEST] <Ana_> Okay I ran it http://pastebin.com/nFqURBZe [00:17:48 CEST] <Ana_> Sorry it took me awhile to start it up [00:18:01 CEST] <furq> oh [00:18:02 CEST] <furq> hevc [00:18:10 CEST] <furq> well that's probably going to be an issue [00:18:31 CEST] <kerio> why [00:18:38 CEST] <furq> i doubt that vegas will import that [00:18:41 CEST] <kerio> anyway yea that's 5.1 audio [00:18:56 CEST] <kerio> Ana_: did sony vegas like the video [00:19:35 CEST] <Ana_> Yeah Vegas opens it after I re-encode it to MP4 [00:19:43 CEST] <Ana_> that part works fine [00:19:54 CEST] <kerio> no, the new one [00:20:01 CEST] <kerio> the one that's the same size and was not reencoded [00:20:22 CEST] <kerio> furq: honestly who even encodes pirate tv series in h265 [00:20:25 CEST] <furq> idiots [00:20:32 CEST] <kerio> i mean, it's better, but [00:20:39 CEST] <Ana_> EVERYONE! I can't find any in h264 jkfdjnklgl [00:20:50 CEST] <furq> usually idiots who don't understand that x265 is worse than x264 unless you use incredibly slow settings [00:21:05 CEST] <Ana_> I double-checked in Vegas and [00:21:08 CEST] <furq> and just encode with preset medium at half the bitrate they would use for x264 and assume it'll look great [00:21:12 CEST] <kerio> D: [00:21:16 CEST] <furq> hint: it does not look great [00:21:27 CEST] <Ana_> as usual with MKV files, they don't even show up in the file explorer when I try to open them in Vegas [00:21:42 CEST] <furq> Ana_: the mp4 that you just made [00:21:42 CEST] <kerio> Ana_: yes, vegas does not understand the matroska container [00:21:46 CEST] <kerio> what about the mp4 you just made? [00:21:52 CEST] <Ana_> That works fine [00:22:00 CEST] <furq> never mind then [00:22:07 CEST] <kerio> are you sure it did? [00:22:34 CEST] <kerio> because everyone on the internet is saying that hevc is not supported in sony vegas [00:22:37 CEST] <DHE> h265 has potential for quality, but the encoders are kinda crap. and the codec itself has some unpleasant licensing behind it... [00:22:55 CEST] <furq> if vegas uses directshow codecs then it'll be able to import hevc using lav filters [00:23:00 CEST] <furq> or ffdshow if it uses vfw [00:23:11 CEST] <furq> it'll probably be a pig to edit but it should actually work [00:24:17 CEST] <Ana_> One second [00:24:34 CEST] <Ana_> WHOA HOLY SHIT [00:25:27 CEST] <Ana_> I though FFMPEG wans't working or just stopped because it wasn't doing anything so I thought it had stopped [00:25:27 CEST] <furq> well that's a cliffhanger [00:25:41 CEST] <Ana_> But no I'm just an idiot it really did just take 15 secs [00:25:45 CEST] <furq> oh [00:25:52 CEST] <Ana_> I'm used to re-encoding taking over an hour oh my god [00:26:08 CEST] <furq> you're remuxing, not reencoding [00:26:08 CEST] <Ana_> So I didn't actually see the new MP4 until now [00:26:15 CEST] <furq> it just copies the streams into a new container [00:26:16 CEST] <Ana_> Okay let me open it in Vegas [00:26:25 CEST] <Ana_> The file size is the exact same oh my god [00:26:40 CEST] <furq> i wouldn't get your hopes up too much just yet [00:26:59 CEST] <Ana_> yeah shit [00:27:19 CEST] <Ana_> It said Vegas couldn't open it [00:27:29 CEST] <Ana_> "most likely a corrupted file or unknown format" [00:27:33 CEST] <Ana_> well darn [00:28:36 CEST] <furq> there are free NLEs which support hevc [00:29:01 CEST] <furq> it looks like vegas uses vfw, and i'm not sure if ffdshow has an hevc decoder [00:30:20 CEST] <Ana_> I'm not sure how people all over the internet do this then because most people who edit use Vegas. Like I've only ever come across one torrent that contained MP4's and I successfully convert the AVIs [00:30:35 CEST] <furq> tbh the fastest solution might be to download some x264 rips instead [00:30:49 CEST] <Ana_> wdym [00:31:05 CEST] <furq> your rips are using hevc (x265) [00:31:19 CEST] <furq> which is new and not widely supported, as you just found out [00:31:20 CEST] <Ana_> and seriously I appreciate you guys taking the time to help me I've been trying to figure out how to do this for like 6 months [00:31:27 CEST] <Ana_> Ohh sorry I misread that [00:31:33 CEST] <Ana_> Yeah maybe [00:31:48 CEST] <Ana_> Do you think that would help the file size problem? [00:31:55 CEST] <furq> you can either reencode those which will take ages and lose quality, or just find some x264 rips [00:32:10 CEST] <furq> they'll probably be a bit bigger [00:32:22 CEST] <furq> also i should point out that i do not condone piracy. i think it's bad and not good [00:32:23 CEST] <Ana_> oh [00:32:42 CEST] <furq> with that out of the way, you should download it [00:33:31 CEST] <Ana_> I agree I'm just a highschooler who likes video editing trust me if buying the DVDs and ripping them work I would gladly pay the money [00:33:48 CEST] <SchrodingersScat> people can file complaints at whogivesashit.com [00:33:54 CEST] <furq> i don't actually care [00:34:31 CEST] <furq> we've had people in here asking how to rip MXFs of movies that weren't coming out for another two days [00:34:31 CEST] <SchrodingersScat> #YOLO #Thuglife [00:34:37 CEST] <Ana_> I don't understand -- why would I download x264 if the file size comes out even bigger? File size is my problem [00:35:02 CEST] <furq> because then you won't have to spend hours reencoding it and losing quality to be able to import it into vegas [00:35:45 CEST] <furq> you should just be able to convert them to mp4 and it'll work fine [00:36:05 CEST] <Ana_> so you're saying either way I'm gonna be stuck with 1GB per episode either way [00:36:52 CEST] <c_14> Either you find rips that are less that 1GB per episode, or you make them less than 1GB per episode by using a sledgehammer. [00:37:06 CEST] <c_14> (If you reeeally want them to be less than 1GB per episode) [00:37:17 CEST] <furq> for SD they should be about 400MB per episode [00:37:21 CEST] <Ana_> the rips are like 500MB but after conversion it's 1GB that's the problem [00:38:10 CEST] <c_14> Ana_: use a different preset, use a lower crf, use a fixed bitrate. But in your case it'll be easier to just download it in a format you can use directly because you'll save cpu time (and irl time) [00:38:39 CEST] <furq> i take it the problem is limited disk space [00:39:18 CEST] <furq> also if these are HD then 500MB is going to look bad [00:39:43 CEST] <Ana_> I have 192 GB but 16 eps X 6 seasons = 96GB [00:40:05 CEST] <Ana_> c14 but won't I still have to convert to MP4 when I get the h264? [00:40:27 CEST] <furq> "convert" meaning remux, not reencode [00:40:42 CEST] <furq> remuxing (-c copy) is just copying the video streams into a new container [00:40:45 CEST] <Ana_> ohh so youre saying the remux will work on h264? [00:40:48 CEST] <furq> yeah [00:40:56 CEST] <Ana_> and vegas will open that? [00:40:59 CEST] <furq> should do [00:41:08 CEST] <Ana_> ohh okay [00:41:25 CEST] <Ana_> alright I get it now thank you [00:42:10 CEST] <furq> apparently series 3 in 480p is 6.7GB, in 720p it's 23GB [00:42:24 CEST] <furq> so if 480p is ok then it should be fine [00:42:45 CEST] <furq> you can always upscale it in vegas if it's going on youtube [00:42:47 CEST] <DHE> sounds about right. 720p is roughly 3x as many pixels as 480p [00:42:49 CEST] <DHE> (very roughly) [00:42:56 CEST] <Ana_> holy shit [00:43:34 CEST] <Ana_> yeah it's weird though that I have a torrent of season 5 that was all originally mp4 and it's great quality and 500MB each ep [00:43:56 CEST] <furq> is that 720p [00:44:38 CEST] <Ana_> sorry I'm an idiot how do I check that [00:45:21 CEST] <furq> nvm it's in that pastebin [00:45:48 CEST] <furq> Stream #0:0: Video: hevc (Main), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 23.98 tbc (default) [00:46:01 CEST] <furq> if this is 500MB for 43 minutes then there's some kind of magic going on if it looks good [00:46:17 CEST] <BtbN> for hevc that's absolutely possible. [00:47:19 CEST] <furq> the 720p x264 rips i'm seeing are 3x that size [00:48:13 CEST] <BtbN> depends on the content, but with a good hevc encoder that's more than possible to look decent [00:49:24 CEST] <Ana_> Okay I found an h264 I'm torrenting it now [00:51:35 CEST] <Ana_> No peers awesome [00:52:54 CEST] <Ana_> Would an AVC codec work? [00:53:03 CEST] <furq> avc is the same as h264 [00:53:13 CEST] <Ana_> oh seriously [00:53:15 CEST] <Ana_> jdsjndjsnkfdhj [00:53:16 CEST] <furq> you probably want to search for x264 though [01:03:27 CEST] <Ana_> Here was the description of the torrent I just got, when it's seeding I'll try remuxing [01:03:27 CEST] <Ana_> http://pastebin.com/49BmAwRU [01:03:40 CEST] <Ana_> Thank you guys so much [01:25:47 CEST] <Ana_> So I remuxed the torrent and it still won't open in Vegas. The files that were torrented were AVI (which doesn't work with Vegas either) but the codec was AVC (you can see in the pastebin above) [01:26:10 CEST] <DHE> so use the same ffmpeg command to remux them to mp4 [01:28:11 CEST] <ChocolateArmpits> Yeah NLEs are very picky if you stray away from broadcast formats [01:29:53 CEST] <Ana_> i did remux them to mp4 from avi [01:30:02 CEST] <Ana_> and Vegas still doesn't open them [01:31:05 CEST] <furq> apparently you should be able to install them if you install ffdshow [01:31:12 CEST] <furq> s/install them/open them/ [01:32:20 CEST] <furq> http://ffdshow-tryout.sourceforge.net/download.php [01:32:22 CEST] <furq> i guess that's the one you want [01:32:58 CEST] <Ana_> I can open them on my computer but not Vegas, would that affect Vegas? [01:33:11 CEST] <furq> i don't know for sure but apparently vegas uses vfw codecs [01:33:15 CEST] <furq> if it does then that should work [01:34:15 CEST] <SchrodingersScat> have we considered not using vegas? [01:34:18 CEST] <furq> if you're not already invested in vegas then you could use shotcut or something [01:34:29 CEST] <furq> that just uses ffmpeg for decoding so it'll support everything ffmpeg does [01:35:23 CEST] <SchrodingersScat> maybe we can use something that doesn't make me want to rip out my own eyes [01:35:34 CEST] <furq> that would exclude all NLEs [01:36:11 CEST] <Ana_> downloaded ffdshow, didn't work [01:36:19 CEST] <furq> did you restart vegas [01:36:23 CEST] <Ana_> yes [01:37:33 CEST] <furq> are vegas and ffdshow both 32-bit or 64-bit [01:37:50 CEST] <furq> they both have to be the same or else it won't work [01:40:12 CEST] <furq> ctrl-shift-esc, rightclick vegas in the list > go to process, if it has *32 after the name then it's 32-bit [01:44:44 CEST] <Ana_> there's no 32 bit OR 64 bit [01:49:13 CEST] <furq> ? [01:49:25 CEST] <furq> if you mean in task manager then if it doesn't say *32 then it's 64-bit [01:50:10 CEST] <kerio> check if you can watch the video in mpc-hc [01:53:03 CEST] <furq> mpc-hc doesn't use ffdshow any more [01:55:29 CEST] <kerio> :( [01:56:19 CEST] <Ana_> yes I can open with Windows Media Player [02:15:55 CEST] <Madd_the_Sane> Is there a reason why swresample is linked against SDL? [02:20:53 CEST] <Madd_the_Sane> Perhaps a better question is why is it linking against SDL (and other frameworks) when it doesnt call function from those libraries on OS X. [02:58:45 CEST] <torstein_> Could anyone help me with this issue? https://devtalk.nvidia.com/default/topic/971350/cuda-setup-and-installation… [03:30:53 CEST] <DHE> you sure your hardware suports HEVC decoding? [03:35:46 CEST] <furq> does cuvid use purevideo [03:35:53 CEST] <furq> if so then hevc isn't supported on the 7 series [03:51:15 CEST] <torstein_> DHE: No, but CUDA HEVC decoding works in Windows with MPC-HC and LAV. [03:52:27 CEST] <torstein_> CUDA should not rely on hardware codec support while CUVID should [04:16:12 CEST] <grepper> is it normal now for -target dvd to override any -vf scale options that occurs afterwards in the command? http://dpaste.com/29PSKDR.txt [04:20:58 CEST] <furq> -s appends -vf scale to the end of the filter chain iirc [04:21:03 CEST] <furq> so i assume -target does the same thing [04:21:18 CEST] <DHE> -target is just a whole bunch of presets, and I think -s is one of them [04:21:29 CEST] <DHE> wonder if you could do the parameters in a different order? [04:21:39 CEST] <furq> try using -s after -target [04:22:44 CEST] <furq> also check that it's not scaling twice with -v debug [04:22:55 CEST] <grepper> well, as the paste shows, I am using -vf scale after -target [04:22:58 CEST] <grepper> okay [04:23:34 CEST] <grepper> ffmpeg -i movie.vob -threads 4 -target ntsc-dvd -qmin 5 -qmax 31 -vf scale=w=352:h=368,pad=width=352:height=480:x=0:y=56,setdar=4/3,fps=30000/1001 -b:v 3600k -b:a 224k -ac 2 -map 0:1 -map 0:2 -y movie-halfdvd.mpg [04:23:46 CEST] <grepper> but its 720:480 in the output [04:25:08 CEST] <furq> if -target sets -s then it appends scale=720:480 to the end of that filterchain [04:25:17 CEST] <furq> so it's scaling to 352:368 and then to 720:480 [04:25:29 CEST] <furq> i assume if you set -s instead of -vf scale then it'll override that [04:26:22 CEST] <grepper> I see [04:26:25 CEST] <furq> i guess you'll need to adjust the pad to match [04:26:49 CEST] <grepper> used to work that anything after the -target option would modify it [04:29:15 CEST] <furq> https://github.com/FFmpeg/FFmpeg/blob/3e5e5bdfef07526aa5c6b15e0484bdc51289c… [04:29:41 CEST] <furq> i guess you could set all those manually instead [04:29:47 CEST] <furq> -s after -target should work though [04:30:48 CEST] <grepper> so use -s and also the -vf scale/pad option ? [04:31:18 CEST] <grepper> or just the pad option I guess [04:31:24 CEST] <grepper> with -s [04:31:45 CEST] <furq> use -s and then pad relative to 720:480 [04:32:17 CEST] <grepper> hm, okay, thanks [04:33:14 CEST] <furq> i just checked, that only scales once [04:39:26 CEST] <grepper> guess I could also just set the -target options myself and use the -vf scale [04:39:57 CEST] <grepper> -bufsize etc [10:10:46 CEST] <serp_> I have a program that generates output that I want streamed over the Internet [10:11:18 CEST] <serp_> the easiest for me would be to write jpeg files and have a separate program compress these and send [10:11:38 CEST] <serp_> I wonder... is motion jpeg literally concatinated jpeg images? [10:12:02 CEST] <serp_> so if I just append a bunch of jpeg images, is that a motion jpeg stream? [10:15:35 CEST] <kerio> afaik subsequent frames can skip some stuff [10:15:43 CEST] <kerio> but concatenated jpegs should work [10:21:50 CEST] <serp_> hmm could I encode a stream from a single image, re-reading the image from disk for each frame [10:22:43 CEST] <serp_> because I know that I will generate images in an uneven pace with a framerate lower than the encode framerate [10:23:02 CEST] <serp_> so I would like the same frame to be encoded multiple times when I miss aa frame [10:24:44 CEST] <kerio> i think you can just -loop 1 maybe? [10:24:49 CEST] <kerio> not sure if that rereads the image [10:24:54 CEST] <kerio> anyway, make sure to update the image atomically [10:25:17 CEST] <kerio> (write the new one with a .tmp extension or whatever then move it over the old one) [11:05:43 CEST] <serp_> kerio: yep -loop 1 works, it rereads the file [11:05:44 CEST] <serp_> thanks [11:06:46 CEST] <kerio> serp_: make sure to set a reasonable framerate [12:00:04 CEST] <laserbled> Hi. Is anyone developing with FFmpegInterop wrapper for windows ? [12:01:04 CEST] <laserbled> it is not able to stream a rtp. no error code or return. it just freezes. any help would be great. thanks [12:26:21 CEST] <pihpah> Any idea how to solve this error? 'Selected ratecontrol mode is not supported by the QSV runtime. Choose a different mode.' [12:26:50 CEST] <pihpah> I've compiled the latest ffmpeg from github but still having this error. [23:12:57 CEST] <emilsp> hello, I'm trying to compile a bit of C, I pass in "-I libswscale", #include<libswscale/swscale.h>, however I still get "src/c/ff.c:20:5: error: unknown type name SwsContext [23:17:59 CEST] <c_14> -I libswscale is wrong [23:18:15 CEST] <emilsp> what should it be ? [23:18:19 CEST] <emilsp> -lswscale ? [23:18:23 CEST] <c_14> You want -I /path/to/swscale/parent [23:18:39 CEST] <c_14> so -I /usr/include if the libswscale dir is in /usr/include [23:18:47 CEST] <emilsp> hmm, ok [23:18:53 CEST] <c_14> -lswscale is linking, that's a different issue [23:19:15 CEST] <c_14> (I hope your font can differentiate between l and I) [23:19:25 CEST] <emilsp> my font indeed can differentiate, yes [23:22:18 CEST] <emilsp> hmm, even with "-I /usr/include" it doesn't find it :( [23:22:38 CEST] <c_14> Well, where is the libswscale directory [23:22:41 CEST] <c_14> the include dir, that is [23:22:51 CEST] <emilsp> it's in /usr/include/libswscale [23:23:00 CEST] <emilsp> everything else is found that's remotely related to ffmpeg [23:23:11 CEST] <emilsp> by just using -I libavutil [23:24:31 CEST] <c_14> If that works, then it's either already in your standard include path, or you have an actual libavutil directory in your cwd [23:24:41 CEST] <emilsp> I most certainly do not [23:24:51 CEST] <emilsp> and yes, it should be in my standard include path [23:27:21 CEST] <c_14> emilsp: https://pb.c-14.de/t/kng.yjRDBh <- does this compile with `cc -lswscale test.c' [23:30:27 CEST] <TheLawyer> which command can I use to record my desktop screen changes? I don't care if it video or bunch of images but I want it be done automatically following the changes on the screen [23:31:20 CEST] <emilsp> hmm, so my header file for swscale has SwsContext defined as an empty struct [23:31:38 CEST] <emilsp> and yes, c_14, that does compile for me as well [23:32:03 CEST] <emilsp> does this mean that I have a wierd version of libswscale ? [23:32:27 CEST] <c_14> and if you have struct SwsContext *b; [23:32:33 CEST] <c_14> *add that [23:32:37 CEST] <c_14> on line 5 or so [23:32:54 CEST] <c_14> That earlier version was just to check if it can find the header at all [23:32:58 CEST] <TheLawyer> worthless people [23:34:00 CEST] <furq> what a nice young man [23:34:09 CEST] <emilsp> c_14 can't find it, seems like the struct is there, but the type isn't defined [23:34:44 CEST] <c_14> Because that builds with my version [23:35:00 CEST] <emilsp> archlinux's a shit [23:36:44 CEST] <emilsp> hmm, it seems I have two versions of ffmpeg installed, one is 1.3 and the other is 2.8 [23:38:08 CEST] <c_14> Well, 1.3 is a tad wee bit old [23:38:09 CEST] <emilsp> anyway, thanks c_14 for setting me on the right path, much appreciated [23:38:24 CEST] <emilsp> yet, firefox, chrome and just about everything but vlc depends on that packager [23:39:08 CEST] <furq> swscontext is defined in swscale_internal.h [23:39:28 CEST] <emilsp> I don't even have that header [23:39:40 CEST] <c_14> because it's internal [23:39:46 CEST] <c_14> It's declared in swscale.h though [23:40:35 CEST] <furq> oh never mind that's pretty much what you said [23:41:16 CEST] <furq> apparently vlc 2 on arch depends on ffmpeg 2.8 [23:41:51 CEST] <furq> actually i read that wrong didn't i [23:41:55 CEST] <furq> i'm doing well here [23:43:18 CEST] <emilsp> gah, which one would you suggest I build against ? [23:44:51 CEST] <furq> 2.8 [23:45:40 CEST] <furq> are you sure it's 1.3 and not 3.1 [23:45:51 CEST] <furq> apparently firefox depends on the main ffmpeg package [00:00:00 CEST] --- Mon Oct 17 2016

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