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October 2014

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ffmpeg-devel.log.20141030
by burek 31 Oct '14

31 Oct '14

[00:56] <cone-751> ffmpeg.git 03Carl Eugen Hoyos 07master:23ec8db8a074: Autodetect libxcb. [00:56] <cone-751> ffmpeg.git 03Carl Eugen Hoyos 07master:f05855414ed4: lavc/utils: Make pix_fmt desc pointer const. [01:51] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:32e666c354e4: avcodec/g2meet: check tile dimensions to avoid integer overflow [02:24] <cone-751> ffmpeg.git 03Martin Storsjö 07master:573b1de2d7f1: movenc: Don't use track_id to decide which track is the first in a moof [02:24] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:647405d1110c: Merge commit '573b1de2d7f1db71030f91ecdded7d0bc071f6b6' [02:36] <cone-751> ffmpeg.git 03Martin Storsjö 07master:b2b79eca6fae: movenc: Don't check the custom IO flag when using faststart [02:36] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:39cce77faf15: Merge commit 'b2b79eca6fae2466a53c5daa163a37e2474364fc' [02:44] <cone-751> ffmpeg.git 03Martin Storsjö 07master:1e0b81abe86d: movenc: Use a local variable consistently [02:44] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:ad600e10da6b: Merge commit '1e0b81abe86dc09dd34d60d57f92de5f12d65818' [03:17] <cone-751> ffmpeg.git 03Martin Storsjö 07master:a9911a1b2df3: doc: Document the omit_tfhd_offset flag to the mp4 muxer [03:17] <cone-751> ffmpeg.git 03Martin Storsjö 07master:600d5ee6b12b: movenc: Signal iso6 in compatible_brands when using tfdt [03:17] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:be787bccc00c: Merge commit 'a9911a1b2df37ca808e06dc30627215f4ec9a9e4' [03:17] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:7f24e1e1a2a9: Merge commit '600d5ee6b12bad144756b0772319bb04796bc528' [06:31] <jamrial> xcbgrab broke a couple fate slots [10:04] <cone-584> ffmpeg.git 03Carl Eugen Hoyos 07master:0e89d006d87a: Fix disabling libxcb-shm and libxcb-xfixes. [10:08] <cone-584> ffmpeg.git 03Carl Eugen Hoyos 07master:8c0ae9015a2b: Fix xcbgrab build if shm is not available. [10:27] <cone-584> ffmpeg.git 03Carl Eugen Hoyos 07master:792f0f20456c: Fix compilation of xcbgrab if xcb/shape.h is not available. [12:37] <cone-584> ffmpeg.git 03Tobias Rapp 07master:0c9f0da0f765: ffprobe.xsd: add packets_and_frames element definition [13:01] <saste> michaelni, reynaldo ping [13:28] <michaelni> saste, ? [13:28] <saste> michaelni, about OPW qualification ranking, when is it due? [13:28] <saste> October 31 or what? [13:29] <saste> I seem to remember definitive ranking was due in date November 7 or something [13:29] <saste> also, in the OPW "View applications" page what's the "rank" thing? [13:29] <michaelni> oct 31 for slots for slots from general funding, something later if you/we pay it [13:30] <michaelni> https://wiki.gnome.org/OutreachProgramForWomen/Admin/ApplicationReview [13:31] <saste> michaelni, can you quantify "something later"? [13:33] <saste> michaelni, is it OK to assume Friday 7 Nov? [13:34] <saste> reynaldo, ^^ [13:34] <michaelni> i see a mail that says nov7 [13:35] <saste> michaelni, confirmed then [13:35] <saste> michaelni, I'm writing a summary mail on ffmpeg-opw [13:36] <saste> another thing, the rank thing is left to organizations, so i suggest to leave it blank, and let the ffmpeg admins to fill it once we defined our internal ranking [13:36] <saste> so mentors won't have to wonder about how to fill it [13:50] <wm4> are good english skills part of the OPW requirements? [13:56] <saste> wm4, why? [13:57] <wm4> saste: because they make communication really hard [13:58] <BtbN> good english skills make communication hard? [14:00] <michaelni> good english is mandatory, yes [14:01] <michaelni> thats unless the mentor and student know a common language in which they can communicate very well [14:01] <saste> michaelni, good communication skills, English is a factor but that's not the only one [14:02] <michaelni> no but i see one or 2 applicants i have great difficulty understanding/communicating with due to their english skill [14:03] <michaelni> and if basic communication doesnt work theres IMO not much hope for the project [14:04] <BtbN> btw., https://github.com/01org/libyami is very interesting [14:04] <BtbN> It's from the intel vaapi guys. Aparently they think their own api is so horrible, they made an official wrapper for it [14:10] <nevcairiel> this is much more than a wrapper around vaapi [14:14] <wm4> lol [14:15] <wm4> more like Intel thought gstreamer is too bad [14:24] <Daemon404> moar frameworks [14:25] <kierank> intel have been emailing me trying to sell expensive frameworks [14:26] <kierank> https://software.intel.com/sites/default/files/managed/cb/77/Intel_Media_Se… [14:36] <michaelni> btw saste, you might want to help arwa debug that segfault, she seems to have a problem with it somehow [14:37] <kierank> __gb__: intel media server studio is just a wrapper around libva, right? [14:41] <michaelni> saste, that is iam thinking she needs help there as there a multiple iterations of the patch failing in apparently the same/similar way [14:48] <cone-584> ffmpeg.git 03Michael Niedermayer 07master:db442c873631: avdevice/xcbgrab: Fix offset support [15:10] <anshul_mahe> I was looking at http://wiki.multimedia.cx/index.php?title=FFmpeg_Summer_Of_Code_2008#Snow_J… [15:11] <anshul_mahe> I know its quite old, but I was I interested in snow codec and would like to work in it [15:12] <anshul_mahe> can any one tell me what part is already complete and what is still left [15:12] <anshul_mahe> like is there any document for "Document the whole Snow algorithm " [15:12] <av500> anshul_mahe: the task is to document it [15:12] <av500> afterwards, you have a document to read :) [15:15] <Compn> https://github.com/kasbah/gsoc/blob/master/ffmpeg/doc/snow.txt [15:15] <Compn> not sure if that has any other info from official snow.txt document [15:16] <anshul_mahe> thanks [15:18] <Daemon404> lolsnow [15:31] <reynaldo> saste: Whats michaelni said ^ [15:31] <reynaldo> we are targeting Oct 31 for having pretty muche veryone ranked [15:32] <wm4> why the fuck is snow still dragged along [15:33] <wm4> by now it's a definitive failure [15:33] <wm4> it's like GNU Hurd [15:33] <wm4> just as video codec [15:33] <wm4> yes there are people actively working on Hurd [15:33] <wm4> and they're getting revolutionary things done like basic audio playback (probably only for the driver writer's hw of course) [15:36] <anshul_mahe> wm4: actually I was doing this as part of MS programme, I wanted to work on some lossless video codec. [15:37] <anshul_mahe> Do you want me to suggest different codec other then snow [15:39] <anshul_mahe> do you have any other suggestion for lossless video codec where it is good exposure to algorithms. [15:39] <kierank> ffv1 [15:39] <kierank> j2k [15:39] <kierank> etc [15:40] <av500> tldr? [15:40] <av500> er, tl3dr [15:40] <anshul_mahe> ok, I will drop the idea snow then and look for ffv1 [16:11] <__gb__> kierank, there are ways to directly use the VA driver that ships with Intel Media Server Studio, or MediaSDK, or whatever is its name nowadays [16:11] <__gb__> however, some encoder features depend on additional MediaSDK logics [16:15] <wm4> i.e. undocumented API artifacts? [16:18] <__gb__> no, finer tuned encoder logics. The API is here: http://cgit.freedesktop.org/vaapi/libva/tree/va/vendor/intel?h=staging but MediaSDK will have encoding modes that won't make you hit your head :) [16:19] <__gb__> more detailed documentation of the various bits there live in the gpu docs [16:25] <kierank> __gb__: I'm guessing the audio stuff is just software then [16:28] <__gb__> yes, indeed [16:28] <__gb__> you wanted to offload stuff to the tensilica chips? :) [16:29] <kierank> just wanted to test out the aac encoder [16:29] <kierank> I am happier though that all the stuff is in libva [16:30] <kierank> since i care not for the higher level stuff [17:23] <cone-584> ffmpeg.git 03Michael Niedermayer 07master:10e32618acce: avcodec/cook: check that the subpacket sizes fit in block_align [18:38] <cone-584> ffmpeg.git 03Michael Niedermayer 07master:e91ba2efa949: avcodec/svq1dec: zero terminate embedded message before printing [19:43] <cone-584> ffmpeg.git 03Michael Niedermayer 07master:47f1596ecef3: avcodec/dxtory: use unaligned copy for unaligned addresses [21:09] <arwa> I am getting this error when I am running fate test " reference file './tests/ref/fate/filter-2xbr' not found", what reference file do I need to add? [21:11] <ubitux> add GEN=1 [21:11] <ubitux> for the first run [21:11] <ubitux> make fate-filter-2xbr GEN=1 [21:12] <arwa> okay [21:12] <arwa> thanks [21:13] <arwa> It is still giving the same error :/ [21:14] <arwa> now, it worked :D [21:14] <arwa> thanks [21:21] <ubitux> arwa: so did you compare the output with the reference? [21:26] <arwa> i got a reference file, but how do I compare it with the output? [21:27] <ubitux> no i mean [21:27] <ubitux> comparing with the reference implementation [21:27] <ubitux> not talking about fate here [21:27] <ubitux> like, are you sure you output the exact same thing? [21:29] <arwa> umm..I dont know actually...! How do [21:29] <arwa> I cross check it? [21:29] <ubitux> well, find the reference implementation and use it [21:29] <arwa> okay [21:30] <arwa> but the current implementations work with opengl [21:31] <ubitux> dunno, make a screenshot [21:31] <arwa> okay [00:00] --- Fri Oct 31 2014

1 0

ffmpeg.log.20141030
by burek 31 Oct '14

31 Oct '14

[02:22] <edman007> Hi, how do I change the sample rate of a an audio file? I got a 44.1kHz mp3 I'm trying to encode to a 48kHz ac3 file... is there an audio filter I need to run it through? [02:35] <edman007> oh, I guess -ar works [03:00] <edibile> it's not possible with a complex filter to avoid transcoding, right? [03:07] <edibile> i'm combining multiple videos into a matrix and trying to reduce cpu resources required but it doesn't appear there is a way to do this without transcoding [03:13] <relaxed> edibile: correct [03:20] <khaosmarmoset> hey. can I do with the ffmpeg binary everything that I can do with ffplay, ffserver, ffprobe? [03:21] <khaosmarmoset> I probably only will need ffmpeg, although I check stuff with ffprobe once in a while [08:51] <Baked_Cake> Baked_Cake_ (~Baked_Cak(a)c-24-20-70-164.hsd1.or.comcast.net) has disconnected: Ping timeout: 256 seconds [08:51] <Baked_Cake> o: [10:00] <wh-hw> hi, all [10:00] <wh-hw> what does this mean : value inf for parameter 'time_base' out of range ? [11:00] <khaosmarmoset> infinity [11:54] <khaosmarmoset> -f tee is giving me grief. [11:55] <khaosmarmoset> when I replace my -f flv out.flv with -f tee -map 0:a -map 0:v "[f=flv]out1.flv|[f=flv]out2.flv" I get a complaint about there being nothing to map. [11:55] <khaosmarmoset> the example with ffmpeg piping to another ffmpeg that uses -c copy works as advertised. [11:55] <khaosmarmoset> but why doesn't -f tee work out of the box? [12:06] <relaxed> khaosmarmoset: pastebin your command and output [12:16] <khaosmarmoset> http://pastebin.com/sg4JTCpN [12:18] <khaosmarmoset> ok I changed the -map 0:v to -map 1:v because of course my first input has no video, just like it says [12:23] Action: khaosmarmoset bestows relaxed the best honorary rubber duck award [12:24] <khaosmarmoset> multiple problems between chair and keyboard. [13:14] <relaxed> khaosmarmoset: it happens :) [13:46] <damnesia> Hi. For a yuv420p | i420 video, is it encoded as YYYYYYY (repeated width*height) UU (repeated w*h / 2) VV (repeated w*h/2) for every frame? [14:06] <termos> Is there a good and general way to increase the input buffer to FFmpeg, I need a way to make it more robust to laggy or unstable input streams [14:08] <khaosmarmoset> termos: not having read the docs at all, what I tried with some success was to add gstreamer in front of ffmpeg and use its queue element to buffer things up [14:09] <khaosmarmoset> termos: I'm sure there's an ffmpeg way to do this too. [14:09] <khaosmarmoset> termos: the gst way works if you like rube goldberg :) [14:09] <termos> ah thanks, unfortunately I won't be able to use the gstreamer way of doing it. Hmm [14:10] <termos> hehe [14:10] <termos> actually I am using the C api, so I guess I need to make my own kind of buffering system before the encoding step [14:12] <relaxed> termos: look at probesize and analyzeduration in the man page [14:16] <termos> I will look into those, I just found something that might work though [14:17] <termos> setting the rtmp_buffer value in AVDictionary and passing it to avformat_find_stream_info() [14:21] <termos> No, it did not work... [14:44] <termos> relaxed: do you know how I can set those using the C api by any chance? I can't figure it out, it's really weird [15:40] <Diogo> hi.... Command: ffmpeg -y -i "http://dl.dropboxusercontent.com/u/2561075/2014-10-30%2014.21.41.mov" -c:v libx264 -f mp4 result.mp4 [15:40] <Diogo> ---> I having a problem with mov attom and lots of HTTP 206 resquest. What is the solution for this probem. I want to move the attom to the begin of file and the source came from url. Because with other format mp4/avi/webm this command works very well [15:42] <kepstin-laptop> Diogo: if you're writing to a file, you can use the '-movflags faststart' option to get ffmpeg to move the moov atom to the beginning of the file. [15:42] <Diogo> but i can't download the file.. [15:42] <Diogo> :( [15:43] <Diogo> because i need to process the file very fast [15:43] <kepstin-laptop> oh, the problem's that you can't play the file off the dropbox url? [15:44] <Diogo> i want that this command works for any video.. [15:44] <Diogo> ffmpeg -y -i "<URL>" .... [15:45] <Diogo> this command works for any format except mov files.. mov files ffmpeg send lots of HTTP 206 request before start encode the video [15:45] <Diogo> i want to know why this happend...because with other formats ffmpeg only do 2/3 http 206 requests.. [15:46] <kepstin-laptop> well, unless the moov atom is at the start of the file, ffmpeg has to look through the file to find it before it can start playing [15:47] <kepstin-laptop> inherent problem in the mov/mp4 format. [15:47] <Diogo> can we save the moov atom in other storage and after that concatenate and pipe to ffmpeg? [15:47] <Diogo> ffprobe <URL> --> this command only do 2 http requests [15:48] <Diogo> the same mov file [15:48] <kepstin-laptop> well, ideally would be that you remux the original file (using ffmpeg with -c copy or the old qt-faststart tool) before uploading it, if you have control of that... [15:48] <Diogo> i can't change the original content :( [15:49] <Diogo> because its a user content if i change any part of file the md5sum change :( [15:49] <kepstin-laptop> well, then either live with it, or download the entire thing locally before encoding. [15:50] <Diogo> i need a pratical solution for this... 1º Http 206 to get moov atom 2º Concatenate in the begin of video 3º start encode.. [15:50] <Diogo> right? [15:51] <kepstin-laptop> the problem is finding the moov atom in the first place [15:51] <kepstin-laptop> btw, your example video has your gps coordinates in it. [15:54] <Diogo> sorry :( [15:58] <Fyr> how do I get mediainfo using ffmpeg? [16:01] <kepstin-laptop> Diogo: basically, to implement something like dropbox has, you have to have users upload the video directly to you so that you can do the fast-start processing on the whole file, possibly saving to a second file, before or while putting it into storage. [16:03] <Nadaesque> Hello. [16:07] <Nadaesque> I was wondering ... I have a source video that is about 1280x720 I would like to put on a DVD for my mom's ancient DVD player. I know I'll need dvdauthor later, but in terms of ffmpeg, is there anything I need besides "-target ntsc-dvd" in the conversion to get it to work, since the original aspect ratio is 16:9 and NTSC for DVDs is very much not 16:9? [16:11] <Mavrik> um [16:11] <Mavrik> NTSC for DVDs can easly be 16:9 :) [16:11] <kepstin-laptop> Nadaesque: NTSC on DVDs can be either 4:3 or 16:9, and ffmpeg will set up the aspect ratio in the stream correctly. [16:11] <Mavrik> target parameter should take care of these things [16:15] <Nadaesque> Really? Even ancient POS DVD players can handle it? [16:17] <Nadaesque> It'll just do the padding automatically? Or will it store it anamorphically? Will it to this only for 4:3 and 16:9, or can other aspect ratios deal with it? [16:17] <kepstin-laptop> Nadaesque: both 4:3 and 19:6 video on dvds are anamorphic [16:18] <kepstin-laptop> usually players have an option to switch between letterboxing and anamorphic video output for widescreen dvds on the analogue output [16:18] <Mavrik> yeah, 16:9 is something that predates DVDs, so all players must support 16:9 AR [16:19] <Nadaesque> I know that the TV pixels are tall rectangles rather than square, so the 720x480 turns into 640x480 when it hits a TV screen. [16:19] <kepstin-laptop> Nadaesque: technically not correct, particularly on crts [16:20] <Nadaesque> No? What is it then? [16:20] <kepstin-laptop> you're right, the pixels are tall rectangles. you're wrong that the resolution changes to 640x480 [16:20] <Mavrik> 16:9 CRTs and other TVs just stretched the NTSC input on 16:9 [16:21] <Nadaesque> I'm just, uh, talking about how you would draw it out with a piece of graph paper on a TV CRT. [16:21] <Nadaesque> Hrm, but other aspect ratios besides 4:3 and 16:9 will require the expression evaluator to perform padding magic in the video filter? [16:21] <kepstin-laptop> since computer monitors have square pixels, computer players often rescale dvd video to 640x480, but CRT tvs don't have pixels [16:22] <kepstin-laptop> remember that each line of video on a CRT is a continuous analogue voltage :) [16:22] <Nadaesque> Yeah. [16:22] <Nadaesque> <=== had to make an oscilloscope in college for his degree. GAH. [16:23] <kepstin-laptop> so the '720 pixels' in the video stream aren't really pixels, but are actually samples used to reconstruct the analogue signal. [16:24] <Nadaesque> So, things with weird aspect ratios will require extra work? [16:24] <kepstin-laptop> yes, things with weird aspect ratios will require extra work :) you'll have to pad or stretch them to either 4:3 or 16:9 [16:24] <Nadaesque> I feared as much. [16:25] <Nadaesque> We sometimes get requests to put something on a DVD so the uh ... less technically adapt professors can play it. [16:25] <Nadaesque> Erf, adept. [16:26] <kepstin-laptop> the real fun happens when you encode a video on dvd as 4:3, and then the viewer has a widescreen tv and sets the tv to stretch the signal (i.e. in 16:9 anamorphic mode) [16:26] <kepstin-laptop> then everything is just wide :( [16:27] <Nadaesque> Well, I will use my mom's thing as a trial run. I have shamed myself by trying to put this Lionel Richie concert she wanted to see on DVD. [16:27] <kepstin-laptop> (only real solution to that problem is to use an upscaling dvd player with HDTV resolution output, preferrably on HDMI) [16:28] <Nadaesque> Well, I want to try a 4:3 thing for the profs and let them just deal, because they don't know what HDMI is or anything. [16:28] <kepstin-laptop> and even then people don't like bars around their video, so they stretch it anyways :/ [16:28] <Nadaesque> I once had this professor, she got an AV cart and plugged the power strip into ... itself. [16:28] <Nadaesque> She didn't understand why it wasn't working. [16:28] <kepstin-laptop> and then you're all http://i5.minus.com/i6DRQ32FboEix.jpg [16:29] <Nadaesque> So, ffmpeg -> dvdauthor -> something to make an .iso -> burnination. [16:31] <kepstin-laptop> hmm, you can't use "just anything" to make the iso; it has to be UDF with possibly some other constraints [16:31] <kepstin-laptop> many iso creation tools do have an option to put them into a dvd-specific mode, tho [16:31] <kepstin-laptop> dvd-video-specific [16:32] <Nadaesque> Yeah, I'll have to find something for that, too. [16:32] <Nadaesque> I knew there was some filesystem funkiness for them and Joliet was probably =NOT IT=. [16:33] <kepstin-laptop> on mkisofs, you just have to use the "-dvd-video" option, I think [16:33] <Nadaesque> Aaah, that sounds good. [16:33] <kepstin-laptop> (read the man page for details, there's some funkiness about case sensitivity) [16:33] <flavioberetti> hello. i need to stream video in (near) realtime between to network hosts with a fast local connection. i would like to stream in raw format, without the time to de/encode before transmission. however if i simply use rawvideo with mpegts for example, i get 'stream is not seekable' because there is no information about frames. [16:33] <flavioberetti> is there a codec, or any way to embed uncompressed video in a network stream? [16:33] <Nadaesque> I guess, since DVDs are MPEG-2 TS, there's no access to (or reason to do) two pass encoding? [16:34] <kepstin-laptop> Nadaesque: container format is irrelevant to 2-pass encoding [16:34] <Nadaesque> So I could do 2-pass for a higher quality? [16:34] <kepstin-laptop> 2-pass encoding is a way to get the maximum quality possible overall for a given desired video bitrate [16:34] <Nadaesque> I thought it was only for H.264 stuff. [16:35] <kepstin-laptop> the concept is unrelated to any particular codec, it's just a method of improving bitrate allocation. [16:35] <Nadaesque> Hunh. I guess I have only seen references to it in conjunction with H.264, then. [16:36] <Nadaesque> Is it just ... overkill for DVD-Video? [16:36] <kepstin-laptop> it only really picked up when people started doing mpeg4-asp (e.g. divx,xvid) video [16:37] <kepstin-laptop> Nadaesque: I wouldn't say so; if you select the biggest bitrate that's dvd-spec complient and allows the video to fit on the disc, 2-pass encoding can let the encoder move bits to places where it'll help quality [16:37] <kepstin-laptop> I *think* the ffmpeg mpeg2 encoder can do 2pass, but I haven't actually checked... [16:38] <kepstin-laptop> yes, looks like it does [16:39] <Nadaesque> I sort of want to develop this into a more serious workflow so we can do this in a more automated fashion with high-quality results, but I can sometimes be Mr. Overkill about things. [16:44] <kepstin-laptop> note that you'll want to select between the "film-dvd" and "ntsc-dvd" targets depending on whether your input is a movie/animation/cinematic-style tv show (24fps) or standard tv (30fps) [16:45] <kepstin-laptop> although I think the only difference between the two is the selection of output framerate [16:46] <Nadaesque> Hrm. I should get a variety of samples and work up a list of potential questions. [16:47] <Nadaesque> Sometimes we get H.264s clearly ganked off of a PAL DVD via Handbrake. [16:47] <Nadaesque> And we get a lot of weeeeeird items in the mix. [16:47] <kepstin-laptop> oh, that'll be fun to deal with. and by fun I mean the complete opposite [16:48] Action: Nadaesque shuffles to the counter and looks up through foggy lenses. "I need this Laserdisc and this Betamax put on DVD for a class?" [16:48] <kepstin-laptop> to encode pal video to ntsc, you need to encode interlaced with a 2:2:2:2:3 pulldown applied, I think :/ [16:48] <kepstin-laptop> (i.e. every 10 fields is turned into 11 fields, to turn the 50 fields/s video into 60 fields/s) [16:49] <Nadaesque> That makes sense. Of course, who knows what kind of mangling the file has had before it reached our hands? [16:50] <kepstin-laptop> if it was a pal encode of a US movie, it was probably sped up from 24 to 25fps first (this often causes noticable audio pitch changes on older stuff) [16:50] <Nadaesque> Oh, yeah, I hate that. [16:50] <Nadaesque> I had a really sweet multi-region DVD player once upon a time with a pitch-shifter in it. [16:50] <Nadaesque> Because that pitch shift was like getting tased in the eardrum. [16:52] Action: kepstin-laptop just tried encoding a random test video in 2-pass mpeg2 video at 8Mbit/s, and was politely told by ffmpeg that "Using all of requested bitrate is not necessary for this video with these parameters." [16:53] <Nadaesque> Heh! [16:53] <kepstin-laptop> (the final output ended up being around 2.5Mbps) [16:53] <relaxed> kepstin-laptop: ffmpeg knows best [16:56] <relaxed> kepstin-laptop: did you try forcing it with -minrate ? [16:57] <Nadaesque> Well, thank you very much for this advice. I will probably return for more refinements as I run across ever-more-unlikely scenarios. [16:57] <kepstin-laptop> why? It's not like I actually need the bitrate to be 8mbit if the quality is fine :) [17:00] <Nadaesque> Must use all available bits. Yussss. [17:05] <eristisk> My H.264 video streams from the Canon 5D Mark III play perfectly fine with ffplay, but VLC and Mplayer give lots of distorted/pixelated frames on Debian [17:06] <eristisk> Mplayer on the terminal complains with many errors "Error while decoding frame!" [17:07] <kepstin-laptop> which debian version? Note that for a while, they've been compiling stuff to use libav rather than ffmpeg. [17:07] <c_14> Tried with the VLC nightly or a decent video player like MPV? [17:07] <eristisk> I am using the Intel GPU on my Ivy Bridge CPU, but there is also an Nvidia card on the system which I normally do not run. [17:08] <eristisk> c_14: VLC nightly I have not tried. I thought it was a VLC error until I noticed the same problem in Mplayer. [17:08] <eristisk> kepstin-laptop: Debian sid... there is weird movements to switch back to ffmpeg, and I think I somehow have both libraries installed, but I am not sure which VLC is using [17:08] <eristisk> I am installing MPV right now [17:09] <eristisk> MPV gives the same pixelated/distorted frames and Error while decoding frame! errors on the terminal [17:09] <c_14> Is the ffplay you're using system or manually compiled? [17:10] <kepstin-laptop> eristisk: just out of curiosity, check `ldd` output on your ffplay and mpv and see which libraries they're using [17:10] <c_14> And what version is it? [17:10] <eristisk> Installed with APT, not compiled. [17:10] <eristisk> ffplay version 2.4.2-1 Copyright (c) 2003-2014 the FFmpeg developers [17:10] <eristisk> built on Oct 6 2014 19:13:39 with gcc 4.9.1 (Debian 4.9.1-14) [17:11] <kepstin-laptop> looks like your ffplay is ffmpeg, but I bet the other players are using libav. [17:12] <eristisk> mpv is using libavcodec.so.56 but ffplay is using libavformat-ffmpeg.so.56 [17:12] <eristisk> kepstin-laptop: Right, seems that way. [17:12] <eristisk> That is evil, how do I kill libav? Can I make VLC use ffmpeg? [17:12] <kepstin-laptop> you have to recompile it. [17:13] <eristisk> Recompile VLC? Damn... [17:13] <kepstin-laptop> well, you might not, but it would be highly recommended to avoid possible incompatibilities :) [17:16] <kepstin-laptop> oh, hey, ffmpeg's made it into debian unstable now, it's not in experimental any more. [17:16] <rcombs> wow, that was fast [17:17] <eristisk> Yeah I know ffmpeg is in unstable, I have it installed. I just want a VLC compiled against ffmpeg libraries now :/ [17:17] <eristisk> This experience makes me pissed off at libav [17:18] <kepstin-laptop> eristisk: you could file a bug with debian saying that vlc fails to play some files, noting that it might be fixed by recompiling with ffmpeg; on the other hand, that might also start up a new flamewar thread ;) [17:19] <relaxed> ffmpeg won't be in Jessie so I doubt that will happen. [17:19] <edibile> it's not possible with a complex filter to avoid transcoding, right? [17:19] <edibile> i'm combining multiple videos into a matrix as a single video stream and trying to reduce cpu resources required but it doesn't appear there is a way to do this without transcoding [17:20] <c_14> edibile: you are correct [17:20] <relaxed> correct [17:20] <edibile> are there any options i should look at to reduce cpu load? this is for streams to iphones, so resolution/audio quality aren't of great concern. [17:21] <edibile> (finding a way to play udp streams on an iphone is another matter...) [17:21] <c_14> edibile: it would help if we had your current commandline [17:21] <edibile> sure, just a second [17:22] <kepstin-laptop> edibile: if you're doing h264 encoding with x264, just change the 'preset' value to a faster setting. [17:24] <edibile> http://pastebin.com/E5kGykCe [17:24] <edibile> ok, will look at the preset option [17:25] <c_14> yeah, preset will be your best bet, a higher crf might also help [17:27] <kepstin-laptop> edibile: preset defaults to 'medium', try changing it to 'fast' or 'veryfast' [17:27] <relaxed> it's already baseline [17:28] <relaxed> and can an iPhone decode mp2 audio? [17:29] <eristisk> When I run apt build-dep vlc then it does not try to install library development headers for ffmpeg libraries. Only -dev packages for libav stuff... [17:30] <relaxed> eristisk: I highly recommend https://github.com/mpv-player/mpv-build [17:32] <eristisk> relaxed: MPV is by default compiled to libav libraries as well... but these sources will compile against ffmpeg on Debian? [17:32] <kepstin-laptop> eristisk: you'll have to edit the packaging files for vlc to install the ffmpeg dev stuff, and you may need to do additional work to get it to use ffmpeg rather than libav (not entirely sure what that entails) [17:32] <eristisk> I am sure MPV is great, but I also like the GUI of VLC (sorry if that is somehow controversial...) [17:32] <relaxed> eristisk: correct [17:33] <edibile> i'm not sure on the iphone decoding mp2 audio. i still haven't found a reliable, free app to stream the udp video [17:33] <relaxed> edibile: you most likely want aac for audio [17:33] <Mavrik> iPhone almost certanly doesn't support MP2 audio out of the box [17:33] <edibile> would reducing frame rate help as well? [17:33] <Mavrik> also streaming baseline video to iPhones is just a huge waste of quality [17:34] <kepstin-laptop> apparently it's required to support some really old models :/ [17:34] <edibile> i'm not sure what "baseline video" means [17:34] <edibile> and what the alternative is [17:35] <relaxed> edibile: http://en.wikipedia.org/wiki/H.264/MPEG-4_AVC#Profiles [17:35] <Mavrik> kepstin-laptop, pretty much anything still used out there supports Main [17:35] <Mavrik> and the quality difference is very noticable [17:35] <relaxed> 320x480 is tiny [17:36] <edibile> i might bump up the resolution a bit, but was using that for testing [17:45] <edibile> great, changing preset to veryfast and profile to main cut cpu usage in half [17:49] <relaxed> edibile: did you see https://trac.ffmpeg.org/wiki/Encode/H.264 ? [17:50] <edibile> yeah, i found that after you guys told me what i was looking for [17:55] <edibile> jus tnoticed the ios compatibility section at the bottom, though [18:13] <edibile> thanks guys, cpu down to 25% of what it was before [18:18] <Diogo> hi, this is possible to revert the movflags faststart option?? ffmpeg -y -i <FILE> -c copy -movflags faststart FILE.mov [18:18] <Diogo> i want revert to original file [18:19] <relaxed> you can omit it but all you'll get is the original streams. [18:32] <edibile> compand is the right filter to use to cut out audio below a specified threshold (squelch, to remove fan noise in this case) [18:32] <edibile> that was supposed to be a question... [18:33] <erikblankinship> following instructions on creating a thumbnail for every second of video in an mp4, I am finding my thumbnails slightly off than those I get with the _same_ video source using javascript/html5 or quicktime export. Suggestions? [18:33] <erikblankinship> following these instructions: https://trac.ffmpeg.org/wiki/Create%20a%20thumbnail%20image%20every%20X%20s… [18:40] <voip__> please help http://pastebin.com/PdmRAcBn [18:46] <kippi_> hey [18:48] <kippi_> Is there away to output if the audio level changes? I would like to plot this, I have seen volume detect however this only outputs at the end, I am looking for somthing that is realtime [18:54] <kaotiko> hi [19:01] <khaosmarmoset> hey. [19:01] <khaosmarmoset> someone mentioned syncing audio and video before as something that needed to be done [19:02] <khaosmarmoset> so at the time I was mostly not actually listening to the audio when I was live streaming and only listening to it when I was doing local tests [19:02] <khaosmarmoset> also I was doing it with a different computer. now I'm trying different computers, and it seems like the audio sync is horrible. [19:03] <voip__> relaxed, can you help me with [mpegts @ 0x316cae0] PES packet size mismatchme=00:31:33.14 bitrate=1005.2kbits/s [19:03] <voip__> Last message repeated 1 times [19:03] <voip__> [aac @ 0x317c7e0] Input buffer exhausted before END element found [19:03] <voip__> Error while decoding stream #0:0: Invalid data found when processing input [19:03] <voip__> [h264 @ 0x38b40e0] negative number of zero coeffs at 29 18 [19:03] <voip__> [h264 @ 0x38b40e0] error while decoding MB 29 18 [19:03] <voip__> http://pastebin.com/PdmRAcBn [19:03] <khaosmarmoset> I used to get only a few warnings like about Non-monotonous DTS in output stream 0:0; etc.etc. [19:04] <khaosmarmoset> but now I get a lot more and the sync seems to change [19:15] <khaosmarmoset> -async 1 or -vsync 1 are doing nothing for me. [19:50] <khaosmarmoset> my problem was that the audio was before the video [19:50] <khaosmarmoset> erh. after. it was lagging a lot [19:50] <khaosmarmoset> i'm not really sure what I'm doing, but before I added a negative -itsoffset to the audio it seemed like -vsync and -async were not doing very much [20:02] <khaosmarmoset> seems to work now. I think the negative offset might be helping the other syncs to do their thing [20:04] <khaosmarmoset> ok. next problem is that if I use -tee to spew to both youtube and the filesystem at the same time everything goes wrong [20:04] <llogan> the tee muxer? [20:04] <khaosmarmoset> right [20:05] <llogan> what do you mean by "everything goes wrong"? [20:05] <khaosmarmoset> I get many more Non-monotonous DTS warnings and the youtube stream is way more corrupted than either recording to a local file or sending to youtube directly [20:05] <khaosmarmoset> phone [20:08] <khaosmarmoset> back. [20:08] <khaosmarmoset> ok I guess I can do that. [20:08] <khaosmarmoset> one moment. [20:17] <khaosmarmoset> http://pastebin.com/uG2N8wH8 [20:18] <khaosmarmoset> so that's the output when I use -tee. when I watch the result on youtube I see a lot of grey areas in the image [20:18] <llogan> add -flags +global_header [20:18] <khaosmarmoset> where? [20:19] <llogan> as an output option [20:19] <khaosmarmoset> so that's before the -f tee, then right? [20:20] <khaosmarmoset> can you see the end of the paste as well? [20:20] <llogan> ffmpeg [global options] [input options] -i input [output options] output [20:22] <khaosmarmoset> ok. I added that [20:22] <khaosmarmoset> but I still get an awful lot of Non-monotonous DTS warnings [20:29] <khaosmarmoset> well, the stream isn't as corrupted as before so perhaps that's okay. the audio seems to be pretty much in sync even. [20:30] <khaosmarmoset> although I'm not sure if that's because of the magic -itsoffset value or because it lets -async and -vsync do their thing [20:31] <khaosmarmoset> regarding the last part of the pastebin, that's a command line I expected would work based on this: http://sonnati.wordpress.com/2011/08/30/ffmpeg-%E2%80%93-the-swiss-army-kni… [20:31] <khaosmarmoset> but I get the Protocol not found error you see in the paste [20:32] <khaosmarmoset> my plan was to separate upload to youtube from save to disk by a few dozen seconds so things wouldn't need to fit so tightly [20:33] <khaosmarmoset> especially in case -tee doesn't work on site using the laptops there. [20:34] <voip__> llogan, i need your help :) [20:49] <llogan> voip__: i don't have an answer this time. try the ffmpeg-user mailing list. [20:50] <voip__> ok [20:57] <khaosmarmoset> why can't I send the flv file directly but get the Protocol not found error? [20:58] <khaosmarmoset> afaict, I'm invoking ffmpeg exactly right there. [21:16] <kippi_> Is there away to output if the audio level changes? I would like to plot this, I have seen volume detect however this only outputs at the end, I am looking for somthing that is realtime [21:38] <Uber-Ich> can I use ffmpeg to convert .xcf to .png or is it limited to vidya only? [21:40] <khaosmarmoset> could you not use gimp for that? [21:41] <Uber-Ich> I could... I would just need to install xorg. [21:42] <Uber-Ich> I might just switch over to my arch box and use gimp instead. Thanks khaosmarmoset! [21:42] <khaosmarmoset> also try imagemagick. [21:43] <khaosmarmoset> probably easier to script convert than use batch gimp [21:43] <khaosmarmoset> he went away didn't he? [21:44] Action: khaosmarmoset considers turning on all the noise [21:55] <Mista-D> to just count "scene change" frames, what is a better method? -vf select='gt(scene\,0.4)' will output the frames, dont want to write them, just count. [21:58] <khaosmarmoset> I count specific kinds of frames using ffprobe_show_frames [21:58] <khaosmarmoset> erh. ffprobe -show_frames [21:58] <khaosmarmoset> then grep and pipe to wc [21:58] <kepstin-laptop> Mista-D: on an ffmpeg command, you can use as output "-c:v null -f null -" and it'll discard the video frames without encoding them and generate no output file. [21:59] <Mista-D> kepstin-laptop: how do I know how many were found? [21:59] <kepstin-laptop> hmm, that's a trickier question :) [22:02] <khaosmarmoset> *sigh* my Protocol not found error only happens on the streaming laptops, but not on my dev box [22:02] <khaosmarmoset> prolly missing some extra package [22:03] Action: kepstin-laptop got the codec wrong anyways [22:03] <khaosmarmoset> I used the same source and and exactly the same command line. [22:03] <llogan> Mista-D: just look at the console output. frame= 53 [22:03] <khaosmarmoset> it's the same binary to boo. [22:04] <khaosmarmoset> boot [22:04] <khaosmarmoset> not any of it makes any kind of sense, because the streaming laptops can stream to youtube just find if they are grabbing from the webcam. [22:09] <khaosmarmoset> aha [22:09] <khaosmarmoset> bah [22:09] <khaosmarmoset> my filename is cam-2014-10-30T23:22:24+0200.flv [22:09] <khaosmarmoset> it tries to parse cam-2014-10-30T23 as the transport protocol [22:09] <khaosmarmoset> on the dev laptop I was using /tmp/cam-... as the path [22:33] <BtbN> put a ./ in front of it? [22:38] <khaosmarmoset> yes [23:22] <kippi_msa> hey [23:39] <ArunC> Hi, is it possible to rotate AVFrame using sws_scale? [23:53] <katayana> how do I play playlists (streaming, m3u) with ffplay? [23:57] <WindozeNT> hi [23:58] <WindozeNT> I'm having problems with FFMPEG [00:00] --- Fri Oct 31 2014

1 0

ffmpeg-devel.log.20141029
by burek 30 Oct '14

30 Oct '14

[00:17] <amalia> How do I submit test results ?/// [00:23] <michaelni> amalia, (https://ffmpeg.org/fate.html) "When you have everything working properly you can create an SSH key pair and send the public key to the FATE server administrator who can be contacted at the email address fate-admin(a)ffmpeg.org." [00:24] <amalia> Hmmm [00:51] <amalia> michaelni, my problem is writing a config file which will be called from fate.sh and generate configure.log, compile.log test.log report and version [00:52] <amalia> I would greatly appreciate if you could give me a sample config file I could adapt for my system since I think will be necessary in submitting my fate results right? [00:54] <michaelni> amalia, tehers one on https://ffmpeg.org/fate.html (you can probably drop all the lines that are marked as optional which is 2/3 of the file) [00:55] <amalia> michaelni, alright thanks [01:07] <cehoyos> nevcairiel: For gcc, the default is -O0, --disable-optimizations for gcc removes -fomit-frame-pointer and leaves -O at the default (-O0) [01:22] <cehoyos> thardin: Can we get rid of the "Field dominance 0 support is not implemented"? I really believe we have enough samples... [01:28] <cone-231> ffmpeg.git 03Michael Niedermayer 07master:f3c0e0bf6f53: avcodec/dnxhddec: treat pix_fmt like width/height [02:08] <amalia_> michaelni, just finished running the fate tests on my Fedora 20 localhost, generated a public key pair and sent to the administrator for fate [02:08] <amalia_> I'm trying to connect with ssh to the fate server and looks like it's asking for a password? i'm I to kill the process at this process since I think I've enabled public authentication with the fate server for my system? [02:12] <michaelni> amalia_, just added the key [02:12] <michaelni> should work now without pw [02:12] <amalia_> ok [02:13] <amalia_> michaelni, still asking for password? [02:13] <amalia_> so what should i do [02:13] <michaelni> fate(a)fate.ffmpeg.org [02:13] <amalia_> michaelni, is that the password? [02:15] <michaelni> no its the username and server, you dont need a password unless theres a password on the ssh private key [02:15] <amalia_> no there's none, [02:16] <amalia_> michaelni, i'm a bit confused here I don't know if the private key i sent should correspond to the details in ~/.ssh/id_rsa? [02:16] <amalia_> because i generated a new key-pair and sent to you probably i'll need to use it some how to login [02:17] <michaelni> you did sent corerctly a public key [02:17] <michaelni> what does ssh fate(a)fate.ffmpeg.org say ? [02:17] <amalia_> michaelni, it says fate(a)fate.ffmpeg.org's password: [02:19] <michaelni> maybe ramiro has an idea, he is here [02:21] <amalia_> michaelni, ramiro looks like i created the key and stored in the a different directory from ~/.ssh/ so looks like i'll have to use the key to setup my identity in ~/.ssh/? [02:21] Action: amalia_ wondering how that's done [02:23] <michaelni> you could just create a seperate account for fate and put the key in its ~/.ssh [02:23] <cehoyos> Could a native speaker please read this: http://pastebin.com/EsMa9Y1e [02:23] <cehoyos> Is it just me or did they completely misunderstood the GPL? [02:25] <amalia_> well michaelni i just added the new authentication to ~/.ssh/ so it's working now [02:26] <amalia_> so my question is how to submit my result whether to uncomment the 'fate_recv' part of the config file and re run the tests? [02:30] <michaelni> when you run "tests/fate.sh /path/to/fate_config" and the ssh command is not commented out in the config then the data shoud get submitted, make sure you choose a unique slot name before [02:31] <amalia_> michaelni, well i ran it the first time with that part commented out [02:31] <amalia_> so I've successfully authenticated with fate [02:31] <amalia_> so I don't know if I'll have to rerun the same command again with that part uncommented or I run the example in 4.3 at the bottom of the wiki? [02:32] <kierank> cehoyos: yes misunderstanding the gpl [02:32] <kierank> as far as I can tell [02:32] Action: amalia_ seeing that running it again tries to fetch those code repos again [02:32] <cehoyos> This is the followup to Rhozet: [02:32] <cehoyos> Rhozet apparently removed FFmpeg from their product [02:33] <michaelni> 4.3 is for locally running fate [02:33] <cehoyos> A German company sells a plugin for Carbon Coder now [02:33] <cehoyos> This is an excerpt of their EULA. [02:33] <cehoyos> I have a suspicion that they charge a lot of money but I don't know yet. [02:35] <cehoyos> http://www.x-dream-media.com/index.php/de/produkte/harmonic-rhozet-carbon-f… [02:35] <amalia_> michaelni, Ok will rerun the fate.sh with the config script together with the uncommented part, was not really understanding why it still had to try to clone the code repos again [02:36] <michaelni> it shouldnt clone them new but just update them [02:36] <amalia_> ok thanks [02:37] <kierank> cehoyos: of course they charge a lot [02:47] <amalia_> michaelni, just ran the tests enabling submission of results to fate [02:47] <amalia_> but don't know how to determine if the process was successful or not [02:48] <michaelni> amalia_, look at fate.ffmpeg.org do you see the resuts = [02:48] <michaelni> ? [02:48] <amalia_> michaelni, ramiro http://paste.kde.org/puiodl6xo this is what i saw on my terminal [02:51] <amalia_> I am not seeing any thing on fate [02:53] <michaelni> it doesnt submit results if it was run withe the same revission last time, thats probably it [02:55] <amalia_> i keep on getting this same error: [02:55] <amalia_> [root@localhost ffmpeg]# tests/fate.sh /home/amalia/fate-config.sh [02:55] <amalia_> HEAD is now at f3c0e0b avcodec/dnxhddec: treat pix_fmt like width/height [02:55] <amalia_> [root@localhost ffmpeg]# [02:55] <j-b> root? [02:55] <amalia_> j-b, yeah [02:55] <cone-231> ffmpeg.git 03Carl Eugen Hoyos 07master:26122145559f: Do not set the lame quality if the user didn't request it. [02:55] <cone-231> ffmpeg.git 03Carl Eugen Hoyos 07master:19a6431ec247: Print a warning if a subtitle demuxer changes utf16 to utf8. [02:55] <cone-231> ffmpeg.git 03Andrey Utkin 07master:b608fba67265: Use v4l2 input format automatically if filename starts with "/dev/video" [02:55] <cone-231> ffmpeg.git 03Carl Eugen Hoyos 07master:238ed47faead: Mention in the documentation that fieldmatch needs cfr input. [02:55] <cone-231> ffmpeg.git 03Michael Niedermayer 07master:5b864470808a: Merge remote-tracking branch 'cehoyos/master' [02:55] <amalia_> cloned and did everything as root [02:56] <j-b> this is dangerous. [02:56] <j-b> You should not do that. [02:56] <michaelni> dont use root unless you have to and fate does not need root [02:57] <michaelni> also as i just pushed a few changes the next fate client run should/might work [02:59] Action: amalia_ is changing the permssions of ffmpeg_sources to normal user [03:00] <michaelni> amalia_, also make sure ~/.ssh contains the correct keys for the user under which fate runs [03:00] <amalia_> michaelni, already added the new key with ssh-add [03:24] <amalia_> michaelni, tried adding the generated key to ~/.ssh i get this error saying the key is unprotected. I'll like to generate another key and send for you to add to the list of hosts and delete the other one [03:28] <jamrial> use chmod on the key ssh complains about [03:29] <amalia_> jamrial, just decided to create a new key which I've added to ~/.ssh/ [03:29] <amalia_> trying to upload and send to fate@fate.... [03:30] <jamrial> there was no need for that. ssh was simply complaining about the key file permissions [03:31] <jamrial> "chmod 600 yourkey" is probably all you needed to do [03:37] <amalia_> did that jamrial and it worked [03:37] <amalia_> just trying to get the test authenticate with the fate server perfectly [03:37] Action: amalia_ feels sleepy already [03:47] <amalia_> jamrial, looks like the new key i generated seems to prevent the previously authenticated key from working even though I used ssh-add to add the public key to ~/.ssh [05:06] <cone-231> ffmpeg.git 03Michael Niedermayer 07master:c1e035ea89c1: avformat/mxfdec: fix null pointer dereference [08:26] <thardin> cehoyos: if they work then sure, I suppose [09:32] <rcombs> <sub2video go away plz> [09:46] <wm4> WHAT THE FLYING FUCK [09:46] <wm4> seriously what [09:47] <wm4> someone please explain this shit to me http://git.videolan.org/?p=ffmpeg.git;a=commit;h=19a6431ec247e4842236292cc5… [09:49] <thardin> the subtitle rendering uses utf-8 maybe? [09:49] <wm4> it's bullshit for ffmpeg.c [09:50] <rcombs> wm4: meant to inform the user that passing "utf-16" as the sub charenc won't work as expected? [09:50] <wm4> it "informs" every user, even players like mpv [09:50] <wm4> why the fuck would I want every user to see this useless message [09:51] <nevcairiel> plenty warnings get added for some dumb ffmpeg case, no matter that the libs are much more often used by other things than ffmpeg.c [09:51] <wm4> I know... and fuck this [09:51] <wm4> I'm considering completely disabling any log message from ffmpeg because it's so useless [09:51] <rcombs> same reason you'd want every user to hear about some struct's time_base member being deprecated as a hint (or whatever that one message is) [09:51] <rcombs> (one day I might find out what that reason is!) [09:53] <rcombs> does mpv ever actually hit that message? [09:54] <cehoyos> The message is only shown on encoding. [09:55] <cehoyos> thardin: What I meant was: Could you remove the message? Or do you believe it has some benefit? [09:55] <wm4> rcombs: why wouldn't it [09:55] <thardin> not sure since I didn't write that stuff [09:55] <wm4> cehoyos: it was definitely added to the decode path, not encode [09:55] <cehoyos> Ok, sorry [09:55] <thardin> I can take a look though [09:56] <cehoyos> Actually, I don't think you are right: Why would decoding ever trigger this message? [09:56] <thardin> well, if there's a zero in there [09:56] <thardin> which usually means "unknown" [09:57] <cehoyos> Yes, every second sample uplaoded by users contain "0" and ask the user to upload the sample. [09:57] <thardin> well, the field order can be parsed from the video essence [09:57] <rcombs> oh, actually, I don't think that commit does what it was meant to do [09:58] <thardin> I wonder what mediainfo says.. [10:00] <rcombs> cehoyos: I think it'd make more sense to put that message around https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/utils.c#L1665 [10:00] <cehoyos> How would libavcodec know that the original file (before "handled" by libavformat) contained utf-16? [10:01] <wm4> by not having a shit API? [10:01] <wm4> you don't know the same for matroska either [10:01] <rcombs> the point of the message is to tell the user not to unnecessarily send `-sub_charenc utf-16`, right? [10:01] <wm4> so you'd happily corrupt utf-8 if the user specified something other than utf-8 [10:01] <cehoyos> But please feel free to send a patch improving (or even reverting) my patch: From the discussion, I did not see that it was controversial at all. [10:01] <cehoyos> rcombs: Yes, you did see the ticket? [10:02] <rcombs> yeah [10:02] <rcombs> cehoyos: so, why not just warn if avctx->sub_charenc matches "utf-16" or similar, which IIUC is what the user should actually be warned about? [10:03] <rcombs> that's what I thought you were going to do, looking at the ticket [10:03] <wm4> yep, utf-16 packets should never happen [10:03] <cehoyos> I am surprised about this argument but please see above: Just send a patch to improve the situation. [10:04] <rcombs> I hope I don't sound argumentative; I'm just trying to understand the problem properly [10:04] <cehoyos> And if reducing the log level of the message, please feel free to do it: As said, I didn't see any controversy about the actual patch yesterday. [10:04] <ubitux> cehoyos: please fix 238ed47faeadf4ed0008da774cf61d6b224e4254 [10:04] <cehoyos> I removed the part that you were objecting to, no? [10:05] <ubitux> the filter doesn't work for cfr input [10:05] <ubitux> +only [10:05] <ubitux> i think we agreed on that [10:05] <ubitux> fieldmatch doesn't care about cfr or no [10:05] <ubitux> it's just field matching, and it works just fine with vfr [10:05] <rcombs> wm4: also, thoughts on how to handle a case where the file has UTF-16 without a BOM? [10:06] <ubitux> the issue lies into decimate [10:06] <ubitux> if any [10:06] <rcombs> (or, are those rare enough to ignore?) [10:06] <cehoyos> But the documentation states that you have to add decimate to fieldmatch or do I misremember? [10:06] <ubitux> maybe, still you're saying in fieldmatch something plently wrong [10:07] <cehoyos> ... it needs to be followed by decimation... [10:07] <ubitux> people can use mpdecimate of even a select or whatever [10:07] <cehoyos> Neither work. [10:07] <cehoyos> For real samples. [10:07] <ubitux> well, your text is wrong for fieldmatch, period [10:08] <ubitux> "The filter currently only works for constant frame rate input." this is not true [10:08] <cehoyos> Why did you write on the mailing list "that you don't care but that you don't understand why pullup works better"? [10:08] <cehoyos> Sorry, I tried to make the message clear to users: [10:08] <ubitux> it was before our discussion [10:08] <cehoyos> It dimply doesn't work. [10:08] <cehoyos> At the same time I didn't want to imply that anything is better or similar. [10:09] <ubitux> oh well whatever, i'm just saying your sentence is just wrong [10:09] <cehoyos> Yes, the discussion explained the technical issue better, but for users, this is what "fieldmatch" does (even if not true). [10:10] <ubitux> well feel free to mislead the users then :) [10:14] <cehoyos> ubitux: Patch sent, please comment [10:15] <ubitux> cehoyos: OK [10:17] <ubitux> you can apply [10:17] <cehoyos> Thank you! [10:19] <cone-751> ffmpeg.git 03Carl Eugen Hoyos 07master:fd1652b3ba6c: Improve the fieldmatch documentation about mixed telecined content. [10:46] <ubitux> cehoyos: regarding #3968, pullup is doing a better job? [10:47] <ubitux> if you can show me a working cli for pullup, and a non working one for fieldmatch+decimate, i'd like to know [10:49] <ubitux> (and if you have the sample somewhere...) [11:32] <cehoyos> ubitux: Yes, but definitely not perfect, it is dropping random frames. [11:32] <cehoyos> The command line from ticket #3968 is sufficient. [11:34] <cehoyos> http://samples.ffmpeg.org/ffmpeg-bugs/trac/ticket3968/ [12:26] <cone-751> ffmpeg.git 03Thomas Mundt 07master:2114e8843256: avformat/mxfenc: AVC Intra support [13:20] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:0c42f47e383f: avcodec/libutvideodec: Try to fix build failure with old libutvideo [14:44] <alexvf> hi, i've tried in #ffmpeg but got no response, hope i can get a response here [14:45] <alexvf> is it possible in the current ffmpeg state to use custom I/O in a non blocking way to read from a network stream? i've tried but it is not working [14:46] <alexvf> i want to know if i can fix the issue or i would be banging my head against a wall [14:46] <wm4> no [14:46] <wm4> all demuxers do blocking access [14:47] <wm4> nothing in the API would even allow non-blocking things [14:48] <alexvf> just to be sure we are talking about the same [14:48] <alexvf> i'm returning AVERROR(EAGAIN) in my read callback [14:48] <wm4> no [14:49] <wm4> won't work in general [14:49] <alexvf> and i suspect it is causing av_read_frame to return an incomplete frame (cause i see noise in the decoded frames) [14:49] <alexvf> is it what is happening? [14:55] <alexvf> wm4: can you confirm my guess? [14:55] <wm4> no [14:56] <wm4> if you don't return the complete read, libavformat assumes there was an error or EOF [14:57] <alexvf> yes, but i reset eof_reached and error to 0 after av_read_frame if it returns AVERROR(EAGAIN) [14:58] <alexvf> but the next av_read_frame that returns without error seems to return an incomplete frame [14:58] <alexvf> is it discarding buffered data when the callback returns an error? [14:58] <wm4> why do you expect that this works? [14:58] <wm4> libavformat reads bytes from avio, not packets [14:59] <wm4> and I'm not sure what it does (depends on the exact demuxer), but generally it will try to recover from an error if this happens [15:02] <alexvf> well, av_read_frame could just return an error and keep the buffered data, and when it is called again (and the callback works) could return a valid frame [15:02] <alexvf> i'm sure i'm missing something but that was the behaviour i expected [15:17] <wm4> alexvf: no, that would be very complicated [15:17] <wm4> alexvf: also, it would call the callback in a loop, which would not result in the "non-blocking" behavior you want [15:18] <wm4> alexvf: I believe you can return short reads from the callback, but only if you return at least 1 byte [15:22] <alexvf> wm4: yep i have checked that [15:22] <alexvf> wm4: when i can get one byte from the network it works ok [15:23] <alexvf> wm4: but if there is no data at all i have to return some error [15:24] <alexvf> wm4: since 0 is interpreted as eof, i chose AVERROR(EAGAIN) which, looking at the code at a first glance, seemed the more sensible return [15:27] <alexvf> wm4: i think that there should be an error meaning "ok, there is no data yet, but there will be soon" and av_read_frame should just return immediately and let the caller manage that situation [15:27] <alexvf> wm4: then, avoid blocking behaviour would be responsability of the caller [15:34] <alexvf> wm4: he could call av_read_frame until it reads a frame, posibly with a sleep and a timeout or make whatever he wants [15:35] <alexvf> wm4: i think i can workaround the issue by calling some function that tells me if it has something to read before calling av_read_frame, although it is a bit hackish in my current design [15:36] <alexvf> wm4: thanks for the help [15:38] <wm4> alexvf: no, lavf does not avoid or allow to avoid blocking behavior [15:39] <wm4> move it to a thread, problem solved [15:42] <alexvf> yes, that is another solution, posible cleaner, although a little longer because it does not fit very well right now :) [15:42] <alexvf> thanks for your time [16:23] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:df74811cd53e: avcodec/utils: Align dimensions by at least their chroma sub-sampling factors. [16:23] <cone-751> ffmpeg.git 03James Almer 07master:931da6a5e9dd: lavd/v4l2: don't use avpriv_ prefix for internal functions [18:10] <cone-751> ffmpeg.git 03James Almer 07master:faa9d2982969: lavu/atomic: add support for the new memory model aware gcc built-ins [18:11] <wm4> kierank: is "[FFmpeg-devel] [Patch]support more AVC Intra formats without SPS/PPS header" more of "bad thing"? [18:11] <kierank> of course [18:11] <kierank> it's trying to unwind vendor lockin [18:11] <wm4> what vendor lockin? [18:12] <wm4> vendors doing out of spec shit with their hw? [18:12] <Daemon404> yes [18:12] <Daemon404> under an NDA too [18:12] <Daemon404> in the guise of an 'open spec' [18:12] <kierank> technically it's not out of spec [18:12] <kierank> technically [18:12] <kierank> you are allowed to signal headers by *any* means in h264 [18:12] <kierank> one of those is out of band [18:12] <Daemon404> yeah but it requires e.g. a special string in the SEI [18:12] <Daemon404> UMID or w/e [18:12] <kierank> to decode no [18:12] <Daemon404> some editors did [18:13] <Daemon404> theyd just crash [18:13] <kierank> yes but that doesn't change the fact the file is legal h264 [18:13] <kierank> the decoders are illegal [18:13] <Daemon404> sure, it's legal [18:13] <Daemon404> the end result is the same though [18:13] <Daemon404> NDA etc [18:13] <kierank> wm4: it's some bullshit game panasonic and now sony play [18:13] <kierank> they can advertise open standards but you need secret stuff to play it [18:14] <wm4> nice [18:15] <kierank> wm4: read http://thomasberglund.no/post/76262479496/we-have-never-been-closer-to-a-st… [18:15] <kierank> if you particularly care [18:15] <kierank> esp my comments at the bottom [18:30] <cone-751> ffmpeg.git 03wm4 07master:e5813d96d67b: avformat/subtitles: reduce log level of UTF-16 warning [18:42] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:a0528d9ddd65: avformat/mpjpeg: make boundary tag user customizable [19:01] <wm4> ubitux: is this code familiar? https://github.com/popcornmix/omxplayer/blob/master/OMXSubtitleTagSami.cpp [19:01] <wm4> appears to convert sami to ass [19:01] <ubitux> not really [19:02] <ubitux> is it better than the one in ffmpeg? [19:02] <wm4> don't know [19:02] <wm4> but considering they have explicit support, while ffmpeg only has it because mplayer did, maybe [19:02] <wm4> maybe point the next person who's interested in improving sami support to this code [19:03] <wm4> or wait [19:03] <wm4> I'm stupid [19:03] <wm4> it parses ass in sami (?) [19:03] <wm4> not generate it [19:03] <wm4> (wat...) [19:06] <ubitux> it seems to look for srt and ass tags into sami [19:06] <ubitux> :P [19:07] <wm4> so yeah... WHAT THE FUCK [19:07] <wm4> I assume vsfilter is to blame again [19:07] <wm4> also accepts ass tags in srt [19:07] <rcombs> vsfilter is usually to blame [19:08] <wm4> but even worse that someone would use that... with sami [19:08] <wm4> of all things [19:08] <rcombs> "subtitles support dumb things!" VSFilter [19:08] <rcombs> "the wrong tags are in the wrong formats!" VSFilter [19:09] <rcombs> "subtitle APIs are becoming overly complicated!" VSFilter [19:09] <rcombs> "ebola is spreading to Texas!" VSFilter [19:09] <rcombs> etc. etc. [19:10] <wm4> "I just spilled by coffee" VSFilter [19:11] <rcombs> from #ffmpeg: "ASS subtitles aren't supported in MP4" VSFilter [19:12] <rcombs> "libav's API is incompatible with ffmpeg in subtle ways!" VSFilter [19:15] <JEEB> fuck-vsfilter.vsfilter-sucks.vsfilter-is-dying.vsfilter-hit-wtc [20:04] <Compn> rcombs : erm, no subtitles are supported in mp4 [20:04] <Compn> because mp4 is a shitty format [20:05] <Compn> only timed_text, cc , [20:05] <Compn> something else [20:09] <cone-751> ffmpeg.git 03Thomas Mundt 07master:1700fa013ef5: avformat/utils: support more AVC Intra formats without SPS/PPS header [20:11] <Compn> kierank : i'd love to see your comments at the bottom of that blog post, but disqus is so bad it wont even load :P [20:13] <Compn> maybe i just need to upgrade my web browser, seems to work in chromium [20:19] <wm4> Compn: what browser? [20:47] <Compn> wm4 : opera before the chrome rebrand. 12.16 or something. sometimes disqus works sometimes not [20:47] <wm4> ok [20:48] <wm4> in konqueror (hilariously broken by now), it only works with the webkit backend [20:49] <cone-751> ffmpeg.git 03Luca Barbato 07master:a9179b5bd6f1: configure: Check only for xcb [20:49] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:363effbb99f7: Merge commit 'a9179b5bd6f143b4a7ff48bb0d00c8f0a7cafb4b' [20:52] <Compn> html standards have gone out the window [20:52] <wm4> this has nothing to do with html [20:52] <wm4> it's all about js [20:52] <Compn> stupid js yes i know [20:52] <Compn> disqus can actually be setup to do html comments though [20:53] <Compn> ive seen it on one or two pages [20:53] <JEEB> also I never thought I'd see this day http://arstechnica.com/information-technology/2014/10/html5-specification-f… [20:55] <Compn> This has led W3C to take work done by WHATWG and use it as the basis for its own work. WHATWG's specs all explicitly permit such usage, but a number of WHATWG contributors have nonetheless complained that W3C has exercised that ability. [20:55] <Compn> lol ^^^^ remind you of anything ? [20:55] <wm4> revolutionary [20:56] <Compn> or any project... [20:56] <iive> well, the only thing that w3c added was the DRM [20:56] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:b3d11437ca55: oggenc: remove unneeded null check [20:56] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:e08ff208c793: Merge commit 'b3d11437ca55d81eeb10c923343ad69b73895fa8' [21:03] <wm4> michaelni: how hard would it be to add packed 4:4:4 formats to libswscale? [21:03] <wm4> in particular I'd be interested in this: " * This plane is an array packed 32-bit pixel data. Within each * 32-bit pixel, bits [31:24] contain A, bits [23:16] contain V, * bits [15:8] contain U, and bits [7:0] contain Y." [21:04] <wm4> that's what vdpau uses [21:05] <michaelni> probably not hard, just needs to be added to all relevant codepathes [21:08] <Compn> wm4 : samples would be useful. [21:14] Action: wm4 throws a nvidia card on Compn's head [21:15] <wm4> michaelni: that sounds hard [21:36] <cone-751> ffmpeg.git 03Vittorio Giovara 07master:f64d7e919eab: mtv: improve header check and avoid division by zero [21:36] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:9a534eda4608: Merge commit 'f64d7e919eabd427f3e6dd4a1219e448c78deb42' [21:56] <cone-751> ffmpeg.git 03Luca Barbato 07master:043ea6f7bfc5: fbdev: Use av_strerror [21:57] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:265c4771cc76: Merge commit '043ea6f7bfc59399b6b3659da785ec4cc68a008e' [21:57] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:7729f4331296: avdevice/fbdev_dec: use errno instead of ret for av_log [22:06] <cone-751> ffmpeg.git 03Luca Barbato 07master:2cd28693a590: jack: Use av_strerror [22:06] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:66b2a3fa31bd: Merge commit '2cd28693a59050717cb7da6cb229e606b1dee356' [22:21] <cone-751> ffmpeg.git 03Vittorio Giovara 07master:e9ba3098319f: assdec: check av_new_packet return value [22:21] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:a3b1e42ec4f4: Merge commit 'e9ba3098319f78c91470c05da988d865491852c5' [22:49] <cone-751> ffmpeg.git 03Vittorio Giovara 07master:84bf64d3598c: bethsoftvid: simplify return handling [22:49] <cone-751> ffmpeg.git 03Michael Niedermayer 07master:a8605be30fdd: Merge commit '84bf64d3598c98a748e609195358ea04b0cfd140' [23:50] <cone-751> ffmpeg.git 03Thomas Mundt 07master:d1b5ad396776: mxfenc: fix indentation after last commit [00:00] --- Thu Oct 30 2014

1 0

ffmpeg.log.20141029
by burek 30 Oct '14

30 Oct '14

[00:00] <deivor> I'm currently trying to record from a camera and microphone (linux) both devices have different "open" times, which means that there's a "buffer" of frames from the video at the beginning [00:00] <deivor> I've set the FPS to 15, but there's a "buffer" of images, which means that the output is at, say, 50 fps for 2 seconds, 40fps for 2 seconds ... and gets closer and closer to the 15fps over time [00:01] <deivor> I'm not really sure how to explain [00:02] <voip__> actualy am not tryng to record file i need stream to 2 IP's, i see strange bitrate=N/A in ststistics [00:02] <deivor> My output is something like this : http://pastebin.com/P1FFwi5U [00:03] <deivor> yes voip__ but the idea is that you rule out problems [00:03] <deivor> if you output to one file and it works, then your command is "OK" [00:03] <deivor> then you test with 2 files, to test if the output command is ok [00:03] <deivor> and if that works, the rtmp servers are wrong [00:03] <voip__> right [00:04] <deivor> also, (not necessary) compile ffmpeg with fdk_aac support (you get better sound with lower cpu usage) [00:09] <voip__> deivor, actualy i am using static compiled ffmpegs [00:11] <deivor> It's not necessary to do it, and most likely a pain to compile if you are on windows [00:11] <deivor> (I don't know if you are on windows or not) [00:11] <voip__> aa m on linux [00:11] <deivor> k [00:12] <voip__> i tryed to write to files instead of IP's. Recorded files are ok on vlc, but strange ststistics is same frame= 1648 fps= 25 q=-1.0 Lsize=N/A time=00:01:06.15 bitrate=N/A dup=11 drop=0 [00:12] <voip__> btrate n/a [00:13] <voip__> actualy i need stream to 2 wowza servers, audio on servers is good, but no video [00:16] <deivor> What happens if you stream to only 1 wowza server? [00:27] <voip__> deivor, with 1 server -f flv rtmp://ServerA or ServerB everything is OK. Problem comes when I trying to send 2 servers like: -f tee -map 0:v -map 0:a "[f=flv]rtmp://ServerA|[f=flv]rtmp://SeverB" [00:28] <deivor> what happens if you try with a piped process ? [00:29] <deivor> https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs [00:31] <voip__> is it will double cpu ussage ? [00:35] <deivor> nope [00:35] <deivor> it will add a little bit of overhead as you have two ffmpeg processes but it's almost nothing [00:38] <voip__> :) [00:39] <voip__> can you please help me covert piped command: ffmpeg/ffmpeg -i http://Source -preset fast -acodec aac -strict experimental -ar 44100 -ac 2 -b:a 96k -vcodec libx264 -b:v 900k -f tee -map 0:v -map 0:a "[f=flv]rtmp://ServerA|[f=flv]rtmp://ServerB" [00:40] <voip__> I just tryed but recieve eror -c copy is not recognized [00:46] <deivor> ffmpeg/ffmpeg -i http://Source -preset fast -acodec aac -strict experimental -ar 44100 -ac 2 -b:a 96k -vcodec libx264 -b:v 900k -f mpegts | ffmpeg/ffmpeg -f fmpegts -i - -c copy -f flv "rtmp://ServerA" -c copy -f flv "rtmp://ServerB" [00:47] <voip__> thank you ! [00:50] <deivor> does it work now? [00:52] <deivor> I'll be back in 45 minutes or so [00:52] <voip__> no :) At least one output file must be specified [00:52] <voip__> np [00:54] <deivor> oh [00:54] <deivor> ffmpeg/ffmpeg -i http://Source -preset fast -acodec aac -strict experimental -ar 44100 -ac 2 -b:a 96k -vcodec libx264 -b:v 900k -f mpegts - | ffmpeg/ffmpeg -f fmpegts -i - -c copy -f flv "rtmp://ServerA" -c copy -f flv "rtmp://ServerB" [00:54] <deivor> dash at the end of the first process [00:54] <deivor> now i'm gone [00:54] <voip__> i removed dash :) [00:54] <voip__> ok np we will talk tmororow :) [01:13] <voip__> guys, what mean : Using AVStream.codec.time_base as a timebase hint to the muxer is deprecated. Set AVStream.time_base instead. [01:13] <voip__> ? [01:32] <deivor> I'm back [01:33] <deivor> change your parameters as the error tells you to [01:52] <jellosea> hello, how do i get CBR for vp8? ive tried two pass but it doesnt give me CBR [01:57] <llogan> jellosea: why do you want CBR? [01:57] <llogan> or maybe i should ask why do you want vp8? [01:58] <deivor> if you can go vp9 / x264 [01:58] <llogan> vp9 is too slow to encode for sane people [02:01] <jellosea> no have to use vp8 [02:01] <jellosea> for my application [02:01] <jellosea> llogan: why do i need CBR? because i need it [02:01] <jellosea> for my application [02:01] <jellosea> i need to create a file with a specifc exact bitrate [02:05] <llogan> jellosea: then you'll have to read this http://www.webmproject.org/docs/encoder-parameters/ [02:06] <llogan> map the options to the libvpx avoptions by looking at 'ffmpeg -h encoder=libvpx' [02:06] <llogan> also https://trac.ffmpeg.org/wiki/Encode/VP8#ConstantBitrate [02:06] <jellosea> i did [02:06] <jellosea> the shit dont work [02:06] <jellosea> i read all dat shit [02:07] <deivor> Is there a specific signal to use for killing ffmpeg? So it closes the current stream correctly [02:07] <deivor> I'm running pkill atm [02:10] <llogan> jellosea: if you think it's a problem specific to libvpx then you'll have to ask in #vp8 or their mailing list webm-discuss [02:11] <jellosea> well theres no --end-usage in ffmpeg [02:11] <jellosea> but its in the webm encoder [02:11] <jellosea> and end-usage in webm says its for cbr [02:11] <jellosea> however ffmpegs instructions for cbr doesnt work [02:13] <jellosea> and im using 2 pass on top of this shit [02:13] <jellosea> how is it giving me really off bitrates with two pass [02:15] <llogan> the wiki article is from other users. it may not be correct. [02:15] <jellosea> whats -g for ffmpeg? [02:17] <llogan> group of pictures [02:17] <jellosea> a few other people seem to have this problem [02:17] <jellosea> but there seems to be no real answer [02:17] <llogan> does it work the way you want using vpxenc? [02:20] <jellosea> what is 3pass [02:20] <jellosea> i havent used vpxenc [02:20] <jellosea> will a 3rd pass help me [02:20] <llogan> 3 pass is an alias for "a waste of time" [02:20] <jellosea> ok [02:21] <jellosea> does the K have to be captialized [02:21] <llogan> if vpxenc doesn't create what you want then the problem is probably not with ffmpeg [02:21] <jellosea> -bitrate:v 250K [02:21] <llogan> no, don't capitalize [02:21] <jellosea> ok i didnt [02:21] <llogan> -b:v 250k [02:21] <jellosea> yea [02:22] <jellosea> well with -loglevel debug it seems that [02:23] <jellosea> libpvx seems to be receiveng cbr mode [02:23] <jellosea> but this bitch wont output cbr [02:25] <deivor> llogan, anything on 'correctly' killing ffmpeg? (for example, ffmpeg will produce a 'corrupt' file if killed when encoding mp4 ) [02:28] <llogan> deivor: how are you currently doing it that makes the corrupt output? [02:29] <deivor> It doesn't, yet (I'm making a web-remote for live-streaming) [02:29] <deivor> I'm just running pkill, but I don't know if it'll mess up the RTMP stream [02:31] <llogan> i can't duplicate the issue using pkill if outputting mp4 [02:35] <relaxed> deivor: try, killall -INT ffmpeg [02:38] <deivor> llogan: If you start transcoding certain codecs and ctrl-c, you'll get a 'broken video' (easy to fix), whereas if you hit 'q' the video stops at the same place but works [02:38] <deivor> that's what I mean [02:38] <deivor> relaxed: thanks I'll try it [02:39] <llogan> unable to duplicate [02:41] <relaxed> llogan: I think it happens when the output container is mp4 [02:42] <llogan> that's what i tried [02:44] <relaxed> yeah, pkill works here too [03:24] <nezZario> Hi folks [03:29] <deivor> I'm trying to get www-data to execute ffmpeg as another user but won't give me access to alsa.. is this ffmpeg's fault? [03:30] <deivor> running as user "curso" I get my script running wonderfully [03:30] <deivor> running as www-data with correct sudo permissions: sudo -u curso script [03:30] <deivor> ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_OPEN failed : Invalid argument(22) [03:30] <deivor> [alsa @ 0x1669510] cannot set parameters (Invalid argument) [03:30] <deivor> btheadset: Input/output error [03:30] <deivor> Should these permissions be handled by ffmpeg or not? (If not, I'll have to check alsa/bluetoothd) [03:35] <deivor> nevermind, I had a stray screen blocking access [04:24] <nezZario> Guys [04:24] <nezZario> How can I convert .mod to .flv or .mp4? [04:32] <llogan> nezZario: ffmpeg -i input.mod output.mp4 [05:09] <jellosea> does the order of arguments matter in ffmpeg [05:19] <llogan> jellosea: yes http://ffmpeg.org/ffmpeg.html#Synopsis [05:22] <nezZario> hmm [05:22] <jellosea> are things like sn and an input file options [05:22] <nezZario> Well let me ask this, .. what would be a good 'something' (heck, I'd go for a book if it's <$20) to teach me about codecs/containers/video/things I'll encounter in ffmpeg? [05:23] <jellosea> lol [05:23] <nezZario> I'm pretty lost.. I did some converting from mod to flv (no idea what codecs?) .. when I re-play it in flv, the quality is pretty decent in the flv but the video is literally a little smaller even though VLC player says they're both 720x480 [05:23] <jellosea> mod? do you mean mov [05:24] <nezZario> no, the file is ".mod" .. "JVC .mod format" or something [05:24] <jellosea> llogan: is -passlogfile video_name.log file? [05:24] <jellosea> llogan: fine rather [05:24] <jellosea> is that how i use a unique two pass log file name [05:27] <llogan> nezZario: https://trac.ffmpeg.org/wiki/Encode/H.264 [05:28] <llogan> jellosea: -sn and -an are output options. see the documentation. it will tell you if the option is global, input and/or output. [05:29] <llogan> http://ffmpeg.org/ffmpeg.html [05:29] <jellosea> oh ok [05:29] <nezZario> llogan: any idea how I check if my ffmpeg is compiled with h264 support? ( I don't think it is, everytime I tried it said invalid encoder: ... or something along those lines) [05:30] <jellosea> so what are generic options [05:30] <jellosea> just type ffmpeg [05:30] <llogan> nezZario: run "ffmpeg" the console output. look for "--enable-libx264" [05:30] <jellosea> llogan: is generic global [05:30] <llogan> i don't know of an option named generic [05:31] <jellosea> no i mean in the documentation [05:31] <jellosea> for example -loglevel [05:31] <jellosea> it doesnt say if its global input or output [05:33] <llogan> i guess they don't all say, but i believe it is considered a global option [05:36] <jellosea> ok [05:44] <jellosea> llogan: what should i set -bufsize if i want strict CBR [05:55] <relaxed> why do you need/want cbr? [05:57] <jellosea> not relevant [06:00] <relaxed> jellosea: https://trac.ffmpeg.org/wiki/Encode/H.264 [06:00] <jellosea> im not even using h264 [06:00] <jellosea> also that shit dont work [06:03] <jellosea> https://gist.github.com/anonymous/a6378ef3d7e4f8baf552 [06:05] <relaxed> please reread fflogger's message [06:11] <jellosea> https://gist.github.com/anonymous/38596fe82b0518589471 [06:13] <relaxed> and you're issue is you think it's not CBR? [06:25] <deivor> Hi, anyone up? [06:29] <jellosea> relaxed yes it is not CBR [06:29] <jellosea> i ask for the bitrate to be 250k [06:30] <jellosea> and the output is like 500k [06:38] <deivor> I have a camera that provides constant FPS, but ffmpeg is giving me 1 fps extra, which makes the video drift from the audio over time [06:38] <deivor> I have ffmpeg -framerate 20 -r 20 -i - [...] -vsync drop -r 20 -async 1 OUTPUT [06:39] <deivor> that's the relevant part, at least [06:40] <jellosea> so relaxed you have anythign to say [06:56] <relaxed> jellosea: how did you verify it's not CBR? [06:59] <jellosea> relaxed: its not hitting my target bitrate [06:59] <jellosea> how do i get it to hit my target bitrate [07:00] <jellosea> Bit rate : 435 Kbps [07:00] <jellosea> in mediainfo [07:00] <relaxed> did you calculate this from the ouput's size or are you just guessing? [07:00] <jellosea> for video [07:00] <jellosea> mediainfo [07:00] <deivor> ffprobe ? [07:00] <relaxed> mediainfo is wrong quite a bit [07:02] <jellosea> its not wrong [07:02] <jellosea> based on the filesize [07:02] <jellosea> and based on ffmpegs output [07:03] <jellosea> so you have no info [07:08] <deivor> Does your output have audio? Are you sure you are comparing audio+video bitrates (and you have set them correctly on the command) ? [07:12] <jellosea> yes [07:12] <jellosea> yes i am [07:12] <jellosea> jesus [07:12] <jellosea> no one is giving me any help [07:12] <jellosea> fucking stupid ass advice [07:13] <relaxed> what's the output's size in MB? [07:13] <jellosea> 14.6 MB [07:13] <jellosea> this is retarded [07:13] <jellosea> all you are gonna ask me is if ithe bitrate is right [07:13] <jellosea> instead of finding out why the bitrate is off [07:14] <jellosea> if you dont know why the bitrate will be off then just say so [07:14] <jellosea> instead of trying to find out that the bitrate is actually right [07:18] <relaxed> we're not paid to sit here and provide you with real-time support, so why don't you develop some patience [07:29] <jellosea> why are you wasting my time if you dont know what the problem is [07:30] <jellosea> just say you dont know instead of [07:30] <jellosea> paste the log [07:30] <jellosea> paste this [07:30] <jellosea> why do you want cbr [07:30] <jellosea> why do you want to encode video [07:30] <jellosea> maybe you should read a book [07:30] <jellosea> like wtf [07:31] <deivor> relaxed: do you know if I can override the image PTS with constant PTS? I'm getting just a little bit over the fps I should from the camera, and over time it goes out of sync [07:34] <relaxed> deivor: pastebin the command and output so i can see what's going on. [07:37] <deivor> http://pastebin.com/mMXy30dr [07:38] <deivor> the input is supposed to be 20fps, it hovers at about 20.3 (at the moment I've the command with -r 20.3) but I'm going to use in sessions about 2 hours long, so a small drift is huge [07:40] <deivor> So I'm thinking dropping the PTS and forcing it to spit 20 frames per second with correct PTS should do it (if fps > 20, drop, if fps < 20 duplicate) [07:43] <relaxed> instead of -r 20 use -framerate 25 [07:44] <relaxed> remove the follow -r -vsync and -async [07:44] <relaxed> following* [07:44] <deivor> framerate 25 for output or input? [07:45] <relaxed> -framerate applies to the input [07:45] <deivor> my input is fixed at "20"fps, why should I use -framerate 25? (isn't framerate used to 'ask' for a certain framerate?) [07:46] <relaxed> I'm going by "Stream #0:0: Video: h264 (High), yuv420p, 1280x720, 25 tbr, 1200k tbn, 50 tbc" [07:47] <relaxed> but first off, start with a slimmed down command and it encode it to a local file [07:48] <deivor> The drift problem is from the camera [07:48] <relaxed> ok [07:48] <deivor> the 25 tbr is because I didn't let ffmpeg analyze the stream (analyzduration and probesize too small) to reduce latency [07:49] <relaxed> I see [07:52] <relaxed> then "ffmpeg -framerate 20 -i input -r 20 ..." should drop or dup frames to maintain sync. [07:54] <relaxed> play with -framerate's value to see if it helps [08:23] <jellosea> how can i add two vf filters [08:24] <relaxed> https://www.ffmpeg.org/ffmpeg-filters.html#Description [08:25] <jellosea> thanks [08:39] <jellosea> if i do -vf scale=-1:480; ass=sub.ass it doesnt work [08:39] <jellosea> something with the i think [08:40] <jellosea> if i do -vf ass=sub.ass it works, or -vf scale=-1:480 it works too [08:40] <jellosea> combining it with quotes and semicolon doesnt seem to work [08:41] <relaxed> -vf scale=-1:480,ass=sub.ass [08:43] <jellosea> works thanks [08:45] <relaxed> if the subs look odd you might have to reverse the filters [09:06] <svvitch> what format to use for png image, as -f param ? [09:08] <svvitch> when adding overlay [09:13] <relaxed> svvitch: you shouldn't need to use -f unless you're piping it in. [09:14] <relaxed> ffmpeg -i video -i image.png -filter_complex overlay output [09:29] <svvitch> relaxed: the problem is that png file is without extension and ffmpeg does not understand what kind of file is [09:36] <relaxed> try, ffmpeg -i video -f image2 image ... [09:36] <relaxed> er, ffmpeg -i video -f image2 -i image [10:42] <alexvf> hi, i need help with custom I/O to read a network stream (2.2.2) [10:42] <alexvf> i had a blocking implementation of the read callback that worked right, but i had to changed to non-blocking because that thread needs to attend some user input [10:43] <alexvf> now, i'm returning AVERROR(EAGAIN) when i can't read from network [10:43] <alexvf> and i'm seeing noise in the decoded images [10:44] <alexvf> is even possible in the current ffmpeg state to have a non blocking custom I/O for network streams? [11:17] <ahessler> how much ram is needed for 2350x1080 ? [11:17] <ahessler> 1GB is not enough it seems [11:22] <ahessler> oh, 2G was enough [11:26] <alexvf> please, someone with some knowledge on AVIOContext can tell me if custom I/O supports AVERROR(EAGAIN)? It seems that av_read_frame retursn incomplete frame when the read callback returns that value [11:40] <akusingla> hello!! I am new to ffmpeg , I wanted to know if this api could be used for clipping videos on server side? [11:50] <LucasG> Hi guys, anyone can share ffserver for a while? I have Connection reset by peer when i`m trying sendstream via http from ffmpeg to my ffserver. Complete log: http://pastebin.com/D9LhWrUc [12:41] <line0> is there any way to tell ffmpeg to not do any character encoding conversions with -sub_charenc? [12:42] <line0> (when extracting subtitles) [12:43] <line0> like, just extract bytes and let something else take care of the rest? [14:00] <relaxed> line0: did you see `ffmpeg -h full|less +/sub_charenc_mode` [14:03] <line0> ah [14:03] <line0> thanks [16:33] <djook> hye huys [16:33] <djook> I have a little question, not really code related [16:34] <djook> I'd like to know if there is a specific country of origin fro the project [16:34] <djook> as vlc is French, mplayer Hungarian and cyrus British.... [16:36] <djook> I found the dns to be Name Server:NS.ARTHIST.MTA.HU [16:36] <djook> Name Server:THOT.BANKI.HU [16:36] <djook> Name Server:NS3.MPLAYERHQ.EU [16:36] <djook> so Hungarian as well ? [16:40] <klaxa|work> afaik the original author of ffmpeg was fabrice bellard [16:40] <iive> ffmpeg is hosted on same server as mplayer, that's why it is hosted in hungary [16:42] <djook> iive, thanks to let me know ... [16:42] <djook> I'll have a closer look at Bellard "origins" [16:42] <iive> it started on sourceforge, then moved to arpi's mphq in hungary [16:43] <iive> then a new machine that KotH set up in Switzerland. [16:44] <iive> after the take over, arpi gave a new hosting machine. [16:44] <djook> In the french wikipedia article about Bellard, He is said to be the first ffmpeg coder/dev [16:45] <djook> iive, I remember that some servers were compromised circa 2010/11 [16:45] <djook> ffmpeg compilation is one of my favorite computer game [16:46] <djook> I got myself an xserver ppc64 just for ffmpeg tests [16:46] <djook> I played a bit with a ps3 in 2011 but du to the 256M of ram [16:46] <djook> the OOM occured very fast [16:46] <iive> hehe, that's great, do you run fate on it? [16:47] <djook> fate ? [16:47] <djook> is it an OS ? [16:47] <iive> it is automated test environment. [16:47] <iive> http://fate.ffmpeg.org/ [16:47] <djook> ok [16:48] <djook> I ll have a look at that [16:49] <djook> For the moment I only encode libx264 \/libfaac [16:50] <iive> these automated tests allow finding out quickly if some change breaks some specific architectures. [16:50] <iive> e.g. when some change is not endian safe. [18:21] <astr> how to sync audio with video? [18:26] <voip_> hi guys [18:26] <voip_> pls help me with 1 stream processing and multiple IP outputs. In ffmpeg status i have someting like frame= 412 fps= 26 q=-1.0 Lsize=N/A time=00:00:16.74 bitrate=N/A dup=19 drop=0. My output hase audio, but no video. [18:26] <voip_> <voip__> http://pastebin.com/wByE8CB8 [18:39] <khaosmarmoset> hey. [18:39] <khaosmarmoset> so regarding my webcam recording project. [18:39] <khaosmarmoset> I can now set the bit rate and the i-frame period to suit youtube and everything seems great [18:40] <khaosmarmoset> the only weird thing is that I can't seem to affect the fps of the webcam myself very much. [18:40] <khaosmarmoset> or at all, really. [18:40] <khaosmarmoset> sometimes when I start the process I get (after stabilisation) 25fps. [18:40] <khaosmarmoset> usually it stabilises to about 15 fps. [18:41] <c_14> Can you/are you using -r as an input option? [18:41] <khaosmarmoset> the bitrate doesn't seem to have very much to do with it. I can set it higher or lower, but usually I drift back into 15 fps. [18:42] <khaosmarmoset> I am using -r, but again it doesn't seem to be doing anything. [18:42] <khaosmarmoset> v4l2-ctl's --set-parm also doesn't seem to be doing very much. [18:42] <khaosmarmoset> ditto with my uvch264ctl's frame interval setting. [18:44] <khaosmarmoset> so basically I have three knobs to twiddle, but none of them work. [18:45] <khaosmarmoset> I don't really mind all that much, because 15fps is enough for me. 24 would be better. 30 would be great [18:45] <c_14> You can get ffmpeg to drop/duplicate frames, but if your webcam isn't being reasonable there isn't much you can do on that end. [18:46] <khaosmarmoset> when using -acodec copy -vcodec copy, then is the fps I see when recording the actual measured fps from the webcam? [18:47] <c_14> The fps= you see when recording is the recording fps, if you can record in realtime, then yes it is. (assuming real-time source) [18:47] <khaosmarmoset> right, ok [18:47] <khaosmarmoset> damn, gotta run out for a bit before the shops close. brb. [18:54] <astr> khaosmarmoset, how d you keep audio and video in syic? [18:54] <astr> sync [19:10] <Lokie> hello, I am trying to change a mkv to a mp4. -codec copy works fine for the video & audio streams but the subtitles (.ass) don't get copied. Shouldn't -codec copy copy all the streams? [19:11] <rcombs> ASS subtitles aren't supported in MP4 [19:11] <Lokie> oh thought it was srt [19:11] <rcombs> neither are those [19:12] <Lokie> oh [19:13] <Lokie> mp4 supports only hardsubs? [19:13] <khaosmarmoset> astr: erh. I'm not sure. [19:13] <rcombs> I think you could convert to mov_text [19:13] <astr> khaosmarmoset, ok [19:14] <khaosmarmoset> astr: which is to say, I just run ffmpeg -f alsa -i hw:1 -f v4l2 -r 24 -vcodec h264 -i /dev/video0 -vcodec copy -acodec aac -strict -2 -f flv [19:14] <khaosmarmoset> astr: if you have any pointers, please let me know! [19:14] <rcombs> khaosmarmoset: 0x3A28213A [19:14] <astr> when 2 "-vcodec" ? [19:14] <astr> when=why [19:15] Action: khaosmarmoset faults rcombs [19:16] Action: rcombs SIGSEGVs [19:16] <khaosmarmoset> astr: the first one chooses the codec for the v4l2 input. the second one says to just move it along [19:16] <khaosmarmoset> (I think. I'm not 100% sure) [19:18] <khaosmarmoset> astr: yeagh, without the first -vcodec h264 ffmpeg resets it to raw YUYV and then the frame rate drops to 5 fps [19:19] <voip_> trate doesn't seem to have very much to do with it. I can set it higher or lower, but usually I drift [19:19] <astr> what about -input_format see ffmpeg -h full |less press "/" and paste "V4L2 indev AVOptions"and enter to search [19:19] <voip_> http://pastebin.com/jePTMDbc [19:19] <astr> don't know only just leaned of it [19:19] <astr> "first -vcodec h264 ffmpeg resets it to raw YUYV" doesn't make sense to me [19:20] <khaosmarmoset> I mean if I run it without the -vcodec h264, then I see that ffmpeg has chosen YUYV as the pixel format according to v4l2-ctl --all [19:21] <khaosmarmoset> this is basically raw uncompressed frames afaict, and the hw only supports 5 fps for that case [19:21] <astr> ok [19:22] <kepstin-laptop> oh, that's an input option on the v4l device, right. [19:23] <Lokie> rcombs I get Stream #0:2 -> #0:2 (ssa -> mov_text) [19:23] <Lokie> and subtitle:10kB but I don't see any subs [19:23] <Lokie> used: -i source.mkv -i source/para04.ass -c copy -c:s mov_text target [19:26] <rcombs> your player might not support it? [19:26] <rcombs> mov_text is rather obscure [19:26] <Lokie> using ff to play it so maybe [19:26] <Lokie> FFmpeg does not (yet) support [19:26] <Lokie> setting all properties for mov_text correctly. mar 2013 [19:28] <voip_> please help http://pastebin.com/jePTMDbc [19:30] <llogan> voip_: loglevel debug is often more annoying that helpful [19:30] <llogan> why are you piping from ffmpeg to ffmpeg? [19:33] <khaosmarmoset> how can I have the output go nowhere? [19:33] <llogan> -f null - [19:33] <llogan> http://ffmpeg.org/ffmpeg-formats.html#null [19:34] <khaosmarmoset> thanks [19:34] <khaosmarmoset> so that has nothing to do with it. [19:35] <khaosmarmoset> I wanted to eliminate the -f flc as something that might be contributing to my problem [19:35] <khaosmarmoset> -f flv even [19:37] <voip_> llogan, i neeed [19:37] <voip_> multiple outputs [19:37] <voip_> https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs#no1, adm my command is ffmpeg -i input1 -i input2 -acodec & -vcodec & -f mpegts - | \ [19:37] <voip_> ffmpeg -f mpegts -i - \ [19:37] <voip_> -c copy output1 \ [19:37] <voip_> -c copy output2 \ [19:37] <voip_> -c copy output3 \ [19:38] <voip_> i dont know why in debug null, but in command i typed right output adress [19:39] <llogan> why not use tee muxer? [19:40] <voip_> i tryed with muxer, dosent work for me [19:41] <voip_> ffmpeg/ffmpeg -i http://Source -preset fast -acodec aac -strict experimental -ar 44100 -ac 2 -b:a 96k -vcodec libx264 -b:v 900k -f tee -map 0:v -map 0:a "[f=flv]rtmp://ServerA|[f=flv]rtmp://ServerB" [19:42] <llogan> did you try the -flags +global_header suggestion? [19:42] <voip_> no, can you help me, write right command ? [19:43] <llogan> i mentioned it yesterday, but you didn't respond [19:43] <voip_> i dont know how use -flags +global_header [19:43] <llogan> just add it as an ouptut option [19:44] <voip_> like: ffmpeg/ffmpeg -i http://Source -preset fast -acodec aac -strict experimental -ar 44100 -ac 2 -b:a 96k -vcodec libx264 -b:v 900k -f tee -map 0:v -map 0:a -flags +global_header "[f=flv]rtmp://ServerA|[f=flv]rtmp://ServerB" ? [19:45] <llogan> sure, try that [19:45] <voip_> lets me chek [19:46] <Lokie> hmm if I were to burn the subs on the mp4 is there a flag that will used the same code / quality from the source? [19:46] <Lokie> -d [19:46] <Lokie> codec* [19:46] <llogan> not with a lossy encoder [19:46] <Lokie> source is a h264 mkv [19:47] <Lokie> not sure that I understood your comment though [19:47] <Lokie> ffprobe gives: Stream #0:0: Video: h264 (High), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 23.81 fps, 23.81 tbr, 1k tbn, 47.95 tbc [19:47] <llogan> you can't get "same quality" if you use an encoder that makes a lossy output [19:48] <llogan> you can get something the looks lossless or nearly so, but technically it is not [19:51] <voip_> llogan, brilliant, it works! but durig encode i see bitrate=N/A why ? [19:52] <llogan> i don't know [19:53] <voip_> thank you ! [19:56] <voip_> guys, during encoding frame= 77 fps= 33 q=24.0 size= , what is meaning variable q=24.0 ? [20:02] <khaosmarmoset> oh [20:02] <khaosmarmoset> wow. [20:03] <khaosmarmoset> when I turn the lights on I get 30 fps just like momma promised [20:03] <khaosmarmoset> woot! [20:03] <kepstin-laptop> yeah, most webcams use slower shutter speeds in the dark, limiting fps. [20:04] <khaosmarmoset> right. turns out I didn't know that! [20:04] <khaosmarmoset> thanks webcam. you were saving me from meself. [20:04] <khaosmarmoset> :-D [20:10] <voip_> guys, during encoding i see ststus: frame= 77 fps= 33 q=24.0 size=.., what is meaning variable q=24.0 ? [20:11] <JEEB> the value is rather useless in modern formats, but it is supposed to note some average quantizer or so [20:11] <JEEB> it really is useless and in general tells you absolutely nothing [20:12] <voip_> oh, thanks [20:15] <voip_> one more question, can you sgggset any special parameters, for taking streams from far away servers (too many hops to server, pinging time too high) [21:46] <voip_> [aac @ 0x317c7e0] Input buffer exhausted before END element found [21:46] <voip_> Error while decoding stream #0:0: Invalid data found when processing input [21:46] <voip_> [h264 @ 0x38b40e0] negative number of zero coeffs at 29 18 [21:46] <voip_> [h264 @ 0x38b40e0] error while decoding MB 29 18 [22:04] <voip_> llogan, do you know how to fix problem with : [22:04] <voip_> [h264 @ 0x38b40e0] negative number of zero coeffs at 29 18 [22:04] <voip_> [h264 @ 0x38b40e0] error while decoding MB 29 18 [22:04] <voip_> [h264 @ 0x38b40e0] concealing 812 DC, 812 AC, 812 MV errors in I frame [22:05] <llogan> voip_: no. and when you have a problem you should always include your command and the console output (using a pastebin service) [22:07] <voip_> llogan, sorry http://pastebin.com/PdmRAcBn [22:44] <voip_> please help http://pastebin.com/PdmRAcBn [23:08] <henriqueleng> I'm recording my desktop but it is using a lot of my CPU, can I speed up this process, maybe with post video processing? [23:11] <henriqueleng> pastie.org/9684356 [23:11] <henriqueleng> [23:12] <llogan> henriqueleng: where's the rest? [23:12] <henriqueleng> llogan: rest of what? [23:12] <llogan> show your EXACT ffmpeg command and the COMPLETE console output [23:13] <llogan> that seems to get ignored more than it doesn't [23:13] <llogan> why? [23:16] <henriqueleng> llogan: http://pastebin.com/ybk4ALuC [23:22] <llogan> henriqueleng: why use the 1.2 release branch instead of git master or most recent release branch? use -framerate instead of -r for x11grab. changing "-b:v 10k" to "-qp 0" may be slightly faster (and it seems to be ignoring your -b:v anyway) [23:24] <llogan> ...also it was set to 10k. no wonder it ignored. [23:26] <llogan> oh, it ignored because you tried to set -b:v and -preset as input options [23:26] <llogan> move preset after "-i :0.0+0.0" [23:33] <henriqueleng> llogan: Qhat is -qp 0 for? [23:34] <llogan> lossless output. it may encode "faster". then you can re-encode it at your leisure. [23:35] <henriqueleng> llogan: And should I maintain the -preset ultrafast? [23:35] <llogan> yes [23:35] <henriqueleng> llogan: Right, thanks for helping [00:00] --- Thu Oct 30 2014

1 0

ffmpeg-devel.log.20141028
by burek 29 Oct '14

29 Oct '14

[02:24] <cone-54> ffmpeg.git 03Michael Niedermayer 07master:5145d22b88b9: avcodec/diracdec: Tighter checks on CODEBLOCKS_X/Y [02:24] <cone-54> ffmpeg.git 03Michael Niedermayer 07master:39680caceebf: avcodec/dirac_arith: fix integer overflow [08:51] <cehoyos> Compn: The xcb options have to be identical, it is supposed to be a drop-in replacement. [08:51] <cehoyos> The pts code was "reimplemented". [08:52] <cehoyos> (I am of course very grateful that you see my point about what they really are but in this case their point can at least be argued.) [09:42] <ubitux> are we really putting back the daemon mode? [09:42] <ubitux> (in ffserver) [09:42] <ubitux> is it really up to ffserver to do that? [09:43] <ubitux> looks like something that could be handled with any decent service manager, in a probably way more reliable way [09:44] <arwa> I am getting this error - " rsync: failed to connect to fate-suite.ffmpeg.org (192.190.173.45): Connection refused (111) " when I am running this command - " make fate-rsync SAMPLES=fate-suite/ ". [09:44] <arwa> what should I do? [09:45] <ubitux> you're like at least the 3rd opw student complaining about this [09:46] <ubitux> is there a great wall of india or something? [09:46] <ubitux> what kind of network are you on? [09:47] <arwa> hahhahahaa :p [09:47] <ubitux> ask your sysadmin to open the necessary ports or whatever [09:47] <arwa> I am behind a proxy....do i need to configure the proxy settings to access this? [09:48] <ubitux> yes [09:51] <ubitux> wm4: https://trac.ffmpeg.org/ticket/4059 [09:53] <wm4> man nicolas [09:53] <wm4> didn't he argue all the fucking time support for encoding should always done at the demuxer level [09:56] <ubitux> oh come on [09:57] <wm4> come on what [09:57] <wm4> Matroska packets are also in utf-8 [09:57] <ubitux> haha this is so stupid [09:57] <wm4> so much for his "consistency" [09:57] <ubitux> i mean your insults :p [09:58] <wm4> his comments make no sense, he just saw that I wrote utf-16 support, so it was obviously Wrong in his view [09:58] <ubitux> cehoyos: maybe a proper fix involves changing the avctx->sub_charenc_mode [09:58] <ubitux> into these decoders [09:58] <ubitux> or something along these lines [09:59] <wm4> "Support for UTF-16 in text files was added savagely at the demuxer level," [09:59] <wm4> "savagely" [09:59] <ubitux> :D [09:59] <wm4> and HE argued all the time that all encodings should be handled at the demuxer level [09:59] <ubitux> i don't exactly remember what was the exact purpose of sub_charenc_mode but i believe he wanted to use that [09:59] <ubitux> but i'll let you two fight over this [09:59] <wm4> so he didn't care about fucking over the API completely [10:00] <wm4> anyway, fucking retard. [10:00] <ubitux> ah seems nicolas suggested the same thing [10:00] <wm4> not to mention that the patch was in review for 6 months [10:00] <ubitux> :) [10:04] <wm4> nicolas' complaint is that his shitty code boils down to corrupting utf-8 input [10:05] <cehoyos> ubitux: Sorry if I misunderstand but I thought I am the one suggesting the hacks? [10:05] <cehoyos> Do you mean that the demuxer should overwrite the user's option? [10:05] <ubitux> i don't know [10:06] <cehoyos> I'll commit the warning later, it will fix the user's case, if anybody has a real fix, I'll be very happy about it! [10:06] <ubitux> the sub_charenc design nicolas asked me to follow when i added it is still kind of obscure to me [10:07] <ubitux> but it looks like it was in that spirit, and it seems that's what nicolas is suggesting as well [10:15] <wm4> IMO it would be fine to ignore the user's settings if the subtitles are known to be in utf-8 and no force mode is specified (such a mode would have to be added) [10:15] <wm4> or it could just not corrupt valid utf-8, unless you want to cater to the odd use-case of fixing double-encoded utf-8 or something [10:16] <thardin> I ý unicode [10:17] Action: ubitux wonders if that fail was on purpose [10:18] <ubitux> wm4: well, if it's set on automatic, your detection should take over it [10:19] <ubitux> but since it's talking about libavcodec etc [10:19] <ubitux> i really don't know how that's suppose to be deal with [10:19] <thardin> ubitux: I'm alluding to this: http://seriot.ch/resources/talks_papers/i_love_unicode_softshake.pdf [10:20] <ubitux> alright ;) [10:20] <wm4> I also think these details should not be handled in libavcodec [10:20] <wm4> but ffmpeg.c, or a helper function [10:20] <wm4> libavcodec is not the fucking userinterface [10:20] <wm4> even if ffmpeg.c seems to think so [10:27] <J_Darnley> "double-encoded utf-8" wat [10:28] <ubitux> this pdf is really cool thardin [10:28] <ubitux> i like the last pages dedicated to all kind of stuff broken on osx [10:31] <thardin> but osx does have the beer mug emoji [10:32] <ubitux> can you use it in a filename? [10:32] <thardin> of course [10:32] <thardin> a coworker of mine uses it instead of $ at the end of his terminal line on fridays [10:32] <thardin> like "use@machine~[beer mug] echo woo, friday!" [10:34] <J_Darnley> :) neat [10:35] <J_Darnley> But I frown upon that horrible bit of unicode. [10:36] <Kovensky> ðº ð» [11:04] <cone-768> ffmpeg.git 03Rémi Denis-Courmont 07master:26ab504ad8d2: vdpau/h264: request MAIN rather than BASELINE VDPAU profile for CBP [11:04] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:ab9ba8887705: Merge commit '26ab504ad8d2b23535c9a0ad43bf1fd0e6aa0893' [11:06] <Daemon404> beer mug emoji eh ðº [11:13] <thardin> [11:15] <thardin> unicode related libraries should have names outside US-ASCII [11:15] <nevcairiel> but then you would need the library to read the library name, which you dont have at that point! [11:15] <thardin> -luoT1 [11:15] <cone-768> ffmpeg.git 03Rémi Denis-Courmont 07master:ce91b2eae6ea: vdpau: return MAIN instead of BASELINE for H.264 CBP [11:16] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:dd5123a04c67: Merge commit 'ce91b2eae6ea52fc1a7003566d26db20ca62d745' [11:29] <cone-768> ffmpeg.git 03Anton Khirnov 07master:4ad1eba01186: lavd: fix building x11grab after a6674d2 [11:29] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:c2e995674fc2: Merge commit '4ad1eba011860224831ce0bb3123f6f55716b68a' [11:48] <thardin> what do I need to have installed to make ffplay again? [11:48] <nevcairiel> sdl [11:49] <thardin> I tried installing libsdl2-dev and reconfiguring. no dice [11:51] <iive> sdl1, afaik [11:52] <thardin> libsdl1.2-dev [11:56] <cehoyos> You need sdl-config in your path. [11:57] <cehoyos> Or check_pkg_config sdl has to succeed. [11:57] <cehoyos> "pkg-config --exists --print-errors sdl" [11:58] <cehoyos> "sdl-config --version" [11:58] <cehoyos> One of them has to succeed [11:58] <cehoyos> For more info, search for "check_pkg_config sdl" in config.log [12:00] <thardin> it works [12:00] <thardin> working on a patch for tickets 4040 and 3278 [12:01] <wm4> reading that unicode doc [12:01] <wm4> they seriously suggest using wchar_t over char? [12:02] <thardin> they probably have their reasons [12:02] <cehoyos> thardin: The patch will be much appreciated! [12:03] <thardin> glad to see the avc-intra patch looking much nicer too [12:03] <wm4> wchar_t does not need to have anything to do with unicode [12:03] <wm4> it was created for ancient multibyte non-sense [12:04] <wm4> like shift-jis and what not [12:06] <cehoyos> https://trac.ffmpeg.org/query?status=!closed&keywords=~mxf shows one or two more issues: Do you have a possibility to test if the file from ticket 3624 is valid? [12:07] <rcombs> wm4: I think they're confusing C/C++ best practices for Windows best practices [12:08] <wm4> that must be the reason [12:08] <wm4> but windows uses utf-16 for wchar_t [12:08] <wm4> which is not ideal at all [12:30] <rcombs> <wm4> but Windows is not ideal at all <-- FTFY [12:39] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:280da99a8fdd: avdevice/xcbgrab: set avclass category [12:39] <cone-768> ffmpeg.git 03Christophe Gisquet 07master:4fa772acbbac: dv: increase VLC reading bits to 10 [12:57] <cone-768> ffmpeg.git 03Christophe Gisquet 07master:beb944786e62: dvenc: mark encoder as intra [14:02] <kierank> http://obe.tv/about-us/obe-blog/item/19-fosdem-call-for-participation [14:02] <kierank> spam [14:02] <kierank> anyway please submit stuff to the fosdem media track [14:42] <thardin> are the samples at http://samples.ffmpeg.org/ffmpeg-bugs/ part of FATE? [14:46] <cehoyos> thardin: How would that be possible? [14:46] <cehoyos> The directory contains several G iirc [14:46] <cehoyos> And not all samples can be decoded bit-exact [14:47] <cehoyos> (I am not arguing we shouldn't create a testcase for every fixed bugs but this means a lot of work and we haven't done it so far.) [14:47] <thardin> I feel like there should be some test that goes "well, it shouldn't crash or hang on these files" [14:48] <cehoyos> Around 38GB [14:48] <thardin> maybe not FATE, but something [14:48] <cehoyos> As said, I don't disagree. [14:48] <cehoyos> It's just (as I regularly write on ffmpeg-user) that time is the only limiting factor. [14:48] <thardin> something to run before each release [14:48] <thardin> ic [14:48] <cehoyos> Do you know of a hang or crash regression? [14:49] <cehoyos> ... only limiting factor for FFmpeg development. [14:49] <thardin> I'm working on http://trac.ffmpeg.org/ticket/3278 and http://trac.ffmpeg.org/ticket/4040 [14:49] <thardin> and I changed a byte in 3278 to demonstrate a tricky case in my patch [14:49] <cehoyos> I know, 3278 is not a regression and 4040 neither hangs nor crashes. [14:50] <cehoyos> I mean: not a current regression [14:50] <thardin> it strikes me that the commit that reverts the hack for 3278 should probablt go after the proper fix [15:38] <thardin> cehoyos, michaelni: patches submitted to ML [15:40] <thardin> I'm slowly getting the feeling that the backward parsing should be made into its own loop instead of being mixed in with the old code [15:44] <wm4> why does mxf require backwards parsing? [15:44] <wm4> morbid curiosity here [15:45] <thardin> because you need to find all partitions [15:46] <Daemon404> ... don't people stream mxf too [15:46] <thardin> each partition points to the last one, and there's the Random Index Pack at the end which tells the position of each partition [15:46] <Daemon404> that soudns pervers.e [15:46] <thardin> yeah, it's so you can stream it to tape [15:46] <wm4> lol [15:46] <Daemon404> i see [15:46] <Daemon404> that sounds perverse, as i said [15:46] <thardin> there's nothing in the partition pack that tells where the *next* partition is [15:47] <thardin> so the demuxer goes "read RIP (if any), parse headerpartition, then the first body partition, then seek to footer partition and parse each PreviousPartition from it until you get back to the first body partition" [15:48] <thardin> then figure out where the essence is in each of those partitions [15:48] <thardin> then curl up in a corner and cry [15:48] <Daemon404> so in a truly streamign situation, you need to keep a buffer available... and there is no guarantee it will be a big enough buffer? [15:48] <nevcairiel> if you have the RIP, do you still need the backwards parsing? [15:48] <thardin> there's no need for any buffer [15:48] <thardin> nevcairiel: no, but there's no reason not to parse backward [15:48] <Daemon404> thardin, how would you do backwards parsing then [15:49] <Daemon404> if you can only read more byets [15:49] <Daemon404> bytes [15:49] <Daemon404> wothout buffered i/o i mena [15:49] <thardin> you gain nothing from not parsing backward. you still need to seek and parse each partition pack [15:49] <nevcairiel> if you dont need random access, you probably dont need to find all partitions right away? [15:49] <thardin> exactly [15:49] <Daemon404> ah ok [15:49] <Compn> cehoyos : a drop in replacement is fine, but why are the strings/description of the options identical ? [15:49] <thardin> that reminds me, I need to make sure mxfdec.c still works on a pipe [15:50] <thardin> yep [15:51] <thardin> I think there's still a few pathological cases where mxfdec will end up taking O(N²) time to parse a file [15:51] <cehoyos> Compn: Because they were copied. [15:51] <cehoyos> As said, I am happy that you see my point now. [15:52] <Compn> ive seen your point from the beginning :P [15:53] <Compn> i think i've pointed out attribution before [15:54] <cehoyos> There are many crystal-clear copyright violations in avconv, I don't think copying an option table and helplessly refactoring some timestamp code is particularly worrisome. [15:55] <Compn> well i'd like to resolve them in ffmpeg :P [15:55] <Compn> ehe [15:57] <Daemon404> gi/g 42 [15:57] <cehoyos> I am not sure there is anything to resolve: The original authors of the table all have agreed several times to the relicensing of their code to LGPL and they have not requested a copyright note in the file header. [15:57] <cehoyos> Good luck with trying to rewrite the pts code (a second time)! [15:58] <Compn> oh ok :P [15:58] <Compn> i forgot to take that into account lol [16:01] <cehoyos> But if it makes you feel better add a comment to the options table with the name of the authors - the copying was done so crudely that git blame -M -C does not help;-( [16:02] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:e70312dfc22c: avcodec/dxa: check dimensions [17:52] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:86e574928536: avformat/mvdec: Check size for validity in var_read_string() [17:52] <cone-768> ffmpeg.git 03Michael Niedermayer 07master:f1c21a200bcb: avformat/mvdec: Check size in read_table() for validity [18:27] <saste> ubitux: do you know the xBR original code? [18:28] <saste> because I see we have a 16MiB rgb2yuv table in the port [18:29] <wm4> 64 mib [18:31] <saste> could we use some shared table in lsws? [18:33] <michaelni> why doesnt it use the 3 lines of code to do rgb->yuv or call sws to turn the whole image into yuv or am i missing something ? [19:05] <ubitux> michaelni: i tried to inline the rgb2yuv convert in the function without a LUT [19:05] <ubitux> and it's a bit slower here [19:05] <ubitux> probably due to the /1000 [19:06] <ubitux> i tried to replace it with a shift but it's not exact with the refrence anymore, and it definitely affects visibly the output [19:06] <ubitux> the "xBR original code" for that part is supposed to be the same as HQx [19:06] <ubitux> for the rest i don't know [19:07] <ubitux> going to reply on the ml [19:11] <cone-768> ffmpeg.git 03Tomas Härdin 07master:fc1b89d887a5: mxfdec: Break out parts of mxf_read_header() into separate functions [19:11] <cone-768> ffmpeg.git 03Tomas Härdin 07master:37c36861550f: mxfdec: Parse PreviousPartition in mxf_seek_to_previous_partition() [19:11] <cone-768> ffmpeg.git 03Tomas Härdin 07master:1b17b64ee4d6: Revert "avformat/mxfdec: detect loops during header parsing" [19:11] <cone-768> ffmpeg.git 03Tomas Härdin 07master:b83affdc94a9: mxfdec: Merge last_partition and footer_partition [19:11] <cone-768> ffmpeg.git 03Tomas Härdin 07master:1a25c336aaaf: mxfdec: Tighten RIP length bounds in mxf_read_random_index_pack() [19:20] <ubitux> btw, if people think the performance impact is not important in comparison to the memory usage, i'm fine changing it [19:20] <ubitux> (regarding hqx lut) [19:21] <ubitux> now it might be interesting to check the asm sources of hqx [19:21] <ubitux> (the original ones) [19:21] <ubitux> and check if it wasn't doing this inaccuracy as well [19:21] <ubitux> in which case i would agree to change that [19:32] <jamrial> ubitux: how about adding both, and compile one or the other depending on CONFIG_SMALL? [19:37] <ubitux> isn't CONFIG_SMALL about the binary size? [19:40] <jamrial> i've seen it used also for memory usage [19:42] <ubitux> mmh [19:42] <ubitux> i just checked the original hqx sources [19:43] <ubitux> and it seems to be bitexact [19:43] <ubitux> so i can probably just adjust to it [19:52] <ubitux> and the original code was supporting only rgb565 [19:52] <ubitux> or something like that [19:52] <ubitux> alright i guess i'm going to change that [20:13] <ubitux> Y = (r + g + b) >> 2; [20:13] <ubitux> u = 128 + ((r - b) >> 2); [20:13] <ubitux> v = 128 + ((-r + 2*g -b)>>3); [20:13] <ubitux> meh. [20:26] <Daemon404> mmm mm [20:57] <amalia> michaelni I have installed Virtual box on my Fedora 20 host OS [20:58] <amalia> I installed a guest OS - Scientific Linux 6.4 and have tried variously to compile ffmpeg on it [20:58] <amalia> I install necessary dependencies like vpx, lame, etc. [21:01] <amalia> The compilation of ffmpeg takes like 30 minutes and leads to a fatal error when compiling a certain tagean.o file. So can't proceed with the make install and make distclean [21:03] <amalia> michaelni. Can I run the FATE tests on my Fedora host for now given that it is not on the list [21:03] <amalia> ? [21:06] <michaelni> is that a multi core CPU ? [21:06] <michaelni> 30min seem a bit long [21:08] <michaelni> also did compile run quicker on the host than the virtualboxed guest ? [21:08] <michaelni> also what is the exact error [21:09] <amalia> yeah it is a multi core CPU [21:09] <amalia> I did not try compiling on the host [21:09] <amalia> michaelni. Can I run the FATE tests on my Fedora host for now given that it is not on the list ? [21:10] <michaelni> are the multiple cores available to the guest os ? [21:10] <amalia> The ffmpeg compilation froze the guest OS and I couldn't retrive it [21:10] <michaelni> virtualbox (settings->System->Processor) allows to adjust the number of cores available [21:10] <amalia> I don't think there are multiple cores available for the guest [21:13] <michaelni> is the host running stable or it crashes and hangs too occasionally ? [21:14] <amalia> The host runs stably [21:15] <amalia> Was thinking of running the FATE test on the Fedora 20 host and submitting the results before continuiing to dabble with the guest OSs on Virtual box [21:15] <amalia> Don't know if this is okay for now [21:17] <michaelni> what version of virtualbox is that ? [21:17] <amalia> 4.3.18 [21:18] <michaelni> well, sure you can setup a fate client on the host if you like but we need to get something in some OS working in a virtualbox [21:22] <michaelni> amalia, also try to access/use the guest via ssh not the GUI of the guest, might be that theres some display driver issue in the guest or host [21:24] <michaelni> amalia, you can also ask on some virtualbox forum or irc channel about the scientific linux freeze, maybe someone there has a idea [21:25] <amalia> Okay [21:28] <ubitux> (note: even with the bithack, it seems slower than the lut) [21:33] <michaelni> amalia, also use confiure with --optflags=-O1 and if you dont need debuging --disable-debug that will cut build time down to half or so [21:34] <michaelni> and ccache should help if you build multiple times [21:34] <michaelni> with --cc='ccache gcc' [21:34] <amalia> Thanks michaelni. That's more comforting [22:03] <amalia> michaelni I just compiled ffmpeg on the fedora 20 host. How do I run the FATE tests ? Any links ? [22:05] <kasper93> https://www.ffmpeg.org/fate.html [22:09] <amalia> kasper93 I get this http://paste.kde.org/pmjqxtzfu [22:10] <amalia> Wich directory is the top-level source directory ? ffmpeg_sources or ffmpeg/ ? [22:20] <kasper93> ./configure --samples=fate-suite/ [22:20] <kasper93> make fate-rsync [22:20] <kasper93> make fate [22:21] <kasper93> "make fate-rsync SAMPLES=fate-suite/" this one indeed doesn't work. [22:21] <llogan> amalia: ffmpeg_sources is not in the ffmpeg source. is it a directory you created? [22:22] <amalia> I think my source is in ffmpeg [22:23] <kasper93> yeah, directory is ok, just wrong command. [22:23] <amalia> llorgan ffmpeg/ is in ffmpeg_sources [22:26] <amalia> kasper93 Much better result after running the ./config command http://paste.kde.org/pz7dkuuqx [22:50] <amalia> Do the tests always take this long ? [22:52] <Daemon404> you could run several at once with make -jN, where N is # of cores [22:52] <michaelni> amalia, here are some benchmarks: https://trac.ffmpeg.org/wiki/CompileBenchmarks [22:52] <michaelni> and yes use -jN [22:52] <amalia> Okay thanks [22:55] <wm4> interesting numbers [22:55] <wm4> so disabling compiler optimizations is much worse for fate than disabling almost all asm [22:55] <nevcairiel> disable-optimizations only goes to O1, doesn't it? [22:56] <ubitux> someone seems derping with a blackberry [22:57] <nevcairiel> hm nah guess it uses whatever is gcc default, it doesnt set any -O [22:58] Action: Daemon404 likes [FFmpeg-devel] new encoder function(urgent) [22:58] <wm4> it still got to do DCE [23:01] <wm4> apple started a trend with their "sent from my expensive piece of shit status symbol" message, huh [23:06] <kierank> Daemon404: yeah that made me lulz [23:06] Action: llogan likes that he also sent "new encoder function(urgent)" to libav-user, -devel, and -user [23:06] <llogan> with multiples sitting in mod queue [23:07] <llogan> he is from India. of course it's urgent. [23:08] <ubitux> wtf is that mike doing... [23:08] <llogan> ubitux: i set the "mod bit" to his account. [23:08] <kierank> llogan: it's a opw student, no? [23:08] <llogan> i don't know. [23:10] <llogan> kierank: looks like you're right. she I guess then. [00:00] --- Wed Oct 29 2014

1 0

ffmpeg.log.20141028
by burek 29 Oct '14

29 Oct '14

[00:04] <voip_> c_14, i tryed -f tee " rtmp://10.110.1.129/live/1|rtmp://10.110.1.130/live/ttt" [00:04] <voip_> result : Output #0, tee, to ' rtmp://10.110.1.129/live/1|rtmp://10.110.1.130/live/ttt': [00:04] <voip_> sorry result: Output #0, tee, to ' rtmp://10.110.1.129/live/1|rtmp://10.110.1.130/live/ttt': [00:05] <voip_> Output file #0 does not contain any stream [00:05] <voip_> But with 1 output it works [00:09] <voip_> c_14, http://pastebin.com/VDiiyLKU [00:11] <c_14> Get rid of the space after the '"' [00:12] <voip_> same error [00:14] <c_14> try adding [f=flv] in front of each of the rtmp:// parts [00:15] <voip_> with bracets ? [00:15] <c_14> yep [00:16] <c_14> "[f=flv]rtmp://10.110.1.129/live/1|[f=flv]rtmp://10.110.1.130/live/ttt" [00:17] <voip_> same eroor : Output #0, tee, to '[f=flv]rtmp://10.110.1.129/live/1|[f=flv]rtmp://10.110.1.130/live/ttt': [00:17] <voip_> Output file #0 does not contain any stream [00:17] <c_14> Throw a -map 0 in there somewhere? [00:18] <voip_> in my command i dont use map [00:20] <c_14> yep, I was asking you to add one to see if it helps. [00:20] <voip_> oh [00:20] <voip_> i will try [00:23] <voip_> so, it shuld be: -f tee -map 0:v -map 0:a "[f=flv]rtmp://10.110.1.129/live/1|[f=flv]rtmp://10.110.1.130/live/ttt" ? [00:23] <c_14> should work, the -f tee might have to be directly in front of the " though [00:30] <voip_> it works! -f tee -map 0:v -map 0:a "[f=flv]rtmp://A|[f=flv]rtmp://B" [00:30] <voip_> thanks [00:30] <c_14> np [00:30] <c_14> Not sure why the tee pseudo muxer drops the implicit maps though... [03:27] <Foxhoundz> What is the best parameters to convert an mp4 video into an mp3 [03:27] <Foxhoundz> while preserving the quality [03:27] <Foxhoundz> bit rate [03:27] <Foxhoundz> etc. [09:10] <BtbN> mp3 video? oO [10:06] <equanox> quit [10:34] <xata> halp [10:36] <xata> i have to ask this question again. what is the fastest way to resize mpeg2, preferrably without recoding? not-ffmpeg stuff is fine, but only ones, that run on linux. [10:38] <bove> What's the simplest way to set pixel aspect ratio to 1:1. Can I do setsar=0 ? [10:39] <BtbN> you can't resize a video without re-encoding it. [10:40] <bove> BtbN: This is during encode [10:45] <bove> Think I found the answer in something I wrote a while back: -vf 'scale=%i:%i,setsar=1' [14:39] <SilverCode> Hi, using the libav API, what is the best way to tell what the container is? [14:40] <SilverCode> hang on, nm found it [14:44] <khaosmarmoset> what's the best way? [14:48] <SilverCode> ok, I'm not sure if it the *best* way, but it works for me: AVFormatContext->iformat->name [14:59] <bencc1> when transcoding on Windows, does ffmpeg uses the GPU by default? [14:59] <bencc1> is it faster and more efficient to use the GPU instead of CPU? [15:03] <sacarasc> Depends what you mean by more efficient... It uses more power, but it's quicker to decode sometimes. [15:06] <bencc1> sacarasc: how can I turn it on and test? [15:06] <bencc1> sacarasc: I'm transcoding from raw video to h264 [15:06] <zybi1> hi [15:08] <zybi1> I use linux, for a film festival, which format should I try to make if my video work is shot in 720p h.264 [15:09] <zybi1> the festival asks for one of these: DCP, HDCAM, QuickTime ProRes 422 und 35mm [15:09] <zybi1> 35 mm definately not [15:09] <zybi1> prores is compressed again, that's why it's bad, too, right= [15:09] <zybi1> ? [15:10] <zybi1> HDCAM does that mean kind of analogue on a HDCAM cassette or is it a file format? on which medium? [15:10] <sacarasc> bencc1: I don't know, I've never done it. And usually, raw video can't be done on GPUs anyway. And encoding on GPUs suck. [15:10] <nasojlsu> HDCAM is digital on a casette [15:11] <bencc1> sacarasc: I'll use CPU for now. thanks [15:11] <nasojlsu> ProRes is a container format for sony, wrapped in quicktime .mov [15:12] <nasojlsu> I don't know if you will be able to do that in linux [15:12] <nasojlsu> i don't know what dcp is... [15:13] <iive> afaik, preres is codec and ffmpeg supports it [15:14] <nasojlsu> iive, you are correct [15:16] <nasojlsu> dcp looks like a special .mxf file. [15:16] <nasojlsu> quicktime preres is probably your best bet [15:16] <nasojlsu> now... time for some coffee [15:16] <iive> prorest, my mistake ;) [15:16] <iive> prores [15:16] <iive> yff [15:17] <nasojlsu> haha yeah. ProRes [15:17] Action: iive needs some of that coffee [15:22] <zybi1> https://en.wikipedia.org/wiki/Digital_Cinema_Package [15:22] <zybi1> that's what dcp stands for [15:43] <Tatou> hi \o/ [15:44] <Tatou> so i have a problem, someone can help me ? <3 [15:45] <Tatou> with this command : ffmpeg -i video.mkv -c:v libx264 -b:v 650k -c:a libfaac -b:a 192k -ac 2 -vf -s 720x404,subtitles=/home/yamaha/torrents/srt/subtitles.srt /home/yamaha/torrents/out/video.mp4 [15:45] <Tatou> the commande -s (scale) doesn't work [15:51] <alexvf> hi all, i want to use custom i/o to demux an incoming stream. What should i return in the read_packet callback when there is no data in the input if i want the callback to continue being called? [15:51] <alexvf> i'm using libavformat 2.2 [16:01] <alexvf> i'm returning AVERROR(EAGAIN) now and i reset AVIOContext's eof_reached and error before calling av_read_frame again [16:02] <alexvf> but, when some call to read_packet callback misses some data, i see noise in the decoded video [16:04] <Mavrik> alexvf, is it possible you're just not properly cutting packets? [16:04] <Mavrik> I've seen some decoders to be finnicky about that - e.g. returning full NAL units and stuff [16:04] <alexvf> it is, but i thought av_read_frame cuts them for me [16:04] <Mavrik> and things like that fact that packet should always have a full frame [16:05] <alexvf> if not, what's the point of av_read_frame? [16:06] <Mavrik> well look at the source [16:07] <Mavrik> it seems that av_read_frame does do that, but it first checks for a stream flag [16:07] <alexvf> i already did but i don't understand quite much [16:07] <alexvf> from what i see, returning AVERROR(EAGAIN) should do the magic, or better said, avoid the magic [16:10] <alexvf> is there some sample on how to use av_read_frame to read from network streams? i think i'm missing something [16:10] <alexvf> i found a custom i/o sample, but only for reading files [16:14] <Mavrik> well.. [16:14] <Mavrik> ffmpeg.c? [16:14] <Mavrik> or ffplay.c? [16:15] <iive> Tatou: I had no idea such syntax is still used, try with -vf scale=720:404,subttiles.... [16:15] <Tatou> ok i test [16:16] <alexvf> i'll take a look to ffplay.c [16:16] <alexvf> thanks Mavrik [16:16] <Tatou> No, it doesn't work :s [16:18] <Tatou> hum, ok [16:21] <w00ds> hello! [16:21] <w00ds> hope everyone is having a great day [16:21] <Tatou> live : it work [16:21] <w00ds> wanted to get some help with hardcoding subtitles in an mp4 [16:21] <Tatou> \o/ [16:21] <Tatou> I test [16:21] <w00ds> trying to use the following command: ffmpeg -i video.avi -vf subtitles=subtitle.srt output_video.avi [16:22] <w00ds> it spits this out though [16:22] <w00ds> https://www.irccloud.com/pastebin/Nzm04tYl [16:23] <w00ds> https://www.irccloud.com/pastebin/7noknHaL [16:23] <w00ds> that is the configuration for ffmpeg im using. [16:25] <kepstin-laptop> w00ds: first check the obvious thing, is the file 'subtitles.srt' in the current directory, with permissions such that it can be read? [16:25] <kepstin-laptop> er, subtitle.srt, whatever. [16:25] <w00ds> yes it is [16:25] <relaxed> and use a normal pastebin site [16:25] <w00ds> running the commands as root [16:25] <w00ds> but yes, it's chmod 777 * [16:26] <w00ds> relaxed: this one is built into my irc client [16:26] <w00ds> i copy paste into irc, it creates a pastebin for me [16:27] <relaxed> that's no excuse [16:28] <w00ds> not an excuse, just educating you [16:31] <relaxed> try one that doesn't require javascript or ask if it can open irc links, which seems viral [16:32] <w00ds> its okay that my question stumps you relaxed [16:33] <w00ds> just keep the chatroom clear for people who do have intelligence though, please. [16:33] <iive> Tatou: good :) [16:33] <Tatou> than's a lot, it work [16:33] <Tatou> I test now x) [16:33] <relaxed> w00ds: you're using my build, smartass [16:36] <w00ds> i'm using your build? [16:37] <relaxed> http://johnvansickle.com/ffmpeg/ [16:37] <w00ds> thats probably why it's not working, haha [16:38] <iive> well, the log says it can't file the subtitle.srt file [16:38] <w00ds> lets compile from scratch, get something actually working [16:38] <w00ds> iive: think it's a bad .srt? [16:38] <iive> more like wrong directory or permissions [16:38] <w00ds> with mencoder it was able to read the file, but the video/audio was all corrupted/messed up ... this is why i like ffmpeg better [16:39] <w00ds> im using my own pastebin [16:39] <w00ds> thanks, but no thanks relaxed [16:40] <iive> your own pastebin demands running a bunch of external scripts to show something... [16:40] <iive> not good. [16:40] <relaxed> iive: exactly [16:41] <w00ds> thats unfortunate [16:41] <w00ds> iive, permissions all 777 [16:41] <w00ds> running as root [16:41] <w00ds> gave full paths to files/srt/mp4 [16:41] <w00ds> same outcome [16:42] <kepstin-laptop> ... wow, how in the world can someone be so bad at making a pastebin that it shows a blank white page without javascript. [16:43] <relaxed> and when you do enable javascript it asks to be the default for opening irc links [16:45] <iive> i missed that... good the browser ask first. [16:46] <iive> w00ds: try to give it absolute pathname to the file, if it fails, trace the file io calls using `strace` [16:46] <w00ds> iive: i did provide absolute paths [16:47] <w00ds> ./root/testing/ffmpeg-2.4.2-64bit-static/ffmpeg -i /vault2/downloads/video.mp4 -vf subtitles=/vault2/downloads/subtitle.srt /vault2/downloads/output_video.mp4 [16:49] <relaxed> fflogger gives good advice for getting help [17:17] <ac_slater_> hey all. I need some libavformat advice. I want to mux two sources (video and data) into an mpegts stream. Any advice on creating program streams? [17:21] <ac_slater_> ah I found some stuff... nevermind me [17:24] <alexvf> Mavrik: neither ffmpeg.c or ffplay.c use custom I/O [17:27] <alexvf> what i see is that AVERROR(EAGAIN) looks like the legit way to mean that av_read_frame has to continue being called [17:34] <edibile> is it possible to stream to multicast ip using solely ffmpeg or is ffserver also required? [17:35] <Mavrik> for multicast ffmpeg is enough [17:35] <Mavrik> udp://224.etc range should just work [17:36] <ac_slater_> edibile: are you just using udp as a transport or something like RTP/RTSP ? [17:37] <edibile> was planning to use udp but am not entirely sure what the implications of using that over rtp/rtsp are [17:37] <ac_slater_> edibile: RTP and RTSP and complicated. If you control both sides of the transmission, then it doesnt really matter [17:38] <Mavrik> uh [17:38] <Mavrik> that makes no sense :P [17:38] <ac_slater_> Mavrik: haha yea it does [17:38] <Mavrik> RTSP is a twoway protocol that kinda doesn't work over multicast :) [17:38] <edibile> i'm using ffmpeg to combine multiple ip camera streams into a matrix and then want to multicast that combined stream so that other devices on my lan can access the stream [17:38] <ac_slater_> Mavrik: RTSP delivers via RTP... the RTSP handshaking is TCP [17:38] <Mavrik> edibile, then just pack it into mpeg2-ts and stream it as UDP stream [17:39] <Mavrik> edibile, all devices on the subnet will be able to play the video as long as your router isn't doing anything funky [17:39] <Mavrik> dealing with RTP/RTSP will just give you headaches in that kind of environment [17:39] <ac_slater_> ^ which is what I said [17:40] <edibile> ok, yeah, i'm not sure what's happening [17:41] <ac_slater_> ill shut up. But I was asking if you were using RTP or planning on it. Since you said 'multicast' in general. [17:42] <edibile> it works fine writing to a local file but when i change to "-f mgegts udp://224.1.1.1:1234" i can't connect [17:42] <Mavrik> I hope that's a typo :D [17:42] <Mavrik> can you do the pastebin? [17:42] <Mavrik> ac_slater_, yeah, RTP is a pain -_- [17:43] <Mavrik> even though, broadcasting SDP over SAP for multicast kinda works [17:44] <edibile> not sure what i messed up, but i'll pastebin [17:44] <ac_slater_> edibile: multicast is odd. 224.1.1.1 is valid but odd [17:44] <edibile> ok, 239.101.101.1:1234 [17:45] <ac_slater_> also remember that 224.0.0.1 is the default group... and it's special [17:49] <edibile> http://pastebin.com/px6axJd1 [17:49] <Mavrik> -_- [17:49] <Mavrik> you'll save yourself ALOT of pain if you always explicitly set codecs. [17:49] <Mavrik> you also didn't paste any important bits: the output. [17:50] <Mavrik> right now we don't even know what format you're streaming in at all [17:54] <edibile> ok, good point, deleted something i shouldn't have [17:55] <edibile> seems it is using RTP though [17:57] <Mavrik> makes no sense. [17:57] <Mavrik> if you don't tell it to use rtp, it won't. [17:57] <Mavrik> you're probably seeing RTP logs from the input since you're consuming RTSP streams ;) [17:58] <edibile> yeah, ok, probably right =) [18:00] <edibile> http://pastebin.com/cLr1Cb6N [18:05] <Mavrik> dude. [18:06] <Mavrik> why aren't you pasting the part with ALL THE STREAM INFORMATION and THE ACTUAL ERROR?! [18:13] <edibile> sorry, didn't know you would want that (first time using ffmpeg) [18:14] <edibile> http://pastebin.com/FKZuufnw [18:39] <edibile> well, it's working fine now [18:39] <edibile> maybe i wasn't letting it load long enough [18:41] <edibile> thanks for the help [18:50] <voip__> Hello guys, plesae help me with 1 stream processing and multiple IP outputs. In ffmpeg status i have someting like frame= 412 fps= 26 q=-1.0 Lsize=N/A time=00:00:16.74 bitrate=N/A dup=19 drop=0. My output hase audio, but no video. [18:50] <voip__> http://pastebin.com/wByE8CB8 [19:30] <Diogo> ffmpeg -i http://<URL>/Teste.mov -c:v libx264 -f mp4 final.mp4 [19:31] <Diogo> anyone know why mov file request lots of HTTP 206 request to encode my video.. [19:31] <Diogo> with other files works wheel [19:50] <Diogo> ? [20:02] <voip__> any help ? [21:01] <gilbahat> Hi, I would like some advice in choosing an intermediate editing codec. I need something which can be decoded and encoded by both ffmpeg and vfw. I prefer something with good enc/dec speed and not too high a bitrate, willing to take some lossyness - mjpeg is the baseline to compare to. [21:03] <gilbahat> I guess I would start with the best recommended for ffmpeg and then see if I can hunt down a vfw codec for it [21:06] <kepstin-laptop> do you need an intra-only codec? [21:09] <llogan> gilbahat: i've used Ut Video with good results as an intermediate, but it may nto fit your "not too high a bitrate" requirement. it's lossless. [21:10] <gilbahat> great question. I believe I do but I may just get away without it: the files will be edited, but are small (relatively speaking, up to 10 minute of raw footage at most), so maybe its not such a big deal. I believe I will need to benchmark to get a decisive answer. [21:12] <kepstin-laptop> if you're considering mjpeg "to high bitrate" and are already using it in a lossy mode, then you'll probably have to go to an inter-coded lossy format. [21:17] <gilbahat> kepstin-laptop: I dont consider the bitrate of mjpeg too high. the problem is, its so outdated that I believe that I could get much better quality for the same bitrate with a modern codec. but as far as intermediate codecs go, most seem to have higher bitrates than mjpeg but significantly better quality (DNxHD, ProRes, Cineform (not sure)), so I will take something which has a bitrate increase for a significant quality increase. Not [21:17] <gilbahat> why theres no demand for that - mj2k didnt seem to catch so well. motion jpeg XR never saw anything more than spec& [21:19] <kepstin-laptop> hmm, most of the newer codecs designed specifically for intermediate work use higher bit depth color, which (assuming the entire processing chain uses high bit depths internally) reduces quality loss over multiple generations. [21:19] <gilbahat> kepstin-laptop: the big question about using an inter-coded format is whether an editing software reading it will seek efficiently (i.e. use key frames, given a format that allows me to set the ratio of them) [21:23] <kepstin-laptop> gilbahat: basically, you're gonna hit a quality per bitrate vs. encoding speed tradeoff, and you need to figure out which codec balances that tradeoff properly for your use case :/ [21:24] <kepstin-laptop> newer codecs can do better quality at a given bitrate, sure, but they do that by spending more time encoding (which is possible because computers have overall gotten faster) [21:25] <gilbahat> kepstin-laptop: its a classic pick 2 isnt it. I have 2 options the way I see it: (1) look at modern popular intermediate codecs and see if any of them threw me a bone in the form of a lossy mode or (2) see if any modern non-intermediate codec offers some intra-only mode with fair bitrates. [21:27] <gilbahat> kepstin-laptop: I know I am not exhausting my processing capacity encoding to mjpeg, so I can definitely get by with a codec that encodes slower. but it looks like that some of the modern things to do are supersets of approaches used in mjpeg - faster and better, because they are smarter so to speak. [21:31] <voip__> guys, plesae help me with 1 stream processing and multiple IP outputs. In ffmpeg status i have someting like frame= 412 fps= 26 q=-1.0 Lsize=N/A time=00:00:16.74 bitrate=N/A dup=19 drop=0. My output has audio, but no video. [21:31] <voip__> http://pastebin.com/wByE8CB8 [21:31] <kepstin-laptop> note that alongside other dedicated intermediate codecs, h264 (via x264) is also interesting to consider; it can be run intra-only, 4:4:4 sampling, high bit depths, etc, and the encoder's well-tuned [21:33] <gilbahat> you know what, I guess that makes the most sense - I can get fair encoding speed, fair bitrates and definitely VFW support and quality encoder. [21:55] <LucasG> Hi guys. I have problem with ffmpeg + ffserver. I would like to send stream via http but i have error Connection reset by peer. Detailed log: http://pastebin.com/kNcADER6 - I think, maybe problem is in network (but it is not firewall problem). Maybe anyone have working ffserver instance and would like to try recive my stream in own ffserver? [21:57] <LucasG> i will know that problem is on ffmpeg or ffserver side, or maybe both [22:42] <kepstin-laptop> for what its worth, I poked some of the irccloud folks on twitter, and their pastebin is actually usable (sort of) without javascript now [22:42] <kepstin-laptop> it just has a link to the raw, but whatever. [22:48] <LucasG> More detailed pastebin: http://pastebin.com/D9LhWrUc [23:00] <maui> hi all! can someone pls tell me how to use a distortion filter in a specific area [23:11] <astr> I would like to save a video stream from a hd web cam and a seport audio source. I found the audio is not in sync. Is there a option to set an audio delay and would this fix the issue? I assume it would and that I would only have to update the audio delay number if I changed my audio DAC? [23:12] <llogan> maui: maybe crop your desired area, then overlay the result on the undistorted version [23:12] <astr> this would mean I could live stream in hd if I have the band width :D [23:23] <LucasG> Any one can help with one little test with its ffserver? [23:23] <llogan> might want to try the ffmpeg-user mailing list if you don't get help here [23:24] <LucasG> thanks, i will try [23:41] <voip__> hi llogan, pls help me with 1 stream processing and multiple IP outputs. In ffmpeg status i have someting like frame= 412 fps= 26 q=-1.0 Lsize=N/A time=00:00:16.74 bitrate=N/A dup=19 drop=0. My output hase audio, but no video. [23:41] <voip__> http://pastebin.com/wByE8CB8 [23:48] <llogan> voip__: sorry, i have almost no experience with rtmp protocol [23:49] <voip__> my problem with 2 outputs , i tink i am using vrong command [23:53] <deivor> does it work if you output to one flv file? [23:54] <llogan> do you need to add "-flags +global_header"? [23:58] <voip__> deivor, you mean my case [23:58] <voip__> ? [23:58] <deivor> yes [00:00] --- Wed Oct 29 2014

1 0

ffmpeg-devel.log.20141027
by burek 28 Oct '14

28 Oct '14

[00:02] <ubitux> i need directions for what people want for bitmap subtitles [00:03] <ubitux> basically, what AVSubtitles lack bitmap wise [00:03] <ubitux> i'll backlog tomorrow, gn [00:06] <J_Darnley> The only thing I want is OCR but I guess that comes after all the other hard desisions. [00:42] <wm4> ubitux: pixel format, also having a common palette for all sub-bitmaps instead of one for each might be helpful [02:31] <rcombs> wm4: ubitux: potentially, handling bitmap data and text data in the same stream [03:21] <wangl> capture images frame from the camera, and every 66msframe rate 15 call av_read_frame,spend time 0ms. [03:22] <wangl> if every 40ms(frame rate 25) call av_read_framespend time 50ms [03:22] <wangl> why? [03:39] <wangl> i have a problem. call av_read_frame, spend time 60ms, why ?? [03:49] <rcombs> slow I/O? [03:50] <rcombs> maybe blocking until the camera sends the next frame? [04:59] <cone-667> ffmpeg.git 03Michael Niedermayer 07master:4898440f6bd1: Move get_avc_nalsize() and find_start_code() to h264.h [04:59] <cone-667> ffmpeg.git 03Michael Niedermayer 07master:3b678da5e386: avcodec/h264: simplify find_start_code() [04:59] <cone-667> ffmpeg.git 03Michael Niedermayer 07master:69a9a90d2ef7: avcodec/h264_parser: rewrite the parse_nal_units() loop logic based on h264.c [05:08] <wangl> @rcombs if av_read_frame is waiting for the next frame, how to make camera to send frame fast? [05:25] <rcombs> wangl: that depends on your camera, I'd expect [05:25] <rcombs> how are you getting input from the camera? [05:32] <wangl> @rcombs: [05:32] <wangl> pFormatCtxSrc_ = avformat_alloc_context(); [05:32] <wangl> AVDictionary* options = NULL; [05:32] <wangl> av_dict_set(&options, "framerate", "25", 0); [05:32] <wangl> AVInputFormat * inputFormat = av_find_input_format("dshow"); [05:32] <wangl> avformat_open_input(&pFormatCtxSrc_, "video=Integrated Camera", inputFormat, &options); [05:32] <wangl> avformat_find_stream_info(pFormatCtxSrc_, &options) [05:32] <wangl> avcodec_find_decoder [05:32] <wangl> avcodec_open2 [05:32] <wangl> av_read_frame [05:35] <wangl> i use the "dshow" [05:44] <rcombs> please use a pastebin site for pastes longer than ~3 lines [05:45] <rcombs> and I know next to nothing about dshow, so sorry to say I probably can't help [05:46] <wangl> that's ok, thank you rcombs [11:21] <ubitux> J_Darnley: yeah that's orthogonal [11:21] <ubitux> wm4: about the pixel format, should we still continue with a very small subset of pixel formats? [11:21] <ubitux> rcombs: yeah right, makes sense [11:24] <wm4> would you use AVFrame or not? [11:24] <ubitux> i still don't know [11:25] <ubitux> it's not exactly easy when you have multiple rectangles [11:25] <ubitux> we can use the extended data but that's starting to get ugly [11:25] <wm4> you need all that extra stuff of course (position etc.) [11:28] <ubitux> yeah exactly [11:28] <ubitux> so my question really is how you want to use the subtitles api [14:16] <cone-536> ffmpeg.git 03Michael Niedermayer 07master:852aaead1fc2: avcodec/hevc_ps: Check default display window bitstream and skip if invalid [15:59] <arwa> Is c in rgb2yuv the uint32 rgb value? [16:02] <ubitux> yes [16:03] <arwa> okay thanks [16:03] <arwa> :) [16:50] <reynaldo> does anyone remember when did FFmpeg got into Debian/experimental? [16:54] <thardin> I'm wtf:ing pretty hard at that mxfdec patch [16:59] <wm4> isn't that normal for mxf [17:00] <thardin> true [17:00] <thardin> hopefully I'll have time later this evening to develop a proper patch. context = https://ffmpeg.org/pipermail/ffmpeg-devel/2014-October/164589.html [17:03] <ubitux> afaict it randomizes the last_pos [17:03] <ubitux> so the next comparison mismatch [17:03] <ubitux> i must say it's horrible [17:03] <ubitux> :D [17:04] <thardin> considering the demuxer already knows what direction it's scanning it should be enough to check whether the position is increasing/decreasing or not [17:05] <thardin> something like last != cur && (last < cur ^^ mxf->parsing_backward) [17:14] <cone-536> ffmpeg.git 03Thilo Borgmann 07master:ed2e97ce6a7a: lavd/avfoundation: Fix compilation for non MAC OS devices by conditional compilation of screen capture capabilities. [18:25] <arwa> hey I am getting segmentation fault for the code I have written. [18:27] <wm4> use gdb or valgrind [18:35] <arwa> I tried debugging it, but It is giving me error for the same function that I copied from hqx filter. [18:36] <arwa> It is showing me that I am accessing a value which is exceeding the bounds, but I printed the value, and I found that its within range [18:36] <arwa> I am not sure why is this happening ? [18:37] <wm4> the _pointer_ is out of range, not the value that you get when dereferencing the pointer [18:37] <wm4> and accessing it will often appear to work fine, but it's still invalid [18:37] <wm4> but in this case, apparently it crashed [18:39] <arwa> The code is here -- " http://pastebin.com/edit.php?i=k4qs1vgu " [18:40] <arwa> In the init function in this code its giving me seg fault for the value of c at 1st iteration. [18:42] <arwa> Here the value of c is acting as a pointer, which is accessing the rgbtoyuv value. [18:42] <wm4> the link doesn't work [18:45] <arwa> okay, try this - " http://privatepaste.com/1cb1aa3872 " [18:49] <wm4> I see nothing too obviously wrong - I'd guess it means you're almost there, but you'll have to debug it yourself [18:50] <wm4> (also it's not like I have bug-detecting eyes) [18:50] <wm4> the 64MB rgbtoyuv lookup table it a bit strange though [18:51] <arwa> What I find strange is I copied this part of the code from hqx filter...!! How can it possibly give me fault?! [18:52] <wm4> maybe it was written with different assumptions in mind [18:52] <arwa> yeah probably. [18:53] <arwa> I will resume looking for the errors, and figure out what is going wrong. [19:04] <ubitux> arwa: your code needs the AVClass pointer [19:04] <ubitux> you still haven't read doc/writing_filters.txt [19:05] <ubitux> mmh actually, my bad, you still don't have any option so it should be fine [19:06] <ubitux> your code doesn't have a priv_size field [19:06] <ubitux> in AVFilter [19:06] <ubitux> so the context is not allocated [20:07] <cehoyos> reynaldo: FFmpeg was uploaded to experimental 2014-09-16 [20:07] <cehoyos> See https://packages.qa.debian.org/f/ffmpeg.html [20:10] <reynaldo> cehoyos: cool, thanks a lot pal [20:13] <cehoyos> You are welcome! [20:21] <arwa> ummm....so, what should I do? [20:21] <ubitux> look at priv_size in hqx [20:25] <ubitux> https://www.indiegogo.com/projects/mox-file-format wut [20:25] <ubitux> > MOX is going to use the MXF container (SMPTE 377M) [20:26] <ubitux> > Video formats will be Dirac, OpenEXR, DPX, PNG, and JPEG. [20:26] <ubitux> :/ [20:28] <nevcairiel> sounds like someone with no ties to the actual open source media world came up with those [20:28] <ubitux> i see opus though [20:29] <nevcairiel> his history seems to be taking opensource stuff and plugging it in closed source software, not any actual experience with crazy format details of mxf <.< [20:30] <wm4> "MOX will read and play consistently on Mac, Windows, Linux, or any other platform. This is because MOX will be an open format based on open standards. It will include an open source C++ library to ensure that any program will be able to easily support MOX files now and forever." [20:31] <bencoh> meh [20:31] <wm4> how do they even get money [20:31] <nevcairiel> so, he wants to implement a new library that reads and decodes all of this? :D [20:31] <bencoh> "yeah, sure, whateverbullshit" [20:31] <wm4> it's like I'm missing a simple trick to get rich here [20:32] <nevcairiel> in $20.000 no less? [20:32] <nevcairiel> well good luck! [21:01] Action: Compn sets mode +b irccloud.com ... [21:16] <cone-536> ffmpeg.git 03Michael Niedermayer 07master:3b20ed85489a: avcodec/sgidec: fix linesize for 16bit [21:16] <cone-536> ffmpeg.git 03Michael Niedermayer 07master:a050cf0c451b: avcodec/sgidec: fix count check [21:35] <gnafu> wm4: "Open source media software developers hate him! Find out this one crazy trick to get $20,000 like that!" [21:38] <J_Darnley> I was tempted to post that xkcd comic into that email thread. [21:45] <cone-536> ffmpeg.git 03Luca Barbato 07master:a6674d2e7771: xcbgrab: XCB-based screen capture [21:45] <cone-536> ffmpeg.git 03Michael Niedermayer 07master:a38a1d516fe6: Merge commit 'a6674d2e7771dbf7a4a5556f5e126be83cadac96' [22:25] <cone-536> ffmpeg.git 03Vittorio Giovara 07master:090c67d586e3: matroskaenc: write correct Display{Width, Height} in stereo encoding [22:25] <cone-536> ffmpeg.git 03Michael Niedermayer 07master:f769671f86ed: Merge commit '090c67d586e3916f9acc49e010b6389d07f97153' [23:04] <cone-536> ffmpeg.git 03Vittorio Giovara 07master:b46b233baffc: filmstripdec: avoid integer overflow [23:04] <cone-536> ffmpeg.git 03Michael Niedermayer 07master:76b9043e903c: Merge commit 'b46b233baffc2076a1a17a264ba9553ae0d4878f' [23:04] <cone-536> ffmpeg.git 03Michael Niedermayer 07master:9612dcd6b24f: avformat/filmstripdec: Fix several integer overflows [23:33] <cone-536> ffmpeg.git 03Vittorio Giovara 07master:e6c66f1e4ea1: bethsoftvid: check return value and clean memory [23:33] <cone-536> ffmpeg.git 03Michael Niedermayer 07master:5ee6527c4386: Merge commit 'e6c66f1e4ea19a3aa6ed999c5cb92e1b682c7600' [23:33] <cone-536> ffmpeg.git 03Michael Niedermayer 07master:0894b9405c0c: avformat/bethsoftvid: print error in case the side date failed to be allocated [23:49] <cone-536> ffmpeg.git 03Tomas Härdin 07master:11467ecf5197: mxfdec: reduce loop bound in mxf_read_pixel_layout() [23:49] <cone-536> ffmpeg.git 03Michael Niedermayer 07master:d4e8cc35b3d9: Merge commit '11467ecf519788f2e2f49c63a17eb600e618489e' [00:00] --- Tue Oct 28 2014

1 0

ffmpeg.log.20141027
by burek 28 Oct '14

28 Oct '14

[00:32] <spectralsun> trying to reencode with this command "ffmpeg -i ns-0f483bb1-59d2-4014-aef6-17d1bb0450b1.mov -acodec copy -g 15 tmp.mov" [00:32] <spectralsun> but it's only making a video of a few seconds long [00:32] <spectralsun> does this message mean anything: "[mov @ 0x39e6480] Packet with invalid duration -30684 in stream 1" ? [00:32] <spectralsun> the video is like a minute and a half, but tmp.mov is only a few seconds [00:39] <joonas_> I'm gonna take a wild guess and say it means your source file is broken. [00:40] <joonas_> you actually get that when writing an invalid packet [00:41] <joonas_> if you turn on debug logging you should see a log entry with the fields in question. [00:43] <sanooj> although if you see that then it looks like it will estimate the packet duration from other data. of course it might be way off though. [03:25] <shevy> anyone knows what I do wrong? [03:25] <shevy> [vorbis @ 0xb0e1880] The encoder 'vorbis' is experimental but experimental codecs are not enabled, add '-strict -2' if you want to use it. [03:26] <shevy> ffmpeg -strict -2 -i foo.wma -acodec vorbis -aq 100 output.ogg [03:28] <sanooj> try adding -strict -2 after your -i [03:28] <sanooj> foo.wma [03:28] <mark4o> shevy: use libvorbis, not vorbis [03:29] Action: sanooj is chaos monkey! [03:30] <sanooj> turns out one's uvcvideo code works a lot better if you actually read the specs instead of intuiting what the ioctls do [03:31] <sanooj> granted, I didn't really want to do that, but whatever... [03:48] <shevy> sanooj thanks that worked a bit better, but still an error http://pastie.org/pastes/9677843/text [03:48] <shevy> "Current FFmpeg Vorbis encoder only supports 2 channels." [03:50] <sanooj> indeed. your input seems to have only one! [03:51] <sanooj> I'm sure there's a widget in ffmpeg to duplicate your mono into stereo, but I only started ffmpeging this weekend [03:51] <shevy> hmm [03:52] <shevy> can I just copy the one channel mono and keep it like that? [03:52] <shevy> oh wait [03:53] <shevy> switching from -acodec vorbis to -acodec libvorbis fixed it [03:53] <sanooj> see, people who actually know said so. ;) [03:54] <shevy> I assumed you to be wiser because you wrote more than he did [03:54] <khaosmonkey> you may have been able to do a channel mapping that duplicates your mono thing, but I'm not sure. [03:54] <khaosmonkey> haha [04:12] <khaosmarmoset> woot! I can now tune uvc tunables in v4l2.c! [04:37] <khaosmarmoset> and youtube is happy! [04:38] Action: khaosmarmoset calls it a job sloppily done! [04:38] <khaosmarmoset> 'night [07:18] <FearlessHawk> Is it possible to repair media that was not deinterlaced before it was converted into progressive? [07:26] <DrSlony> hi JEEB, has -preset been deprecated in git? [07:36] <DrSlony> ./programs/code-ffmpeg/ffmpeg -i MOV_0818.mp4 -ac 2 -c:a libfdk_aac -b:a 128k -c:v libx264 -preset slower -crf 23 -level 4.1 -maxrate 512k -bufsize 2560k ~/2014-10-26.mp4 [07:36] <DrSlony> Unrecognized option 'preset'. [07:36] <DrSlony> Error splitting the argument list: Option not found [07:39] <ruby_on_tails> whats the difference between ffmpeg and avconv? [07:39] <kevmitch> avconv is the command line program of the libav fork of ffmpeg [07:39] <kevmitch> they have roughly the same functionality [07:40] <JEEBsv> DrSlony: no. but it is libx264 specific [07:40] <JEEBsv> -codecs | grep x264 [07:40] <DrSlony> JEEBsv im guessing im lacking some ./configure parameter [07:41] <JEEBsv> --enable [07:41] <JEEBsv> argh [07:41] <JEEBsv> fuck touchscreen keyboards [07:41] <DrSlony> yes [07:41] <JEEBsv> --enable-gpl --enable-libx264 [07:42] <ruby_on_tails> why cant someone make a format which is supported everywhere and there is a single command to convert videos to that format [07:42] <ruby_on_tails> working with videos is so confusing [07:42] <ruby_on_tails> i just transcoded a video using avonv and its size is 1/5th of that of the original without any quality difference, i am so scared now [09:02] <AminCX> Hello, I'm trying to create a mp4-dash mpd file with mp4box and it give me an error "[DASH]: Segment duration variation is higher than the +/- 50% allowed by DASH-IF (min 0.333, max 6) - please reconsider encoding", i'm using this command "ffmpeg -i test2.mp4 -codec:v libx264 -vprofile baseline -preset veryslow -crf 1 -threads 0 -maxrate 235k -bufsize 235k -vf scale=-2:240 -an -y test2-240-235k.mp [09:02] <AminCX> 4" to encode the files, what is wrong here ? [09:38] <khaosmarmoset> Your computer is TOO SLOW to encode these files! Or it could be an audio driver problem. [09:44] <khaosmarmoset> AminCX: seriously though, I of course have no clue. sounds like dash and libx264 disagree on something. have you tried asking the x264 guys? [09:51] <AminCX> khaosmarmoset: computer sounds good, i7 4770, no my first thought was that if anyone should know its ffmpeg developers. [09:55] <BtbN> AminCX, you need to set a fixed gop length. [09:55] <BtbN> not too long, 2 or 4 seconds [09:57] <AminCX> BtbN: Thank you, i'm trying it now. [10:04] <DrSlony> JEEBsv the encoded video plays with about 10 frames per second using mpv, but perfectly fine when i play it using ffplay or via the web browser. mpv plays every other video perfectly, so im confused whats happening... do you have any ideas? [10:07] <DrSlony> errr.. ffprobe shows the encoded video has "59.42 fps" but the source video has "29.70 fps" [10:07] <DrSlony> ./programs/code-ffmpeg/ffmpeg -i MOV_0818.mp4 -t 10 -ac 2 -c:a libfdk_aac -b:a 128k -c:v libx264 -preset slow -crf 23 -level 4.1 ~/2014-10-26_10_crf23_l41_p.mp4 [10:16] <xata> Hello. [10:17] <xata> What is the fastest way (less cpu-hungry) to decrease the video's file size? Disregarding the quality, just do it. [10:18] <xata> And do it fast [10:19] <xata> Maybe you can advice some channel/forum/knowledge base so i could read myself? I understand that this is not totally ffmpeg question, but i will use ffmpeg for this, so here i am :) [10:20] <DrSlony> ffmpeg -i in.mp4 -an -c:v libx264 -preset ultrafast -crf 25 out.mp4 [10:20] <DrSlony> xata ^ one possibility [10:23] <xata> DrSlony: Unrecognized option 'preset'. [10:24] <xata> DrSlony: :C [10:24] <DrSlony> try -vpre instead [10:30] <xata> DrSlony: now it says it does not understand -crf [10:30] <DrSlony> either youre using an old ffmpeg or it doesnt have the right stuff compiled into it [10:32] <xata> DrSlony: yeah. looks like second one. thanks anyway [10:32] <DrSlony> JEEBsv i dont know why the fps of the video is changed after transcoding [11:45] <Tatou> Hi everybody [12:18] <AminCX> BtbN, would you please suggest me a way to set a fixed gop length ? i tried -g and -keyint but i'm not sure. [12:21] <c_14> -force_key_frames expr:gte(t,n_forced*5) <- will force a keyframe every 5s [12:24] <AminCX> c_14, thank you, i'm trying it now. [12:28] <Tatou> hi, do you know why it doesn't work ? :p [12:28] <Tatou> ffmpeg -i video.mkv -vf scale=720:400 subtitles=subtitle.srt out.mp4 [12:28] <Tatou> i have an error message : [12:28] <c_14> scale=720:400,subtitles=subtitle.srt [12:28] <Tatou> okay i test [12:29] <Tatou> thanks c14 :) [12:52] <AminCX> c_14, i tried this "ffmpeg -i test2.mp4 -codec:v libx264 -vprofile baseline -preset slow -force_key_frames expr:gte(t,n_forced*1) -threads 0 -maxrate 235k -vf scale=-2:240 -an -y test2-240-235k.mp4" and still got that problem with mp4box, it says "[DASH]: Segment duration variation is higher than the +/- 50% allowed by DASH-IF (min 0.333, max 3) - please reconsider encoding" with -dash 3000... [12:52] <AminCX> ...parameter. [12:56] <c_14> hmm, no clue then. Sorry. [13:29] <Tatou> I would like to pass AC3 6 channels to AAC 2 channels (stereo) with a video, someone know the command ? I didn't find [13:45] <relaxed> Tatou: what's the output of -> ffmpeg -codecs 2>/dev/null|grep DEA.*aac [13:46] <Tatou> hum [13:46] <Tatou> i don't understand *_* [13:46] <relaxed> run the command and paste its output here [13:47] <Tatou> okay :p [13:49] <Tatou> I have this response : DEA.L. aac AAC (Advanced Audio Coding) (encoders: aac libfaac ) [13:49] <relaxed> ffmpeg -i input -c:v copy -c:a libfaac -b:a 192k -ac 2 output [13:51] <Tatou> okay I test, thank's for the response [13:57] <Tatou> it work :p [13:57] <Tatou> and with that [13:57] <Tatou> I would like to burn subtitles [13:58] <Tatou> so [13:58] <Tatou> i test this command : ffmpeg -i video.mkv -vf scale=720:400,-c:v copy -c:a libfaac,-b:a 192k,-ac 2,subtitles=sub.srt output.mp4 [13:58] <Tatou> but it doesn't work, do you know why ? :p [13:59] <Mavrik> em [13:59] <Mavrik> it's YOU that has the error output. [13:59] <Mavrik> how would WE know? :P [14:00] <Tatou> hum [14:00] <Tatou> xD [14:15] <Tatou> so i test an anover option [14:15] <Tatou> i made this command : ffmpeg -i input.mp4 -c:v libx264 -preset fast -crf 22 -c:a copy subtitles=srt.srt output.mp4 [14:15] <Tatou> and i test also : ffmpeg -i input.mp4 -c:v libx264 -preset fast -crf 22 -c:a copy,subtitles=srt.srt output.mp4 [14:15] <Tatou> someone can help me please ? :D [14:18] <Mavrik> you still didn't pastebin the output and the error you get. [14:18] <Tatou> ok [14:19] <Tatou> so for this : ffmpeg -i input.mp4 -c:v libx264 -preset fast -crf 22 -c:a copy subtitles=srt.srt output.mp4 [14:21] <Tatou> http://pastebin.shortrip.org/?58e20f9a93c9b23a#86xDc/p3IBH8dVXQyfr6YnMGJBvL… [14:21] <Mavrik> Tatou, did you not see this: "subtitles=File.srt: No such file or directory" :) [14:21] <Tatou> yes [14:22] <Hello71> > Javascript is required for ZeroBin to work. [14:22] <Tatou> but the file or directory is good :p [14:24] <Tatou> for this : ffmpeg -i input.mp4 -c:v libx264 -preset fast -crf 22 -c:a copy,subtitles=srt.srt output.mp4 [14:25] <Tatou> i have this error : http://pastebin.shortrip.org/?39731f27e9b86c11#+RLsVyIYtTT8tzAOmICX3vm3Veg7… [14:27] <Mavrik> yeah, that because "copy,subtitles" isn't a codec. [14:27] <Mavrik> :) [14:28] <Mavrik> ugh [14:28] <Mavrik> yeah, I see [14:28] <Mavrik> "subtitles=srt.srt" makes no sense [14:28] <Mavrik> why is that there? :) [14:28] <Tatou> my file [14:28] <Tatou> ^^ [14:28] <Mavrik> you're probably trying to add a video filter - which has to be specified with "vf" [14:28] <Mavrik> "-vf subtitles=..." [14:28] <Mavrik> not as codec., [14:28] <Tatou> yesss [14:29] <Mavrik> if you want to bake subtitles into video [14:29] <Mavrik> not sure if .srt is supported yet tho [14:29] <Tatou> i don't understand the connection [14:30] <Tatou> I just would burn srt in my video and put the fast preset :p [14:33] <Mavrik> ? [14:34] <Mavrik> you need to tell ffmpeg with "vf" swtich to use a subtitle video filter. [14:34] <Mavrik> subtitle video filter does the burning [14:34] <Mavrik> which part is giving you issues? :) [14:36] <Tatou> so [14:37] <Tatou> what should I do ? x) [14:37] <Tatou> i have to delete vf ? [14:38] <Tatou> ah no [14:39] <Tatou> 2 seconds x) [14:40] <Tatou> Ah it work [14:41] <Tatou> thank's for your help [14:41] <Tatou> i made this command : ffmpeg -i input.mp4 -c:v libx264 -preset fast -crf 22 -c:a copy -vf subtitles=File.srt output.mp4 [14:41] <Tatou> and i forgot vf :p [15:22] <ale345354> Hello [15:23] <ale345354> One question: I'm using ffmpeg version N-42642-g7db0f9b- from John Van Sinckle website. SHould I use an upper to date version? [15:26] <c_14> Probably, yes. [15:28] <ale345354> Where can I find detailed information on how to update my ffmpeg version via linux terminal? [15:30] <DHE> that version looks like it's dated May 2014 [15:33] <c_14> ale345354: just download a new static build and overwrite it? [15:34] <ale345354> yes it says "built on May 6 2014 02:25:05 with gcc 4.8 (Debian 4.8.2-21)" [15:35] <ale345354> The point is I don't understand how to download a new static build, from where and how to overwrite it, sorry [15:35] <c_14> http://johnvansickle.com/ffmpeg/ [15:35] <c_14> just wget/curl/whatever the latest static for your architecture. [15:35] <c_14> Then unpack it with tar xvJf file.tar.xz [15:36] <c_14> Then move the 'ffmpeg' binary to wherever your current 'ffmpeg' binary is. [15:37] <ale345354> I'm trying: would you recommend me downloading the with git or without git release of today? [15:39] <c_14> I'd recommend git. [15:43] <xata> what is fastest encoder? Faster than x264 with veryfast? speed of encoding and size matters, quality does not [15:44] <c_14> x264 with ultrafast? [15:44] <klaxa> is there a dummy encoder? guaranteed to be fast and small but no quality at all [15:44] <c_14> -c:v copy [15:44] <ubitux> getting a small output is hard [15:45] <ubitux> you'd better trash your video before feeding it to x264 with ultrafast [15:45] <ubitux> maybe x264 can be smart about that though [15:45] <ubitux> ultra fast with a very high crf maybe [15:46] <klaxa> what is your exact usecase anyway? [15:46] <ale345354> I remember I did something like you told me back in May. I found two folders from that exact date: one is named "ffmpeg_build" and has inside 3 folders (include, lib and share). The other one is named "ffmpeg-git-20140506-64bit-static" and inside has two folders (manpage, prsets) and a bunch of files. What should I do now? I unpacked what I downloaded. [15:46] <ubitux> the usecase seems to be a numeric diarrhea [15:46] <ubitux> fast and crappy [15:47] <c_14> ale345354: `cp ffmpeg-git-*/ffmpeg `which ffmpeg`' <- basically that [15:50] <ale345354> Sorry? SHould I simply copy the folder it just finished extracting on my desktop (ffmpeg-git-20141027-64bit-static) and overwrite the other old folder which has the similar name (ffmpeg-git-20140506-64bit-static) and I'm done? [15:50] <ale345354> I mean cut not copy [15:50] <c_14> What does `which ffmpeg' return? [15:50] <c_14> Or have you been calling ffmpeg with an absolute path? [15:51] <ale345354> I don't know. writing which ffmpeg in the terminal says: /home/ale/ffmpeg-git-20140506-64bit-static/ffmpeg [15:52] <c_14> You _could_ just overwrite the directory and it's contents if you want. Might be mildly confusing because the date on the dir would not match the ffmpeg build date, but it would work. [15:52] <c_14> Either that or change your PATH variable so it points to the new folder instead of the old one. [15:52] <ale345354> Maybe I?m invokng ffmpeg with an absolute path, I mean [15:53] <ale345354> If I want to convert a file in the folder for example /home/ale/Desktop/a/b/c/example, I just go in the /example folder, open there the terminal and just invoke "ffmpeg -i file ..." [15:55] <ale345354> How can I change my PATH? [15:55] <ale345354> so it points to the new folder? [15:56] <xata> klaxa: live streaming of video from an arm board. problem is that input video is 12mbps, and output should be max 1mbps [15:57] <xata> ubitux: tried ultrafast with 50 crf, still 2/3 of needed speed [15:58] <ubitux> decode your source with lowres [15:58] <ubitux> or just scale it down before sending to the encoder [15:58] <c_14> ale345354: if you're using bash, probably in .bashrc or /etc/profile or /etc/bashrc or .bash_profile [15:59] <xata> ubitux: how do i lowres? [15:59] <ubitux> xata: not a good idea [15:59] <ubitux> it's a decoding option, you don't want that [15:59] <ubitux> also, it's codec specific [15:59] <ubitux> just -vf scale=iw/2:ih/2 [16:00] <ubitux> should help your speed :p [16:00] <kepstin-laptop> xata: you could also try lowering the video framerate (dropping frames) [16:00] <ale345354> Sorry? I don't understand [16:01] <ale345354> What can I do? [16:01] <c_14> ale345354: open ~/.bashrc ~/.bash_profile /etc/profile /etc/bashrc in your favorite text editor and look for something along the lines of export PATH= [16:03] <ale345354> found! [16:03] <c_14> Now look for the part that says /home/ale/ffmpeg-git-20140506-64bit-static and change it to where the new dir is [16:03] <ale345354> there's written at the end "export PATH=/home/ale/ffmpeg-git-20140506-64bit-static:${PATH}" [16:04] <xata> ubitux: nah, scaling slows down the encoding speed [16:04] <ubitux> heh, slow scaling then, interesting [16:04] <c_14> Though you might just want to rename the dir to ffmpeg and put that in the PATH so you don't have to change it again next time [16:04] <ubitux> xata: --tune zerolatency? [16:05] <ale345354> Ok now the which ffmpeg command tells me /home/ale/ffmpeg-git-20141027-64bit-static/ffmpeg [16:05] <ale345354> What should I do to rename the dir to ffmpeg and put that in the PATH? [16:05] <c_14> mv ffmpeg-git-20141027-64bit-static ffmpeg [16:06] <ale345354> ok and then? [16:06] <c_14> Then just go back to the PATH variable and replace the string with "export PATH=/home/ale/ffmpeg/${PATH}" [16:06] <c_14> eh [16:06] <c_14> : instead of / after ffmpeg [16:07] <c_14> ie PATH=/home/ale/ffmpeg:${PATH} [16:09] <ale345354> So it searches by itself the binary inside the newer directory I will extract there? [16:09] <c_14> yes [16:10] <ale345354> oh :-0 [16:11] <ale345354> when I do "mv ffmpeg-git-20141027-64bit-static ffmpeg" shouldn't I do "mv ffmpeg-git-20141027-64bit-static ffmpeg_build"? [16:11] <ale345354> Because I have the folder ffmpeg_build in my /home, not the folder ffmpeg [16:12] <kepstin-laptop> ale345354: you're renaming the directory 'ffmpeg-git-20141027-64bit-static' to be just 'ffmpeg' so you can use the same directory name even if you change the ffmpeg version. [16:13] <ale345354> Ok, you used the move command just the rename the directory? [16:14] <c_14> yep [16:14] <ale345354> :-0 I'm speechless [16:14] <ale345354> Is it safe? [16:14] <DrSlony> Help, when I compress this 20s clip I have two issues: ffprobe shows 59.42 fps instead of 29.70, and it jerks in MPV (a good mplayer fork which had not jerked on any other video) but plays smoooth in chrome and vlc. Files: http://filebin.net/ffmpeg0fps Command: [16:14] <DrSlony> ./programs/code-ffmpeg/ffmpeg -i 20_copy.mp4 -ac 2 -c:a libfdk_aac -b:a 128k -c:v libx264 -preset slower -crf 23 -level 4.1 -maxrate 4096k -bufsize 2560k 20_copy_crf23_l41_mr4096_bs2560.mp4 [16:15] <c_14> ale345354: as long as you don't go overwriting things, sure [16:18] <c_14> DrSlony: doesn't jerk for me. mpv 0.6.1 [16:19] <DrSlony> 0.6.1 too [16:19] <DrSlony> what fps does the file have? [16:19] <c_14> 59.42 [16:20] <DrSlony> source had 29.70, why was it increased using my command? could you test on your end? [16:21] <ale345354> Ok guys, now all is working fine [16:22] <ale345354> I say a big THANK YOU to you all ffmpeg developers and [16:22] <ale345354> since you've been so kind to me, I'm rewarding you all with me singing for you a Cinderella song: [16:23] <ale345354> I took a walk down a road it's the road I was meant to staaaaaaay, I see the fire in your eyes but a man's gott make his waaaaay, So are you tough enough for my love ??? Just close your eyes to the heaven above!!!! I'm comin hoooome, I'm comin hooooome [16:23] <ale345354> Thank you again [16:24] <c_14> DrSlony: hmm, it increases the fps when I do it as well, strange [16:24] <DrSlony> so this is not normal, correct? [16:24] <c_14> DrSlony: is the source variable framerate perhaps? [16:24] <DrSlony> i checked the docs as much as sanity would allow and didnt find any caveats [16:24] <DrSlony> how do i check? [16:25] <DrSlony> its from a sony xperia z [16:25] <DrSlony> i did a direct stream copy of the first 20 seconds, so i guesss it should have the same fps as the full clip [16:25] <DrSlony> its on the filebin link i pasted [16:26] <c_14> DrSlony: use -vsync vfr [16:26] <c_14> When I add that to the command the output has 29.73 fps [16:26] <DrSlony> ill try [16:29] <DrSlony> didnt work. where do i put that part exactly? [16:29] <DrSlony> oh i typo'd [16:30] <DrSlony> retrying [16:31] <DrSlony> yup, confirm that it worked and the mpv playback is smooth now :) thanks! [16:32] <c_14> np [16:59] <zybi1> hi [17:00] <zybi1> what would be the command line to encode mkv (lossless) to QuickTime ProRes 422 please? [17:03] <DrSlony> oh dear [17:03] <zybi1> it's for a film festival [17:04] <zybi1> or is it easier to make a working DCP which is fully DCI/SMPTE compliant? [17:05] <c_14> zybi1: I'm pretty sure prores is a lossy codec... [17:06] <zybi1> c_14: thanks for the hint. I know that. But I'm searching for a solution which is possible for me to reach. Deadline is october 31st [17:06] <zybi1> I don't have money to give it to somebody to create a working DCP for me.. [17:07] <zybi1> and since they don't accept lossless MKV - [17:07] <c_14> If you just want to encode to prores, ffmpeg -i input -c:v prores -b:v [bitrate] outfile.prores [17:07] <c_14> You can probably keep it in a matroska container. [17:07] <c_14> I think prores also supports qscale [17:07] <c_14> But don't quote me on that. [17:07] <c_14> Never used it before. [17:09] <c_14> If the video has audio and you don't want to touch it, add -c:a copy [17:10] <zybi1> flac is no problem for osx/windows ? [17:10] <DrSlony> zybi1 http://opendcp.org/ [17:11] <zybi1> DrSlony: you have experience with that? also with festivals? [17:11] <DrSlony> negatory [17:11] <zybi1> how long will it take about? [17:11] <zybi1> alright [17:11] <DrSlony> only DCPs im experienced with are something else in photography [17:12] <zybi1> ya, it seems to be a strong tool [17:12] <DrSlony> theres also http://dcpomatic.com/ [17:20] <zybi1> thanks to you two [23:07] <voip_> hello guys [23:10] <voip_> I am taking stream and sending to DestinationA: [23:10] <voip_> ffmpeg -i http://SourceIP -preset fast -acodec aac -strict experimental -ar 44100 -ac 2 -b:a 96k -vcodec libx264 -b:v 900k -f flv rtmp://DestinationA [23:10] <voip_> How to send same stream at the same time also to DestinationB ? [23:22] <c_14> https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs#Teepseudo-muxer [23:30] <voip_> c_14, in samples multiple encoding, and multiple inputs. I need for 1 input, same encoding but multiple out . Can you please help ? [23:31] <c_14> I just gave you the link? just use the tee pseudo-muxer [23:32] <c_14> ie -f tee "[f=flv]rtmp://DestinationA|destinationB" [23:45] <voip_> c_14, thank you very much ! [00:00] --- Tue Oct 28 2014

1 0

ffmpeg-devel.log.20141026
by burek 27 Oct '14

27 Oct '14

[01:23] <iive> there seems to be unusually high level of trolling this month. [02:49] <Compn> iive : getting ready for holiday trolling level [02:00] <cone-669> ffmpeg.git 03Martin Storsjö 07master:851ace79a307: wtv: Avoid needlessly calling gmtime twice with the same argument [02:00] <cone-669> ffmpeg.git 03Michael Niedermayer 07master:50697ac5b2e7: Merge commit '851ace79a307bea54b44bd6f7ecd3b7861c28ec6' [02:25] <cone-669> ffmpeg.git 03Martin Storsjö 07master:9dcf2397219c: lavf: Check the return value of strftime [02:25] <cone-669> ffmpeg.git 03Michael Niedermayer 07master:3b709fd91274: Merge commit '9dcf2397219ca796f0fafce2a703770d6fd09920' [02:32] <cone-669> ffmpeg.git 03Martin Storsjö 07master:3f8f1c6ff24e: lavu: Provide fallbacks for gmtime_r and localtime_r [02:32] <cone-669> ffmpeg.git 03Michael Niedermayer 07master:edb069e55689: Merge commit '3f8f1c6ff24ee858eb5b0bf47ef6d4605299a87e' [03:08] <cone-669> ffmpeg.git 03Martin Storsjö 07master:82ee7d0dda0f: Use gmtime_r instead of gmtime and localtime_r instead of localtime [03:09] <cone-669> ffmpeg.git 03Michael Niedermayer 07master:4a39d4c65a72: Merge commit '82ee7d0dda0fec8cdb670f4e844bf5c2927ad9de' [03:22] <cone-669> ffmpeg.git 03Martin Storsjö 07master:6b9b42cc5576: drawtext: Remove the ifdef for localtime_r [03:22] <cone-669> ffmpeg.git 03Michael Niedermayer 07master:4360c6ff610c: Merge commit '6b9b42cc5576e1819ad1e29d98940066fd14b2d6' [03:29] <cone-669> ffmpeg.git 03Martin Storsjö 07master:ed6dad3737bf: lavf: Implement ff_brktimegm using gmtime_r [03:29] <cone-669> ffmpeg.git 03Michael Niedermayer 07master:abaa41b6426c: Merge commit 'ed6dad3737bf7bb2d5e9fa9511dfdb44806010e8' [12:27] <kurosu> since the addition of tests/ref/fate/sub-*, changes are applied locally automatically (line endings?) [12:27] <kurosu> do I need to modify my git config in some way? [12:27] <kurosu> I have autocrlf = false [12:31] <kurosu> (btw under windows) [12:40] <kurosu> huh, alternated between different configs, switched back to the original, and the issue vanished [16:03] <cehoyos> pross: Hi, how did you test cinedec? [16:04] <cehoyos> I compared the tiff file in http://samples.mplayerhq.hu/V-codecs/Phantom_Cine/phantom_cine/ with what FFmpeg produces and it looks quite different. [16:05] <cehoyos> Do I just miss something important? [16:30] <akira4> umm...ubitux I was wondering about what to do next. Do you have anything in mind? [16:34] <ubitux> akira4: well, so far STL support is OK, we can look at supporting the markup, but there is probably more interesting things to do subtitles wise [16:34] <ubitux> you may want to have a look to https://trac.ffmpeg.org/query?status=!closed&keywords=~sub [16:35] <ubitux> you could start considering the vobsub muxer, but this might be quite troublesome for now so better postpone it for when you're more familiar with the code [16:35] <akira4> oh. Okay. [16:36] <ubitux> so this is https://trac.ffmpeg.org/ticket/2391 and https://trac.ffmpeg.org/ticket/2035 is kind of related [16:36] <ubitux> but it's a bit sensible [16:37] <ubitux> let me see what you could work on... [16:38] <akira4> cool [16:42] <ubitux> most of the current issue are related to the subtitles in various containers [16:43] <ubitux> like, in mpeg-ts, mov, or mkv [16:43] <akira4> I see... [16:43] <ubitux> and also bitmap related [16:43] <ubitux> you'll probably have to get familiar with these containers, and how bitmap subtitles work to deal with these issues [16:44] <ubitux> so far, for the big pictures, we have various big problems [16:44] <akira4> hmm. [16:44] <ubitux> for the big picture* sorry [16:44] <ubitux> like, for text subtitles, the decoded text is broken, but i'm currently working on this, so there isn't much you can do about it [16:44] <ubitux> and also, it's very API change related [16:45] <ubitux> the idea is to move the decoded subtitles structure from libavcodec to libavutil [16:45] <ubitux> that's what we did for audio and video (AVFrame) which is in libavutil [16:45] <ubitux> the main motivation for this is to be able to access them in libavfilter, without a dependency to libavcodec [16:46] <akira4> so then what exactly does libavcodec have? [16:46] <ubitux> so there are some boring api design involved here [16:46] <ubitux> libavcodec only has decoders and encoders [16:46] <akira4> okay. [16:46] <ubitux> libavfilter filters frames, but it doesn't need to know about codecs [16:47] <ubitux> like, you could have an application working on RGB data and willing to process the frames [16:47] <ubitux> without involving the codecs [16:47] <ubitux> that's the idea of not having a dependency on libavcodec in libavfilter [16:47] <ubitux> libavutil is basically the shared dependency of all the other libs [16:48] <ubitux> so that's where we put the common stuff, here AVFrame, and soon the subtitles [16:48] <akira4> big change huh [16:48] <ubitux> yeah, kind of [16:48] <ubitux> so anyway, for the text subtitles, i'm working on moving the structure to libavutil, and i'm not exactly sure what to do for the bitmap ones [16:48] <ubitux> (currently the structure is shared between bitmaps and text) [16:49] <ubitux> (AVSubtitle) [16:49] <akira4> oh. [16:49] <ubitux> anyway, so this is one of the issue [16:50] <ubitux> another big one is related to closed captions [16:50] <ubitux> they don't fit really well in the text and bitmap model, because they come along the video frames (afaiu) [16:50] <ubitux> kierank might be able to talk about it more [16:50] <ubitux> currently we seem to have a libzvbi wrapper to deal with that [16:51] <ubitux> i'm not exactly sure how well it works, but it wasn't exactly pretty last time i looked [16:51] <kierank> there are two types of closed captions [16:51] <kierank> one where it's a normal data stream [16:51] <kierank> and one where it sits in the video frame [16:51] <akira4> hmm. okay [16:52] <ubitux> kierank: do you think it would make sense to make akira4 work on the CC? [16:52] <ubitux> afaik that's a long standing issue, and you seemed interested in such support [16:52] <kierank> it's hard/impossible to integrate [16:52] <ubitux> haha [16:52] <akira4> ouch [16:53] <ubitux> akira4: alright, so here is another big issue, let's move to another one [16:53] <akira4> sure. [16:53] <ubitux> next, i could talk about the bitmap subtitles [16:54] <ubitux> this includes typically DVD, DVB or PGS (bluray ones) subtitles [16:54] <akira4> yep. They seem interesting [16:54] <kierank> ubitux: ebu-tt might be interesting? [16:54] <kierank> instead of cc [16:55] <ubitux> kierank: good question... are you interesting in this? i mentioned it in the wiki but i have no idea of the use cases [16:55] <kierank> ubitux: well it's similar to webvtt [16:57] <ubitux> i guess we can have a look to this, but it looks like another big thing with no demand so far [16:57] <ubitux> unless i missed it [16:57] <ubitux> well, i'm gonna finish my little speech about bitmap before we move to this [16:58] <akira4> alright. [16:58] <ubitux> akira4: so yeah, currently our DVD stack is not exactly good [16:58] <akira4> DVD stack? [16:59] <ubitux> yeah, like, if you want to take your dvd and encode it with ffmpeg well... it's not exactly the best way to do it [16:59] <ubitux> which is a shame given that the project is called FFmpeg and DVD are in mpeg :p [16:59] <akira4> haha. I see. [16:59] <ubitux> but basically dvd are into mpeg-ps containers (the .VOB files), and have also various files such as .IFO and stuff like that [17:00] <ubitux> which contain information about the subtitles [17:00] <ubitux> but my knowledge of this is fairly limited [17:00] <JEEB> the DVD specs were never pushed out into the hands of those who are not supposed to know [17:00] <JEEB> so yeah :/ [17:01] <JEEB> videolan seems to be maintaining the DVD reading and format parsing libraries these days [17:01] <ubitux> and as result, some dvd libs showed up years ago [17:01] <ubitux> libdvdread/libdvdcss [17:01] <ubitux> which are used by players to get a "clean" mpeg stream and stuff like that [17:01] <ubitux> indeed, now videolan maintains these [17:02] <akira4> hmm. [17:02] <ubitux> you can find them here: http://git.videolan.org/ [17:02] <ubitux> look at libdvd{nav,css,read} [17:02] <ubitux> that's what players use when they want to play dvds [17:02] <ubitux> and ideally, ffmpeg should be using those [17:03] <ubitux> so anyway, so far you can actually get video and audio somehow with ffmpeg even without those libs but the subtitles support is as you might guess, pretty bad [17:03] <akira4> I see. [17:03] <ubitux> some external tools are able to extract the subtitles from the dvd into a .sub and .idx couple [17:04] <ubitux> the .sub contains the bitmaps data, and the .idx is a text file informing the offset of the bitmaps, with timestamps, language, and various other metadata such as colors [17:04] <ubitux> in ffmpeg we have support for .sub/.idx for reading [17:04] <ubitux> ideally, we should be able to create such file [17:05] <ubitux> but that's a sensible issue [17:05] <ubitux> (because dvd subtitles are kind of tricky) [17:06] <akira4> okay. [17:06] <ubitux> so anyway, the dvd subtitles themselves can be stored in various containers [17:06] <ubitux> you can find them in the .VOB of the DVDs (with the .IFO giving additionnal information) [17:06] <ubitux> you can find them in .SUB files (with the .IDX giving timing and additional information) [17:06] <ubitux> (those being extracted from the VOB and IFO by random tools) [17:07] <ubitux> you can also find them muxed into .mkv [17:07] <ubitux> ffmpeg supports them in the 2 later (sub/idx and mkv), and maybe other formats [17:07] <ubitux> but directly from dvd, so far, we can't [17:07] <ubitux> SO, back to the subject, if we want such support [17:08] <ubitux> we probably want to add support for libdvdread typically [17:08] <ubitux> so we can read streams from the dvd, notably the subtitles [17:08] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:a9564e859bfc: avformat/mlvdec: Check the return code from strftime() [17:08] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:f0390638e0e1: avformat/matroskadec: Check the return code from strftime() [17:08] <cone-646> ffmpeg.git 03Michael Niedermayer 07master:c0002ddb019d: ffmpeg_opt: Check the return code from strftime() [17:08] <ubitux> and then be able for instance to create a .mkv with the subtitles out of the original dvd [17:08] <ubitux> which is very interesting for a user who wants to rip his own dvds [17:08] <ubitux> (which are currently dying) [17:09] <akira4> hmm. [17:09] <ubitux> so, this isn't a subtitles only project, but it could be interesting [17:09] <ubitux> and actually, an old patch already exists [17:09] <ubitux> which you could look into [17:09] <ubitux> are you interested in this, or we look at the ebu-tt thing? [17:09] <akira4> I'm interested in this one [17:10] <ubitux> the dvd? [17:10] <akira4> yep [17:10] <ubitux> alright, cool, i'm very also interested in seeing such support in ffmpeg [17:10] <ubitux> gonna dig Stefano's DVD patch then [17:10] <ubitux> give me a moment [17:10] <akira4> sure. [17:11] <ubitux> https://trac.ffmpeg.org/ticket/3280 a related ticket [17:12] <ubitux> this iso hasn't any subtitles though [17:12] <ubitux> akira4: ah yeah, there is also the dvdnav support which is basically refering to the menu and shit when you read a dvd [17:12] <ubitux> but we don't want to care about this for now [17:13] <akira4> okay. [17:13] <ubitux> libdvdread is to be able to read the dvd, libdvdcss is about the security shit, and libdvdnav is for the click menu stuff [17:13] <ubitux> you'll be only interested in libdvdread basically [17:13] <akira4> cool. [17:14] <ubitux> where is that patch grumbl. [17:15] <cehoyos> The link is in the ticket iiurc [17:15] <cehoyos> iiuc [17:15] <ubitux> ah! http://lists.ffmpeg.org/pipermail/ffmpeg-devel/2012-January/119842.html [17:15] <ubitux> here it is [17:16] <ubitux> i think it will be wise to create a "dvd" format instead of protocol, which has too many limitation [17:16] <ubitux> otherwise, you'll be unable to export chapters and stuff [17:16] <ubitux> so... better pick the format model [17:16] <wm4> please don't add crappy unusable shit like the current bluray support [17:16] <wm4> make it useful for everyone [17:16] <wm4> but I can see it coming... [17:17] <akira4> I'll try. [17:17] <ubitux> wm4: yeah bluray is implemented through a protocol so it suck a bit [17:18] <wm4> "a bit" [17:18] <ubitux> thought, it was useful to have such support last time to dump a part of the video [17:18] <wm4> it's utterly unusable, except for creating bad transcodes [17:18] <ubitux> well, it was useful to dump the raw video at least [17:19] <ubitux> akira4: so anyway yeah, not useful for subtitles immediately, but if we can get the dvd subtitles from such support that will be great [17:19] <ubitux> akira4: because currently, you can still feed ffmpeg the vob and it will do just fine for the a/v [17:19] <wm4> my suggestion is implementing dvd support as demuxer [17:19] <ubitux> wm4: that's what i suggested as well [17:19] <ubitux> akira4: but for the subtitles... not so much :) [17:20] <ubitux> hopefully libdvdread is smart about the subtitles [17:20] <ubitux> otherwise... you might need to parse IFO files and stuff [17:20] <ubitux> you'll have to check this stuff by yourself [17:20] <ubitux> that's all i know about this [17:20] <akira4> hmm. okay. [17:21] <wm4> ubitux: libdvdread is a low level piece of shit that requires you to deal with the parsed dvd structures yourself [17:21] <ubitux> don't frighten akira4 :( [17:21] <ubitux> akira4: so yeah, your mission if you accept it would be to add dvd support as a demuxer (just like libavformat/stldec.c) using these libs [17:21] <ubitux> but instead of outputing just a single subtitles streams, it will output various audio, video and subtitles from it [17:21] <wm4> ubitux: frighten? this stuff _is_ bad and complicated [17:22] <ubitux> akira4: in other words, you want to be able to do ffmpeg -f dvd -i mydvd.iso -c copy out.mkv [17:22] <ubitux> (mkv can contain all the mpeg streams with no transcode right?) [17:23] <akira4> yeah. [17:23] <wm4> but please WITHOUT relying on the timestamp rewriting code in ffmpeg.c [17:23] <ubitux> how is that related? [17:24] <ubitux> do we have examples of dvd reading somewhere that could be used as a base? [17:24] <ubitux> wm4: mpv still has it? [17:25] <JEEB> VLC's is probably the least bad [17:28] <wm4> ubitux: has what? [17:28] <wm4> JEEB: I think vlc makes some simplifications [17:28] <ubitux> libdvd* support to read from dvd:// [17:28] <wm4> ubitux: yes, both [17:29] <ubitux> mmh GPL though [17:31] <wm4> ubitux: just like libdvdread? [17:31] <ubitux> yeah, but i mean i can't tell akira to look at this when adding the demuxer which will probably be lgpl [17:32] <wm4> does that make any sense? [17:33] <wm4> or are you expecting a gpl-free libdvdread replacement [17:33] <ubitux> vlc was relicensed to lgpl, this could happen libdvd* or a later fork/rewrite [17:33] <ubitux> not having to worry about the demuxer license will simplify this [17:49] <cone-646> ffmpeg.git 03Matthew Oliver 07master:2060f4cbba62: avutil/intmath: enable builtin intrinsics for icl and msvc. [17:52] <akira4> so then what do I do? [18:01] <ubitux> akira4: you can look at the patch from Stefano [18:01] <ubitux> it is implemented as a protocol so you'll have to implement it as a demuxer instead [18:02] <ubitux> then maybe look a bit at vlc code to see how to use it [18:02] <akira4> okay. I'll do that. [18:05] <cehoyos> Should it be a device? Or is that just nonsense? [18:06] <ubitux> not sure it makes much difference [18:06] <ubitux> just a different directory right? [18:07] <cehoyos> I may really misunderstand the basics but doesn't mean implementing it as a demuxer that an ugly hack will be needed for the used protocol? [18:07] <cehoyos> I thought devices are a little like protocol+demuxer [18:09] <ubitux> afaik devices are just (de)muxers in a different directory [18:09] <cehoyos> So which different protocols can be used for video4linux? [18:10] <cehoyos> Or said differently: Which protocol should be used with the libdvdread demuxer? [18:10] <ubitux> none? [18:10] <ubitux> -f dvd -i ... [18:11] <ubitux> just like lavfi (which is a device, right) [18:11] <cehoyos> But I believe our current demuxers either work with all or at least several protocols, or am I wrong? [18:11] <ubitux> but it's just a AVFMT_NOFILE flag or something like that [18:11] <ubitux> cehoyos: not sure what you mean here [18:12] <ubitux> like, -f v4l2 -i http:///dev/video0 ? [18:12] <ubitux> :p [18:12] <ubitux> can we really do that? [18:12] <ubitux> does it make any sense? [18:12] <cehoyos> No, that's why I believe that a device != a demuxer. [18:12] <ubitux> i wonder if there is really any technical differences [18:13] <cehoyos> And while I still may miss something (or everything) the more I think about it the more obvious it seems to me that dvdread must be a device. [18:14] <ubitux> well, if you prefer, maybe [18:14] <ubitux> it just sounds like a different directory to me :P [18:15] <cehoyos> Then please answer again: Which protocols should the libdvdread demuxer support? [18:15] <cehoyos> I would have expected that it cannot even work with file:// [18:16] <ubitux> well, you could have -f v4l2 -i http://mydvd.iso maybe? but dunno [18:16] <ubitux> http://foobar.com/mydvd.iso [18:16] <ubitux> not sure the avio layer is gonna like that much though [18:16] <wm4> libdvdread doesn't have support for this [18:16] <ubitux> but... let's say smb:// then [18:16] <wm4> although I believe j-b was interested in it [18:16] <cehoyos> So iiuc, you mean there is no problem using the file protocol with libdvdread? In this case, please ignore my comments, I simply misunderstood [18:17] <ubitux> i don't know libdvdread [18:17] <ubitux> not sure if you can define your own avio layer [18:17] <wm4> no, libdvdread AFAIK normally needs a dvd folder [18:17] <cehoyos> I thought it will not / cannot work with our current file protocol [18:17] <ubitux> wm4: so libdvdread handles the whole io layer? [18:17] <wm4> yes [18:17] <cehoyos> How else can it work? [18:18] <ubitux> well you could have io callbacks [18:18] <ubitux> for read & seek [18:18] <ubitux> which could be plugged on avio [18:19] <ubitux> it would make sense to me to be able to -f dvd -i smb://192.168.12.34/mydvd.iso [18:19] <ubitux> at some point [18:19] <cehoyos> Of course! [18:19] <wm4> so naive [18:19] <ubitux> haha sure, but in theory [18:19] <cehoyos> But do you believe that our current file protocol will support libdvdread demuxer? [18:20] <ubitux> in theory i don't think it would be an issue [18:20] <wm4> AFAIK libdvdread has no iso code (I might be wrong), and wants directory accesses [18:20] <ubitux> huh? i remember being able to read dvds from iso with mplayer [18:20] <wm4> same for libbluray [18:20] <cehoyos> Didn't you mount them? [18:20] <wm4> are you sure mplayer didn't try to read the iso as mpeg data [18:20] <ubitux> mmh not sure [18:20] <ubitux> good question [18:21] <nevcairiel> libbluray at least has a pluggable IO layer, you could give it IO functions that somehow read directly out of an ISO, but you would need to add that yourself [18:21] <ubitux> yeah sure, you would do the mapping to avio, shouldn't be impossible [18:22] <ubitux> but anyway, yeah maybe i was wrong about the iso [18:22] <ubitux> -dvd-device <path to device> (DVD only) [18:22] <ubitux> Specify the DVD device or .iso filename (default: /dev/dvd). You can also specify a [18:22] <ubitux> directory that contains files previously copied directly from a DVD (with e.g. vob [18:22] <ubitux> copy). [18:22] <ubitux> this is from mplayer manpage though [18:23] <wm4> then maybe it can do iso, but has no IO callbacks [18:23] <ubitux> yeah, right, for now :) [18:23] <ubitux> but it doesn't sound impossible [18:25] <ubitux> there is already a callback system in place [18:25] <ubitux> because of the dvdcss support [18:25] <ubitux> it just needs to be exposed somehow [19:18] <cone-646> ffmpeg.git 03Christophe Gisquet 07master:6fe02d25ee68: dv: fix weight table for 2x4x8 transform [19:39] <cone-646> ffmpeg.git 03Christophe Gisquet 07master:05da586fef86: dv: move inverse weight tables to decoder [20:00] <ubitux> arwa: please learn to bottom post :) [20:01] <arwa> umm....okay :D [20:13] <ubitux> arwa: use valgrind by the way [20:13] <ubitux> to debug the segfault [20:13] <ubitux> it might not crash on your system because you're lucky [20:14] <ubitux> it will show you where you're doing it wrong in memory [20:27] <arwa> okay [20:27] <arwa> I actually understood what is going wrong [20:28] <arwa> for boundry pixels, it is accessing the pixels which are not the part of the image. [20:29] <arwa> For that case should I zero pad the image? [20:31] <arwa> But when I was going through the writing_filters doc, they mentioned not to change the linesize. [20:37] <ubitux> xbr doesn't define how boundaries should be handled? [20:38] <ubitux> hqx just considers there is no difference when it's out of the boundaries [20:38] <ubitux> iirc [20:39] <ubitux> let me check actually [20:39] <ubitux> yeah well right, it compares against itself so.. [21:22] <arwa> " cp: cannot create regular file `ffmpeg': Text file busy " what does this mean? [21:23] <arwa> I googled it. It is suggesting that multiple processes are running, so i need to kill the process [21:23] <arwa> but after killing the process, its still giving me the same error [21:27] <ubitux> check with lsof [21:27] <ubitux> you just have a ffmpeg running somewhere [22:39] <cone-646> ffmpeg.git 03Kieran Kunhya 07master:74141f693ded: avcodec: Add support for Opus in MPEG-TS [22:39] <cone-646> ffmpeg.git 03Kieran Kunhya 07master:61e42c1124e4: avformat/mpegts: Add support for Opus in MPEG-TS [23:14] <kierank> woah [23:14] <nevcairiel> wee! [23:15] <nevcairiel> can we get muxing now too? [23:16] <kierank> probably [23:19] <kierank> will need to see if ffmpeg doesn't misdetect opus in ts as mp3 first [23:35] <cehoyos> kierank: As said, the only bug I know of is a transport stream that claims ac-3 but contains mp2 and plays fine with WMP / fails with FFmpeg (ticket 1798) [00:00] --- Mon Oct 27 2014

1 0

ffmpeg.log.20141026
by burek 27 Oct '14

27 Oct '14

[00:12] <deivor__> Hi [00:13] <deivor__> Does anyone know which AAC library uses the least CPU (live encoding) ? I'm using a raspberry pi and I hit about 30% CPU when encoding from ALSA -> AAC (mono, 64kbps). I'm currently using the native AAC library, but could compile fdk_aac or libfaac [00:16] <BtbN> try it? [00:16] <BtbN> fdk with afterburner disabled might be the best, but seriously, just try it. [00:18] <Lac3rat3d> i'm trying to burn in subtitle stream to video stream, but it's not working. anyone have experience/ideas? [00:18] <Lac3rat3d> ok hang on [00:19] <deivor__> @BtbN Ok, I was asking because I'm fiddling with the cross-compiling options and it takes some time to get it right [00:21] <Lac3rat3d> c_14: http://www.pastebin.com/Bd4H9Vr9 [00:24] <Lac3rat3d> any ideas? [00:26] <Lac3rat3d> Assertion fs->on_event failed at /home/kyle/software/ffmpeg/source/ffmpeg-git/li [00:26] <Lac3rat3d> bavfilter/framesync.c:276 [00:26] <Lac3rat3d> that error seems like there is a hardcoded path [00:26] <c_14> Lac3rat3d: https://trac.ffmpeg.org/wiki/HowToBurnSubtitlesIntoVideoy <- that's what you're looking for [00:27] <Lac3rat3d> ya that's the tutorial i used [00:27] <c_14> that location is compile-time, ie where the loc was when you compiled [00:27] <Lac3rat3d> i didn't compile it [00:27] <Lac3rat3d> it was precompiled [00:27] <c_14> subrip are text iirc [00:28] <c_14> Therefore, don't use the overlay filter, use the subtitle filter. [00:28] <Lac3rat3d> i got it from here: http://ffmpeg.zeranoe.com/builds/ [00:28] <deivor__> BtbN you meant that I should disable afterburner at runtime or compile time? [00:28] <Lac3rat3d> can you give me an example of what you mean? [00:29] <c_14> ffmpeg -i mkv -vf subtitles=mkv out.mkv [00:29] <Lac3rat3d> that will burn the subtitles from stream 0:3 into stream 0:0? [00:29] <c_14> That will burn the first subtitle track it finds in the file 'mkv' into the first video stream it finds [00:30] <c_14> Which should be what you want judging by the ffmpeg output. [00:30] <Lac3rat3d> sweet [00:30] <Lac3rat3d> i'll try it out [00:30] <Lac3rat3d> thanks, brb [00:30] <Lac3rat3d> yes, that's exactly what i nt [00:30] <Lac3rat3d> *want [00:31] <Lac3rat3d> didn't work :/ [00:31] <c_14> what's it say? [00:31] <Lac3rat3d> [Parsed_subtitles_0 @ 0000000002c4ec00] Unable to open mkv [00:31] <Lac3rat3d> [AVFilterGraph @ 0000000002af7cc0] Error initializing filter 'subtitles' with ar [00:31] <Lac3rat3d> gs 'mkv' [00:31] <Lac3rat3d> Error opening filters! [00:32] <c_14> that 'mkv' is supposed to be the name of the input file [00:32] <Lac3rat3d> ohh, my bad [00:32] <c_14> The current ffmpeg api for burning subtitles is -sadly- a tad awkward. [00:33] <Lac3rat3d> hmm it's doing something... [00:33] <Lac3rat3d> Fontconfig error: Cannot load default config file [00:33] <Lac3rat3d> [Parsed_subtitles_0 @ 00000000054cd3a0] No usable fontconfig configuration file [00:33] <Lac3rat3d> found, using fallback. [00:33] <Lac3rat3d> Fontconfig error: Cannot load default config file [00:33] <c_14> You on windows? [00:33] <Lac3rat3d> yea [00:33] <deivor__> try limiting the output movie length to test faster (-t 2:00 for 2 minutes) [00:33] <c_14> ugh, where was that htread again [00:34] <Lac3rat3d> seems like it's frozen now [00:34] <c_14> http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=10&t=318&start=20 <- this one i think... [00:35] <Lac3rat3d> what am i looking for? [00:36] <deivor__> Is there any way to disable *everything* when building ffmpeg but certain things? I only need x264, flv (wrapper), AAC and the internal RTMP library. Or do I need to add --disable-{each thing} ? [00:41] <c_14> Lac3rat3d: I think it's the response from camelotmsl which describes creating a fonts.conf on windows [00:41] <c_14> deivor__: --disable-everything or --disable-all [00:42] <Lac3rat3d> ok, i'll try it out. thanks c_14, i'll post back later [00:43] <deivor__> Thanks c_14 [00:44] <deivor__> BtbN, fdk_aac with -afterburner 0 uses 13% of the CPU instead of 24% (vs ffmpeg's internal aac), it's a huge improvement [00:44] <BtbN> how much does it use with afterburner enabled? [00:44] <deivor__> 24%, just like aac [00:46] <deivor__> I'm currently using the PI for live rtmp streaming (720p@30fps with the HW camera) and displaying a slideshow (hw accelerated).. in some cases (fast-forward through slideshows) the cpu peaks and lags the stream. I'm trying to minimize the peaking problem [00:48] <deivor__> ATM with the rtmp streaming and aac audio from ALSA (bt microphone) it's using 22.6% CPU which is fantastic [00:50] <Lac3rat3d> c_14: i made those env variables, and the file. it's not throwing the error anymore [00:50] <Lac3rat3d> but i don't know if it's actually doing something lol [00:51] <c_14> Is it burning cpu? [00:51] <Lac3rat3d> barely [00:51] <c_14> Is the framecount updating? [00:52] <c_14> At the bottom of the ffmpeg output? [00:52] <Lac3rat3d> nope [00:52] <Lac3rat3d> just frozer this: [00:52] <Lac3rat3d> Stream #0:1(eng): Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 1536 kb/s [00:52] <Lac3rat3d> Stream #0:2(ind): Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 1536 kb/s (de [00:52] <Lac3rat3d> fault) (forced) [00:52] <Lac3rat3d> Stream #0:3(eng): Subtitle: subrip (default) (forced) [00:52] <Lac3rat3d> Metadata: [00:52] <Lac3rat3d> title : English-Forced [00:52] <c_14> Press h 2-3 times [00:52] <Lac3rat3d> nothing [00:54] <c_14> hmm, not sure that works under Windows but it was worth a shot [00:54] <c_14> Can you pastebin your command line and all current output? [00:54] <Lac3rat3d> yep [00:54] <Lac3rat3d> hang on [00:55] <Lac3rat3d> http://pastebin.com/1ACCDAVP [00:55] <c_14> eeeeeep [00:56] <c_14> Don't tell me the input and output filename is the same? [00:56] <c_14> ok, it isn't [00:56] <c_14> nvmd [00:56] <c_14> just very similar [00:56] <Lac3rat3d> ya, output has 2 at the end [00:57] <c_14> Hmm, can you cancel that command and try again with -loglevel debug ? [00:57] <c_14> then pastebin that? [00:57] <Lac3rat3d> sure [00:57] <Lac3rat3d> oh wait! [00:57] <Lac3rat3d> it's doing something now [00:57] <Lac3rat3d> !! [00:57] <c_14> ok [00:57] <c_14> slow io? [00:57] <Lac3rat3d> i just left it at that piont where it seemed frozen this whole time [00:57] <Lac3rat3d> meh, ide hdd [00:57] <Lac3rat3d> shouldn't be that slow [00:57] <Lac3rat3d> but still [00:57] <Lac3rat3d> it's doing something now [00:57] <Lac3rat3d> lots of output [00:57] <c_14> Lots of hex output? [00:57] <Lac3rat3d> yep [00:58] <c_14> You can probably press h until it goes away [00:58] <Lac3rat3d> green, yellow, purple [00:58] <c_14> That's what I was trying to accomplish earlier. [00:58] <Lac3rat3d> ohh [00:58] <Lac3rat3d> ya [00:58] <Lac3rat3d> output changes with each 'h' [00:59] <c_14> yep [00:59] <Lac3rat3d> prob take a while since it has to burn it into the video stream? [00:59] <c_14> It has to reencode the entire video, yes. [00:59] <Lac3rat3d> cool [00:59] <Lac3rat3d> i'll just let it do it's thing then [00:59] <Lac3rat3d> thanks for your help man :) [00:59] <c_14> Not sure why anyone would ever need a hexdump of the current encoding output on STDERR, but it is useful for checking if ffmpeg is doing something. [01:00] <Lac3rat3d> yep, now i know :) [01:01] <deivor__> lacerated, try with the -t flag so you can check if the output is correct before encoding the file completely [01:01] <Lac3rat3d> too late :/ [01:01] <Lac3rat3d> but, what does that do? [01:01] <Lac3rat3d> -t [01:01] <c_14> If you're outputting to mkv, you can play the output while it's encoding. [01:02] <Lac3rat3d> ohhh [01:02] <deivor__> oh [01:02] <c_14> -t limits the duration of the output [01:02] <deivor__> if you use -t 5:00 it encodes 5 minutes and then stops [01:02] <Lac3rat3d> ohhhh [01:02] <Lac3rat3d> cool [01:02] <Lac3rat3d> i se [01:02] <Lac3rat3d> like a sample [01:02] <deivor__> yes [02:22] <bencc> is there -tt option similar to -ss option? [02:22] <bencc> or do I have to calculate how much time and use -t? [02:22] <c_14> there's -t and -to [02:24] <bencc> c_14: so -to is similar to -ss? [02:24] <c_14> yep [02:24] <c_14> But only if you don't use -ss or use -ss as an output option. [02:24] <c_14> If you use -ss as an input options, the timestamps are reset and -to works just like -t [02:24] <bencc> why can't I use -ss? [02:25] <bencc> so I can't say: start at 10 seconds and stop at 20 seconds? [02:25] <c_14> https://trac.ffmpeg.org/wiki/Seeking%20with%20FFmpeg#Cuttingsmallsections [02:25] <c_14> Should all be explained there. [02:34] <bencc> c_14: thanks [02:34] <bencc> not very convenient because I need to calculate by hand [02:00] <norbert> hi, is there an easy way to speed up a video (4x) and do it in a way that audio also speeds up? [02:00] <norbert> and that fps is still proper [02:01] <c_14> https://trac.ffmpeg.org/wiki/How%20to%20speed%20up%20/%20slow%20down%20a%20… [02:02] <norbert> c_14: thanks, I did find that page, but didn't understand the -16 and 0.125*PTS [02:02] <norbert> I have 60fps [02:03] <norbert> I guess -r is probably 60? [02:03] <norbert> but what should I do with setpts? [02:03] <norbert> I have it running already and I'm not sure this will give me audio [02:03] <norbert> maybe at the end of the transcoding it adds audio to the file [02:04] <norbert> I'm trying to use ffmpeg because kdenlive removes audio when I speed up the video with the available motion effect [02:05] <norbert> which makes me a sad Linux panda [02:05] <norbert> but one day things will work out of the box [02:06] <norbert> no, transcoding is done, but no audio [02:06] <c_14> What's your current command, and what do you want to do. [02:06] <c_14> Ie speed up/slow down by what amount. [02:06] <norbert> I can extract the audio as a wav and work on that separately, but, yeah [02:07] <norbert> I tried: ffmpeg -i ToT_2.mp4 -r 60 -filter:v "setpts=0.125*PTS" -an output.mkv [02:07] <norbert> I want that but with audio [02:08] <c_14> add -filter:a 'asetpts=0.125*PTS' and remove the -an [02:08] <norbert> I'll try it with -acodec libvo_aacenc [02:09] <c_14> If you don't want to drop frames, use -r 240 [02:09] <norbert> doesn't appear to speed up the audio [02:10] <norbert> adds audio though, so one step forward :) [02:10] <norbert> if we manage to find out how to do it... [02:11] <norbert> ...maybe I should e-mail ffmpeg.org folks to update https://trac.ffmpeg.org/wiki/How%20to%20speed%20up%20/%20slow%20down%20a%20… [02:11] <norbert> because I'm guessing almost everyone who wants this still wants audio to work - and also sped up [02:11] <c_14> I can edit the page. [02:11] <c_14> It should work though. [02:11] <c_14> What's your current commandline. [02:12] <c_14> s/\./? [02:12] <norbert> ffmpeg -i ToT_2.mp4 -r 60 -filter:v "setpts=0.125*PTS" -filter:a "asetpts=0.125*PTS" -r 240 output.mkv [02:12] <c_14> eh [02:12] <c_14> atempo [02:12] <c_14> not asetpts [02:12] <c_14> just use atempo=4 [02:12] <c_14> I forgot that the name was different... [02:13] <norbert> http://justpaste.it/hpki [02:14] <c_14> use atempo=2,atempo=2 [02:14] <c_14> oh, and atempo filters are multiplicative, not additive [02:14] <c_14> Important to remember. [02:15] <norbert> I removed -r 240 [02:15] <norbert> because with atempo=etc is was now 240fps [02:15] <norbert> without -r 240 it goes back to 60 [02:15] <norbert> the result has faster audio but it's not synchronized [02:15] <norbert> audio not fast enough, it seems [02:16] <norbert> about 1/2 what it needs to be [02:16] <norbert> what exactly does 0.125 do, why does that make it 4x faster [02:17] <norbert> I'd think that would make something 8x faster/slower [02:17] <norbert> I'm now trying it double speed, with: ffmpeg -i ToT_2.mp4 -r 60 -filter:v "setpts=0.250*PTS" -filter:a 'atempo=2' output.mkv [02:17] <norbert> but the audio is still too slow [02:19] <c_14> eh [02:19] <c_14> It would make it 8x faster. [02:19] <c_14> I'm pretty sure that's a typo on the wiki. [02:19] <c_14> Will fix. [02:20] <norbert> why does (a)tempo not accept things outside the 0.5-2 range? [02:20] <c_14> Excellent question. [02:20] <c_14> Next question. [02:20] <norbert> I think to make the audio twice as fast, I need atempo=4, for some reason [02:20] <c_14> I'd have to read the source code for that one. [02:21] <c_14> atempo=2 would make the audio twice as fast aka 100% faster [02:21] <norbert> maybe because the setpts speeds audio up [02:21] <norbert> but it doesn't [02:21] <c_14> The setpts doesn't touch audio samples. [02:21] <norbert> ok [02:22] <norbert> anyways, so I'm now using: ffmpeg -i ToT_2.mp4 -r 60 -filter:v "setpts=0.250*PTS" -filter:a 'atempo=2' output.mkv [02:22] <norbert> and while audio is faster, it's clearly way slower than the video [02:22] <norbert> oh wait [02:22] <c_14> That's because you're making the video 4x faster and the audio 2x faster. [02:22] <norbert> 0.500 is what I would need for setpts [02:23] <norbert> but I want it 0.250 with atempo=4 :) [02:23] <c_14> use atempo=2,atempo=2 [02:24] <norbert> thanks, now the result is as I wanted [02:47] <norbert> c_14: maybe add -filter:a 'atempo=2,atempo=2' ? [02:48] <norbert> to the wiki, I mean :) [02:48] <c_14> It's listed under speeding up/slowing down audio [02:49] <c_14> I could add a section speeding up/slowing down video + audio if you think that would be easier to understand. [02:49] <norbert> oh, man, I must be asleep [02:49] <norbert> no, it's just late here, I wasn't paying attention :) [02:51] <deivor__> I have some files with de-synced audio/video (initially, doesn't get worse with time). Currently I'm "eyeballing" the difference, splitting the files in audio and video tracks and joining them with -itsoffset N. This works but is very "manual". Is there any way to do it faster? [02:51] <deivor__> I've set up a script that does this but maybe I'm just doing it the wrong way [02:51] <deivor__> (And with 'faster' I don't mean encode time) [02:51] <c_14> use -codec copy ? [02:52] <deivor__> I mean the way that I'm doing it (split, join with itsoffset) seems 'wrong'. My script takes a filename and an offset and splits it and joins it back with the offset [02:52] <deivor__> I thought maybe there was another ("proper") way to do it [02:54] <c_14> you could give ffmpeg the original file twice, put itsoffset in front of one of the inputs, and then just map the streams as you want them [02:54] <c_14> That'll save you the extracting to separate files part. [02:55] <c_14> You might also be able to use -async/-vsync [07:08] <ruby_on_tails> i combined a mp4 video and a mp3 audio into a new mp4 video, when i play the video on my ipad chrome the audio doesnt work, if i play the audio file separately on chrome it works [07:11] <ruby_on_tails> http://pastebin.com/S86GDZDG this is the ffprobe of the final video which isnt working [07:16] <ruby_on_tails> it works on desktops, mac, android mobile [07:16] <ruby_on_tails> just not working properly on ipad [07:47] <ruby_on_tails> anyone alive? [08:13] <voltagex> is it possible to turn DASH video back into something more normal without loss of quality? [09:00] <rcombs> voltagex: sure, remux it [09:03] <sanooj> anyone used uvc to export some of its controls to v4l2? [11:14] <plutarde> Does pure ffmpeg support variable frame rate (VFR) to encode? [12:04] <nikolasborrel> Hi. Im trying to install ffmpeg 2.4.2 on a Debian server following the instructions given here: http://www.e-rave.nl/installing-ffmpeg-on-debian-squeeze-and-newer [12:04] <nikolasborrel> When configuring ffmpeg I get [12:04] <nikolasborrel> ERROR: freetype2 not found. [12:04] <nikolasborrel> Do you have a solution for that? freetype is installed at /usr/include/freetype2/freetype [12:06] <sanooj> yes! apt-get install libfreetype2-dev or so [12:07] <Akagi201> Which lib should I link when I compile ffmpeg I want to use G711 audio. [12:07] <Akagi201> Any help is very nice. [12:07] <sanooj> actually what you want to do is to apt-get build-dep ffmpeg [12:07] <sanooj> ^ nikolasborrel [12:08] <Akagi201> sanooj: I compile from source [12:09] <Akagi201> I use it on a device, so I want to know the exactly lib [12:09] <sanooj> yeah sorry Akagi201, I wasn't talking to you. I don't know about G711 [12:09] <Akagi201> nothing [12:09] <sanooj> nikolasborrel: it's libfreetype6-dev, not libfreetype2-dev. [12:11] <nikolasborrel> Thanks sanooj. libfreetype6-dev is already the newest version.. I have found this post, it might be related& http://ffmpeg.org/pipermail/ffmpeg-user/2014-July/022603.html [12:12] <nikolasborrel> I need support for m4a and as I understand, the version installed using apt-get will not support this due to legal issues? [12:12] <nikolasborrel> *apt-get install ffmpeg [12:12] <c_14> nikolasborrel: `locate freetype2.pc' [12:14] <c_14> Akagi201: pcm_mulaw and pcm_alaw should be ffmpeg-internal [12:14] <Akagi201> c_14: you mean don't have to --enable-libxxx when compile the code [12:16] <c_14> yep [12:17] <nikolasborrel> c_14: freetype is installed at /usr/include/freetype2/freetype [12:18] <c_14> Is the freetype2.pc there as well? [12:19] <c_14> The configure script uses pkg-config to determine whether it exists and what flags are needed. [12:19] <c_14> If it can't find the freetype2.pc file, it won't find the lib. [12:21] <nikolasborrel> c:14: freetype2.pc is located here /usr/lib/x86_64-linux-gnu/pkgconfig/freetype2.pc [12:22] <c_14> `export PKG_CONFIG_PATH=/usr/lib/x86_64-linux-gnu/pkgconfig' then try running pkg-config --exists --print-errors freetype2 [12:23] <nikolasborrel> Im installing apt-get build-dep ffmpeg at the moment as suggested by sanooj, but will it include the acc/m4a codecs? [12:24] <nikolasborrel> *aac [12:25] <c_14> build-dep doesn't install ffmpeg, it only installs the build dependencies thereof [12:26] <nikolasborrel> pkg-config --exists --print-errors freetype2 doesnt output anything. Is that good&? [12:26] <c_14> yep [12:26] <c_14> Now try running configure again. [12:30] <kevmitch> nikolasborrel: do the build dep, then add libfdk-aac-dev and configure ffmpeg with --enable-nonfree --enable-libfdk-aac [12:30] <kevmitch> . [12:35] <nikolasborrel> c_14: I still get the same problem with freetype2 not found. Tough stuff :-( [12:36] <c_14> nikolasborrel: You did that in the same terminal where you exported PKG_CONFIG_PATH ? [12:36] <nikolasborrel> Yes [12:39] <nikolasborrel> Hmm& Something happened now! [12:42] <nikolasborrel> The build-dep ffmpeg is crunching in another terminal, dont know if its doing good thing for my build [12:43] <nikolasborrel> However, I get a bunch of errors when running make. I will try kevmitch suggestion when build-dep has finished [13:10] <nikolasborrel> I have problems installing build-dep ffmpeg: spamassassin fails with Cant locate HTML/Parser.pm. I then tried to install using perl: install Mail::SpamAssassin but I get undefined symbol: Perl_Gthr_key_ptr [13:10] <nikolasborrel> Any Perl hackers out there&? [13:11] <nikolasborrel> I tried deleting /usr/local/lib64/perl5/auto/version/ without luck [13:12] <c_14> SpamAssassin isn't a dependency of FFmpeg... [13:12] <c_14> That would be strange. [13:14] <nikolasborrel> Setting up spamassassin (3.3.2-5+deb7u1) & after sudo apt-get build-dep ffmpeg [13:14] <sanooj> awesome [13:15] <phonie> has anyone expirienced glith-corruption of file with 2-pass libvpx-vp9? [13:15] <sanooj> anikolasborrel: that's very weird. I don't have spamassasin installed even though I've run that command [13:17] <sanooj> nikolasborrel: I'm running an updated debian wheezy. [13:17] <moroni> guys. [13:17] <moroni> has anyone expirienced glith-corruption of file with 2-pass libvpx-vp9? [13:18] <moroni> happens https://pp.vk.me/c618119/v618119557/1e9bc/hbIvQ-4s4xk.jpg [13:19] <nikolasborrel> Im trying to install on a Dreamhost VPS server& Dont know whats installed here. [13:20] <JEEB> moroni, I think vp9 libvpx is really broken in parts but you'd want to make sure it's current code you're testing and that the issue is really the encoder [13:20] <nikolasborrel> Should I upgrade Debian? I have never worked with debian before.. [13:21] <sanooj> probably not. [13:23] <c_14> nikolasborrel: I'd probably `apt-get autoremove' and then try following this: https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu [13:24] <moroni> JEEB, honestly, just tried to play in Chrome. all fine. [13:25] <JEEB> are you up-to-date with the lavc decoder then? [13:34] <maui> Hi, is it possible to convert mp4 lossless to webm? [13:35] <c_14> maui: only if the mp4 is vp8 or vp9 [13:36] <maui> where can i see that? [13:36] <c_14> ffprobe input.mp4 [13:37] <moroni> codec not supported for mp4 container. [13:38] <moroni> maui, if you want to losslessly convert, you can use -b:v 0 -crf 0. just file size issue. [13:39] <c_14> moroni: do vp8/vp9 have a lossless mode? [13:39] <JEEB> moroni, -b:v 0 is incorrect in any case, and c_14's question is completely valid [13:40] <JEEB> not to mention that no-one knows if crf zero even maps to something that could be lossless with libvpx [13:40] <moroni> JEEB, elsehow bitrate setted to 200 kbps. [13:40] <moroni> i know nothing. it is raw. but usable. [13:40] <JEEB> with libx264, yes, crf 0 with a 8bit encoder is lossless [13:40] <JEEB> but as soon as you go for f.ex. 10bit libx264 CRF 0 is no longer lossless [13:41] <JEEB> you have to use quantizer zero with libx264 to make sure it's lossless in that case [13:41] <JEEB> (the range of values became bigger and CRF was matched so that the lossy values would more or less match) [13:41] <JEEB> in any case, I don't even know if libvpx has a lossless mode :P [13:42] <JEEB> so the simple answer is, "no you cannot do lossless conversions from content that is muxed into mp4 to content that is muxed into webm [matroska subset]" [13:42] <JEEB> mostly because mp4 doesn't have formats supported by the latter subset defined so there's no case of "I can just copy the streams over" [13:42] <nikolasborrel> UPDATE: I managed to install with great help from you guys. Thanks a lot! Dont know exactly what did the trick, but `export PKG_CONFIG_PATH=/usr/lib/x86_64-linux-gnu/pkgconfig, build-dep ffmpeg and some fiddling around did the trick. THANKS! [13:43] <voltagex> JEEB: where can I go to read up on this? I'm interested in lossless video for archival [13:43] <JEEB> voltagex, regarding what? [13:43] <sanooj> sigh. so I got uvcdynctrl to teach v4l2 about the iframeperiod control. it can read the current value and limits as a v4l2 control. wootwoot [13:43] <JEEB> which encoder etc [13:43] <sanooj> but it can't write to it. it always writes zero, and it doesn't say why [13:43] <voltagex> JEEB: I never knew there were lossless settings for x264 [13:43] <JEEB> just set quantizer to zero and that is lossless [13:43] <sanooj> headdesk [13:44] <JEEB> crf set to zero works with a 8bit build [13:44] <JEEB> but it will not be lossless with 9bit or 10bit builds [13:44] <JEEB> (the bitness of output, which is decided when you build libx264) [13:57] <moroni> (VP9-Specific Options --lossless=<arg> Lossless mode) [14:11] <maui> what filter can i use if i want to cut 50 pixel off the bottom? [14:11] <c_14> crop? [14:12] <c_14> hmm, although [14:12] <c_14> nah, that should work [14:12] <maui> but crop also cuts the top side? [14:13] <c_14> crop=iw:ih-50:0:0 <- should do it [14:14] <maui> thx [16:25] <Lac3rat3d> c_14 you around? [16:25] <c_14> Sure. [16:25] <Lac3rat3d> :D [16:26] <Lac3rat3d> so that thing you were helping me with yesterday worked, burned the subtitles into the video stream [16:26] <Lac3rat3d> but, the bitrate went way down from source to output. how can i specify same bitrate as input? [16:26] <c_14> You're encoding x264? [16:26] <Lac3rat3d> yep [16:27] <c_14> https://trac.ffmpeg.org/wiki/Encode/H.264 <- you can find useful info here [16:27] <Lac3rat3d> ya i've been looking at that [16:27] <Lac3rat3d> but doesn't seem like you can just specify use whatever bitrate is in source [16:28] <c_14> Nah, you'd have to guess at the source bitrate using ffprobe and subtracting the audio bitrate. [16:28] <Lac3rat3d> source is x264 as well [16:28] <Lac3rat3d> hmmm okay [16:29] <Lac3rat3d> how do you use ffprobe? [16:30] <c_14> ffprobe file [16:31] <Lac3rat3d> oh, it's the same output as ffmpeg -i file [16:31] <c_14> Minus one error message, yes. [16:31] <c_14> You can also give ffprobe options that make the output easier to parse though. [16:32] <Lac3rat3d> seems like source is ~11mb/s [16:32] <Lac3rat3d> vbr though [16:34] <c_14> Either pick something like -b:v 11m or pick a crf that gets you something similar. [16:35] <c_14> (rule of thumb, start in the low 20s and adjust as necessary) [16:35] <Lac3rat3d> hmm ok, i'll play around with it :) [16:35] <Lac3rat3d> thanks! [17:44] <kaotiko> hi [18:01] <joonas_> hi kaotiko [18:11] <ggVGc> I have some AVI files that I need to have in mpg. is it possible to do this transcoding without/with very little quality loss, if I don't care about output size? [18:12] <ggVGc> or does mpeg4 just really suck so I can't have it? [18:18] <c_14> ffmpeg -i avi -c copy out.mpg [18:26] <ggVGc> c_14: yeah, I tried it. I guess I need to reencode anyway. I am trying to get the avi to play on a device that only supports "All mpeg formats", but it's old. So I am trying to find out the best way to transode with the least quality loss into a format it understands :( [18:26] <ggVGc> when I used vcodec out, and out it in a .mpg, it found the file but didn't play it [18:26] <ggVGc> ehh, vcodec copy, I meant [18:29] <ggVGc> oh, damnit, the original stream is xvid :/ [18:30] <ggVGc> so, how do I reencode in mpeg4 with the absolute highest quality(disregarding file size)? [18:32] <sacarasc> XviD is an implementation of MPEG4. [18:33] <ggVGc> sacarasc: yeah, but the box that I am trying ot use doesn't seem to understand it for some reason [18:33] <ggVGc> but it understands if I use -vcodec mpeg4 in ffmpeg [18:33] <ggVGc> from the xvid stream [18:33] <ggVGc> not sure why [18:33] <ggVGc> maybe I am doing something else wrong [18:34] <sacarasc> ffmpeg -i xvid-input.avi -c copy -vtag DIV3 output.avi [18:34] <sacarasc> Try that. [18:34] <ggVGc> sacarasc: so, can I make ffmpeg transcode the xvid stream into mpeg4 without loss? (basically just changing the name of the code) [18:34] <ggVGc> okay, thanks [18:35] <sacarasc> Yeah, in theory. Just gotta find one that works, I guess. [18:36] <ggVGc> I really dislike video conversion :( [18:37] <ggVGc> hm, that's didn't work [18:37] <ggVGc> maybe it's the resolution that is the problem [18:37] <ggVGc> sacarasc: what does vtag DIV3 mean? [18:38] <ggVGc> i.e what is DIV3 [18:38] <JEEB> divx ;-) [18:38] <sacarasc> DivX 3, which was a rather old encoder of MPEG4. [18:38] <ggVGc> alright, so that should have worked then I guess [18:38] <JEEB> IIRC a hacked MS MPEG-4 encoder that let you output it into AVI [18:38] <ggVGc> JEEB: but I am trying to get a divx stream to not be divx [18:38] <ggVGc> without losing quality [18:39] <ggVGc> and my target container is mpg, not avi. [18:39] <JEEB> well that divx is not really MPEG-4 Part 2 following [18:39] <JEEB> since the old hacked MS encoder wasn't really compliant, IIRC [18:39] <ggVGc> it works if I output with -vcodec mpeg4, but then I get quality loss because of transcore [18:40] <ggVGc> transcode* [18:40] <ggVGc> a lot of quality loss [18:40] <ggVGc> even with very high bitrate [18:42] <flarunt> if your device is old and doesnt play divx then id assume by "all mpeg formats" it means mpeg1/2 [18:45] <ggVGc> flarunt: I used -b:v 2000k now, and outputted to .mpg, and it worked [18:45] <ggVGc> but the file doubled in size [18:45] <ggVGc> but that's fine [18:45] <ggVGc> thanks guys [19:39] <joonas_> alright. uvcdevctrl was a flop. I can teach it about the XU provided by uvcvideo that controls the i-frame period, but can't set it using v4l2-ctl. [19:39] <joonas_> it reads it and the bounds correctly, but setting the value always stores zero [19:40] <joonas_> gstreamer can talk to uvcvideo directly to set the bitrates and i-frame period, but getting it to add audio to an rtmp stream is proving to be a nightmare. [20:19] <DrSlony> Hi, please advise what settings to use for high quality x264 compression so that compatibility with youtube, modern smartphones and other modern players is maintained. [20:22] <DrSlony> se topic [20:24] <DrSlony> ignore "se topic" [20:32] <sacarasc> DrSlony: Find the oldest devices you want to support and check what they allow. [20:39] <DrSlony> thats not realistic advice [20:39] <DrSlony> "find what they support" [20:39] <JEEB> well, there are two things you have to take into mention when you want to support some hardware device [20:39] <JEEB> profile and level [20:39] <JEEB> profile you can think of as features [20:39] <JEEB> level is how much memory that decoder has, basically [20:39] <DrSlony> modern smartphone [20:40] <DrSlony> *modern smartphones [20:40] <DrSlony> and youtube [20:40] <JEEB> is it ok for iphones if it only works from 3GS forward? [20:40] <JEEB> if yes, then go for not setting a profile (it will set one by the amount of features you use), and level 4.1 [20:40] <JEEB> youtube isn't a limiting factor at all because it re-encodes anyways [20:41] <JEEB> it just om nom noms whatever you feed it [20:41] <JEEB> (it uses an old version of ffmpeg for the decoding) [20:41] <DrSlony> yes 3GS is fine [20:42] <JEEB> well, then it should be simple enough as I said [20:42] <DrSlony> yes but the last time i did video conversion, some 2 years ago, using advice i got from here i ended up with h.264 videos that youtube couldnt decode :) so now while asking i'd like to cover that base :) [20:42] <JEEB> the only problem are the bulkloads of android devices because even newer ones from china can have very crappy hardware decoders [20:42] <JEEB> well, I do remember that until some years ago youtube didn't support lossless H.264 [20:43] <JEEB> since they never updated their ffmpeg decoder [20:43] <SirVivorX> so, here is a stupid question indeed ( I do plead stupidity in this situation...), but what file do i use to install ffmpeg on my mbp 10.10 (no server) with? But with some minor help here, I will donate $$$ to the cause :-) [20:43] <SirVivorX> I just don't see it [20:43] <JEEB> SirVivorX, just use homebrew and install ffmpeg [20:43] <JEEB> http://brew.sh/ [20:43] <DrSlony> thank you JEEB, i will get back to you in a few minutes [20:43] <SirVivorX> d/l homebrew for ffmpeg? [20:44] <JEEB> you install homebrew, and then tell it to build you ffmpeg [20:44] <JEEB> and you will get yourself ffmpeg [20:44] <JEEB> homebrew is the least retarded package manager for OS X [20:44] <SirVivorX> sounds easy if I knew what to do...lol [20:44] <SirVivorX> ok, thx [20:47] <SirVivorX> http://www.renevolution.com/how-to-install-ffmpeg-on-mac-os-x/ [20:47] <SirVivorX> k, thx :-) [20:47] <JEEB> ok, yes. that doesn't sound too retarded [20:48] <DrSlony> JEEB i'm on ffmpeg-2.2.9, do you know of any issues i should look out for? [20:49] <DrSlony> all versions newer than that are hard-masked in gentoo as they cause issues with other software [20:49] <JEEB> just make sure they don't cause issues with anything you actually have installed [20:49] <JEEB> because usually the reasons for not updating in gentoo with libav/ffmpeg are rather iffy. [20:50] <JEEB> but yeah, I don't know of any specific issues, but how would I know if I use the latest all the time :P [20:50] <JEEB> no-one really keeps any eyes on anything else but the latest release, if even that [20:50] <JEEB> the releases are purely for distros [20:51] <DrSlony> yes i know, im a dev too :) [20:52] <JEEB> you would really like the automagic level following though [20:52] <JEEB> and I don't remember when that went in [20:52] <JEEB> basically, before -level 4.1 would only set the flag [20:52] <JEEB> and not actually make sure that the stream you were pushing out would comply to it [20:52] <JEEB> (regarding the amount of reference frames used) [20:53] <JEEB> then at some point finally after years of talking about it someone did something about it [20:53] <JEEB> and now ffmpeg's -level does the same as x264cli's --level [20:53] <JEEB> as in, it limits your reference frames according to the level limitations [20:54] <JEEB> http://git.videolan.org/?p=ffmpeg.git;a=commit;h=0aac9b76bc7770a90d1317f997… [20:54] <JEEB> may 2014 [20:54] <JEEB> although to be honest, you could have a separate ffmpeg binary with statically linked in libraries if you're afraid of updating your whole system :) [20:55] <JEEB> if you compile in the repo you will get exactly that [20:55] <JEEB> with the default configuration [20:57] <DrSlony> i could try that [20:58] <DrSlony> by the way, is this command sane? [20:58] <DrSlony> ffmpeg -i in.mp4 -ac 2 -c:a libfdk_aac -global_quality 3 -c:v libx264 -preset slower -crf 23 -level 4.1 -threads 0 out.mp4 [20:58] <JEEB> will you be putting this so that the output is as-is available to someone via limited bandwidth? [20:58] <JEEB> such as the internet [20:59] <DrSlony> i like your website now, its clear [20:59] <DrSlony> well i intend to use this on all my old videos recorded with a bridge type camera, tablet and smartphone [20:59] <cortexman> can i output a wav as float values between 0 and 1? [21:00] <DrSlony> so the input is generally not the best quality and a bit grainy [21:00] <DrSlony> i dont care for preserving grain [21:00] <JEEB> well, I asked you a simple question [21:00] <DrSlony> and the destination will be probably google drive [21:00] <JEEB> and the answer shall be yes or no [21:00] <DrSlony> ah [21:00] <DrSlony> yes [21:00] <JEEB> ok, then you also need VBV [21:00] <JEEB> which is -maxrate and -bufsize [21:00] <DrSlony> verifie dby visa [21:01] <cortexman> when i convert to pcm using sox it is in some kind of binary format still [21:01] <cortexman> i want just a float array [21:01] <JEEB> basically set maxrate to the minimum network speed you want someone to be able to watch the clip with without further buffering [21:01] <JEEB> and bufsize is the amount of buffering the player (in general) would do, you have to match it between the player and the encoder [21:02] <JEEB> if you have a player that gives you the amount of buffering in seconds instead of bits [21:02] <JEEB> then <seconds to buffer>*<maxrate> [21:02] <JEEB> is the way to calculate it [21:02] <JEEB> also about the sanity of that command line: threads auto (zero) is the default for libx264, global quality I have no idea of, make sure it only affects audio if that's what you are wishing to use it for [21:03] <JEEB> otherwise it looks ok, you should adjust the preset according to your wishes regarding speed vs compression [21:03] <JEEB> and crf should be adjusted according to what you like eye-wise [21:03] <JEEB> CRF does the primary rate control with a constant rate factor (closest we have right now to 'constant quality' kind of thing), and VBV will limit it so that it will not go over the maxrate/bufsize model you set [21:04] <JEEB> also I've actually always set the level without the dot, but it should work with the dot as well :) [21:06] <JEEB> you should probably test without VBV first a few shorter clips and try to find your sweet spot regarding CRF [21:06] <JEEB> do note that the value that is your sweet spot will depend somewhat on the type of content, and more with the frame rate and the resolution of it [21:06] <JEEB> although usually after you have found your sweet spot once, it's not too far even if things change [21:06] <cortexman> does anyone know how to get ffmpeg to output a pcm file as an actual array of floating point values... [21:07] <DrSlony> i have tested crf and preset in the past, and these are the settings that i found satisfactory [21:07] <JEEB> well, then add VBV to it and that'd be it I guess [21:08] <JEEB> and you can remove at the very least the threads one, and make sure what the hell the global_quality thing does [21:08] <JEEB> if it has anything to do with video, you should remove it [21:08] <DrSlony> yes im just checking blobal_quality - i wrote this down 2 years ago without an explanation of why i chose "3" [21:09] <DrSlony> i cant find an explanation of why i chose 3, so i deleted it and im going with this: [21:09] <DrSlony> -c:a libfdk_aac -profile:a aac_he -b:a 128k [21:09] <mark4o> cortexman: ffmpeg -i inputfile -f f32le out.pcm [21:10] <JEEB> DrSlony, at that bit rate you shouldn't need HE-AAC [21:10] <JEEB> you can leave the profile out [21:10] <cortexman> mark4o, this is the output https://gist.githubusercontent.com/brianmingus/c698eeaf6b9d1a31b110/raw/334… [21:10] <JEEB> HE-AAC starts being useful when you go way under 100kbps [21:11] <cortexman> (not a float array..) [21:12] <mark4o> cortexman: that looks like a float array & dumped as text [21:13] <cortexman> not useful but thanks [21:13] <DrSlony> thanks JEEB! [21:14] <DrSlony> ffmpeg -i in.mp4 -ac 2 -c:a libfdk_aac -b:a 128k -c:v libx264 -preset slower -crf 23 -level 4.1 -maxrate 512k -bufsize 2560k out.mp4 [21:14] <JEEB> seems like a bit low maxrate, but if you feel like that's the thing you'll be going with, fine by me :) [21:15] <JEEB> not that I know what kind of content you're encoding [21:30] <kenny__> Hello, there is possibility to set input for ffserver as udp source (not only feed)? [21:31] <DrSlony> ah yes, perhaps 2000k for maxrate is better? [21:31] <DrSlony> the input is video captured using smartphones, tablets, bridge cameras [21:32] <kenny__> i have av_interleaved_write_frame(): Connection reset by peer error if i try send stream from ffmpeg to ffserver via http :/ [21:33] <kenny__> i also try with rtsp but with error: Could not write header for output file #0 (incorrect codec parameters ?): Invalid data found when processing input [21:36] <kenny__> DrSlony: i need to transcode multicast input stream (with ffmpeg), pass them to ffserver and distribute it to enc-clients. ffserver is on other server than ffmpeg [21:38] <DrSlony> kenny__ hi, im not the right guy to ask :) [21:39] <kenny__> ok :) [21:39] <kenny__> anyone? :( [21:43] <xconspirisist> Hi folks, I'm trying to understand why raw footage from my GoPRO is so large, and why re-encoding at the same resolution reduces the size. ffprobe foo.mkv = Video: h264 (Main), yuvj420p(pc, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 59.94 fps, 59.94 tbr, 1k tbn, 119.88 tbc (default) [21:50] <DrSlony> well the gopro cpu is not as powerful as your desktop computer's so i cant spend as much time compressing [21:51] <DrSlony> further, ffmpeg might use newer techniques than the old firmware does [21:51] <DrSlony> further, you will lose quality on every recompression [21:52] <JEEB> with the large exception of using lossless coding, but generally people don't use it [21:52] <iive> libx264 is the encoder that ffmpeg uses and it is state-of-the-art one. [21:53] <xconspirisist> DrSlony: ah, so does that line from ffprobe say that a codec is being used, and simply, ffmpeg would just do a better job of encoding if I used the same codec again? [21:54] <xconspirisist> DrSlony: I had assumed it was pretty much like a "null" codec on the gopro/video cameras, and it was storing every frame in full? [21:54] <JEEB> you can take the video format out of the equation, basically simply put what you used with ffmpeg compressed better than whatever gopro did [21:55] <JEEB> of course, just being able to compress something better doesn't mean anything by itself, but if you cannot notice the difference then it was done without affecting visual quality for your eyes [21:56] <xconspirisist> JEEB interersting stuff. [21:57] <JEEB> cameras generally don't create files for end user consumption so they don't need to compress things well. and they usually have enough free space to then compensate by just using more bits [21:58] <JEEB> so being able to compress something more than a camera could with similar picture quality isn't exactly anything surprising per se [21:58] <xconspirisist> At the moment I'm using the concat "demuxer" which is awesome, it's nice and fast, just it's producing outputs with a filesize the sum of all input files (as you'd expect from concat) [21:58] <xconspirisist> that works super well, just I guess now I need to re-encode to get a smaller file that I can share around. [21:59] <DrSlony> xconspirisist resolution has nothing to do with filesize when compression is involved [21:59] <xconspirisist> yup, sure [21:59] <JEEB> well, it both has and doesn't have [22:00] <JEEB> but in general if you're not encoding a very simple scene a larger resolution picture will need more bits to be encoded at the same quality level compared to a smaller resolution picture [22:02] <kenny__> anyone can help? [22:02] <JEEB> this is of course under the presumption that everything else is static :P [22:03] <JEEB> if you encode a lower resolution picture with very ineffective ways and then encode a higher resolution picture or pictures in more effective ways, then of course the bigger picture might end up smaller [22:11] <DrSlony> JEEB latest git ffmp\eg compiled without issue s:) [22:11] <DrSlony> no issues other than my typos [00:00] --- Mon Oct 27 2014

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