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April 2015

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ffmpeg-devel.log.20150414
by burek 15 Apr '15

15 Apr '15

[02:32:45 CEST] <cone-579> ffmpeg 03Michael Niedermayer 07master:89b3c5215a51: avcodec/atrac3plusdec: Free mdct contexts at the end [03:10:40 CEST] <Timothy_Gu> pengvado: thx. seems like a bug in Haiku's gcc then. [03:17:50 CEST] <cone-579> ffmpeg 03Carl Eugen Hoyos 07master:d0c895d3c585: lavf/mpegtsenc: Accept 0x000001 as startcode for hevc. [03:17:51 CEST] <cone-579> ffmpeg 03Michael Niedermayer 07master:42a144f2c8a4: Merge remote-tracking branch 'cehoyos/master' [03:48:51 CEST] <cone-579> ffmpeg 03Rostislav Pehlivanov 07master:013498ba15ae: aacenc: Adjust the initial offset for PNS values [12:59:42 CEST] <michaelni> Timothy_Gu, http://fate.ffmpeg.org:8080/log/aarch64-linux-qemu-ubuntu-gcc-4.8-n2.6/2015… <-- looks like binary giberish [13:01:13 CEST] <michaelni> fate also should show some overall statistics of passsed/failed for what is displayed, fate currently does with its bar graph thing at the top [13:01:25 CEST] <michaelni> this allows one to quickly see if there are new failures [13:04:22 CEST] <michaelni> in the results column in old fate in cells with failures theres a tiny black triangle one can click on to get a list of failures that avoids having to open the next page and search for what failed [13:07:49 CEST] <michaelni> http://fate.ffmpeg.org:8080/report/x86-opensolaris-suncc5.10-n2.4/201504120… "Partially failed: 2226 / 2226 passed! <-- what failed ? (no old fate doesnt say either i think but it displays as success which may be wrong) [13:40:56 CEST] <cone-487> ffmpeg 03wm4 07master:2de7650451d9: mmal: Move system headers before local headers [13:40:56 CEST] <cone-487> ffmpeg 03Michael Niedermayer 07master:7c521d452892: Merge commit '2de7650451d90520135d8cf6f96caa0658824208' [13:57:10 CEST] <cone-487> ffmpeg 03wm4 07master:49e531ff4706: mmal: Reference MMAL VC lib explicitly [13:57:11 CEST] <cone-487> ffmpeg 03Michael Niedermayer 07master:6e86ee97c8ee: Merge commit '49e531ff4706279c7eed774f2c8b896942924ce2' [14:11:59 CEST] <cone-487> ffmpeg 03wm4 07master:13e4f2c7f785: mpeg4videodec: Remove useless messages [14:12:00 CEST] <cone-487> ffmpeg 03Michael Niedermayer 07master:410317237663: Merge commit '13e4f2c7f7857feb6885f0c2ea8c9ddf438f46ea' [14:42:18 CEST] <cone-487> ffmpeg 03Michael Niedermayer 07master:bca1577ed3c4: avcodec/mpeg4videodec: change log level to waning for incomplete header messages [14:42:19 CEST] <cone-487> ffmpeg 03Michael Niedermayer 07master:4ba662b0320b: avcodec/mpeg4videodec: Improve wording of the missing header warning messages [15:21:18 CEST] <cone-487> ffmpeg 03Paul B Mahol 07master:c2fc46635131: avcodec/alsdec: use av_malloc(z)_array() [15:45:27 CEST] <cone-487> ffmpeg 03Michael Niedermayer 07master:343f34c0b8f4: avcodec/get_bits: print details of the location of the missing bit in check_marker() [15:45:28 CEST] <cone-487> ffmpeg 03Michael Niedermayer 07master:37d93fdbf0fe: avcodec/mpeg12dec: Use check_marker() [16:17:09 CEST] <Timothy_Gu> michaelni: yes I am aware of these two issues (im experimenting) [16:17:55 CEST] <Timothy_Gu> michaelni: I intentionally left opensolaris that way because if you readd the log there is a segfault that is not counted as a test [16:18:51 CEST] <Timothy_Gu> michaelni: log fixed for now [16:31:04 CEST] <michaelni> thx [18:16:18 CEST] <cone-487> ffmpeg 03Rostislav Pehlivanov 07master:a61c75e9f77a: aacdec: Use macros for constants [20:08:10 CEST] <cone-487> ffmpeg 03Stephen Hutchinson 07master:a6a45e5a9a9d: avisynth: Fix detection of AviSynth 2.5 [20:08:11 CEST] <cone-487> ffmpeg 03Michael Niedermayer 07master:64b37d3e08e4: Merge commit 'a6a45e5a9a9dfb725456f62e053532e505aadcfc' [20:08:36 CEST] <lglinskih> kierank: I sent my patch in the morning and still don't have any Aomments. Maybe I did something wrong?=( Here are my letters: https://ffmpeg.org/pipermail/ffmpeg-devel/2015-April/171754.html https://ffmpeg.org/pipermail/ffmpeg-devel/2015-April/171755.html [20:11:26 CEST] <Timothy_Gu> lglinskih: FFmpeg developers have lives :) Be patient, and if still nobody replies "ping" it (sending an email with content "ping" in the patch thread) [20:20:06 CEST] <wm4> I'll look at it too, after I'm done fighting with mp3 [20:20:34 CEST] <Daemon404> you cannot win [20:20:39 CEST] <Daemon404> everything is mp3 [20:20:40 CEST] <Daemon404> even you. [20:21:53 CEST] <lglinskih> Timothy_Gu: =) [20:43:24 CEST] <wm4> so uh what's the quickest way to compare two wav files with some fuzzy tolerance? [20:43:49 CEST] <Daemon404> tiny_psnr [20:43:49 CEST] <Daemon404> ? [20:45:14 CEST] <BtbN> You could go and substract them, no idea if any tool exists to do so. [20:45:36 CEST] <cone-487> ffmpeg 03Stephen Hutchinson 07master:7a5f44476696: avisynth: Bump minimum required version to interface version 6 [20:45:37 CEST] <cone-487> ffmpeg 03Michael Niedermayer 07master:ee69229dd806: Merge commit '7a5f44476696b495e0de658ed859bcb8520b93d4' [20:45:42 CEST] <wm4> tiny_psnr works nicely, and even supposedly skips wav headers [20:56:06 CEST] <cone-487> ffmpeg 03Vignesh Venkatasubramanian 07master:30e2f87d2eac: webdashenc: replace unchecked av_malloc with stack allocation [21:10:05 CEST] <kierank> lglinskih: sorry I am a bit busy today, will look tomorrow [21:44:20 CEST] <cone-487> ffmpeg 03wm4 07master:2b5f34f788d8: avformat, avcodec: log discard padding [21:53:18 CEST] <cone-487> ffmpeg 03wm4 07master:95f524aff513: lavc: use correct type for printf() argument [23:05:47 CEST] <rcombs> wm4: you could do something extremely convoluted using lavfi's amerge+aeval filters :P [00:00:00 CEST] --- Wed Apr 15 2015

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ffmpeg.log.20150414
by burek 15 Apr '15

15 Apr '15

[00:00:55 CEST] <gallardo> c_14: how about dirac? [00:01:13 CEST] <gallardo> I tried lossless h264 but colors differ a bit [00:01:33 CEST] <c_14> What pixel format did you use? [00:01:44 CEST] <gallardo> idk lol [00:02:01 CEST] <c_14> Maybe try with yuv444p [00:02:03 CEST] <gallardo> yuv420p [00:04:12 CEST] <gallardo> c_14: use for the source or for the encoding [00:04:20 CEST] <c_14> for the encoding [00:05:15 CEST] <gallardo> c_14, like this?: ffmpeg -i source.y4m -pixel_format yuv444p -c:v libx264 -preset veryslow -qp 0 -y test.mkv [00:06:12 CEST] <c_14> -pix_fmt, but other than that looks ok. Though if you want a fast lossless encoding, you might not want -preset veryslow [00:09:08 CEST] <gallardo> yes I was just testing lol [00:09:21 CEST] <gallardo> I see that my problem comes from the source [00:09:38 CEST] <gallardo> I mean, from the vncrec -movie outpet [00:09:42 CEST] <gallardo> output* [00:31:49 CEST] <Airwave> Hi. I'm trying to decode a g2m file using ffmpeg. It's shown as "Stream #0:2(nob): Video: none (G2M5 / 0x354D3247), 2560x1440, 229 kb/s, 1k tbr, 1k tbn, 1k tbc". From what I see, ffmpeg should have support for G2M5, but it still says "No decoder for stream #0:2, filtering impossible". [00:33:29 CEST] <klaxa> !pastebin Airwave [00:33:35 CEST] <klaxa> err [00:33:47 CEST] <Airwave> Sure, just a sec. [00:34:26 CEST] <Airwave> fflogger: Well, it's just one line: [00:34:40 CEST] <Airwave> ffmpeg -i foo.g2m bar.mkv [00:34:40 CEST] <klaxa> the complete output is hardly just one line [00:34:48 CEST] <Airwave> Ah, you want the output too of course. [00:34:52 CEST] <Airwave> Sec. [00:35:20 CEST] <jxself> It is helpful to know details, such as the FFmpeg version and configuration. This is why the complete output is desired. ;) [00:36:22 CEST] <Airwave> I understand. [00:36:32 CEST] <Prelude2004c> hey guys.. quick question.. i have a transport stream with 2 audios ( audio 1 = 5.1 and audio 2 = descriptive video ) .. i have tried in ffmpeg to do -c:a:1 and -c:a:2 and on the -c:a:1 i would output one to stereo 2 channel and another to 5.1.. how does one do that ? [00:36:40 CEST] <Prelude2004c> some reason its not really happening [00:36:48 CEST] <Airwave> klaxa: http://fpaste.org/210523/ [00:38:10 CEST] <jxself> My first thought: the 2.4 series may be too old. [00:38:22 CEST] Action: DragonsLord is away: AWAY [00:38:42 CEST] <DragonsLord> Good night & sweet dreams :) [00:39:38 CEST] <klaxa> Airwave: can you run ffmpeg -codecs and see if g2m is there as decodable? [00:39:51 CEST] <klaxa> i ran ffmpeg -codecs | grep -i g2m [00:40:32 CEST] <Airwave> klaxa: Whoops, I ran it that time with the distro version instead of the locally built newest version. [00:40:44 CEST] <Airwave> Just a sec and I'll run it again, and also give you the codec info. [00:42:44 CEST] <Prelude2004c> http://pastebin.com/gy8nWpVV [00:42:54 CEST] <Prelude2004c> can someone have a look at this and give me a pointer on what is wrong with it [00:42:55 CEST] <Airwave> klaxa: http://fpaste.org/210525/96496514/ [00:43:10 CEST] <Airwave> Sorry about ffmpeg's build info spam. Not sure why it's giving so much duplicate info. [00:43:38 CEST] <Airwave> klaxa: D.V.L. g2m Go2Meeting [00:43:46 CEST] <klaxa> hmm weird [00:44:12 CEST] <klaxa> maybe try: ffmpeg -c:v g2m -i foo.g2m bar.mkv [00:45:57 CEST] <Airwave> klaxa: I tried that earlier. I gives a lot of the following, and produces an audio-only file: [00:46:01 CEST] <Airwave> [g2m @ 0xc243c0] Wrong magic 47324D35 [00:46:04 CEST] <Airwave> Error while decoding stream #0:2: Invalid data found when processing input [00:47:45 CEST] <klaxa> either this specific codec is not supported (yet) or your file is corrupt [00:49:35 CEST] <Airwave> klaxa: It was supposed to be added in https://ffmpeg.org/pipermail/ffmpeg-cvslog/2014-October/082499.html, apparently. [00:52:36 CEST] <klaxa> oh, if you can, file a bugreport and upload a sample [00:53:10 CEST] <Airwave> I'm afraid I can't distribute this file. [00:54:01 CEST] <klaxa> hmm... that's a problem, heh [00:54:20 CEST] <jxself> It becomes difficult to investigate things. [00:55:52 CEST] <Airwave> Yeah. Sorry. [01:14:56 CEST] <koz_> What audio format is less lossy - MP3 or AIFF? I'm asking here because you guys are likely to know. [01:18:03 CEST] <koz_> I've started this collection of resources: https://notabug.org/koz.ross/awesome-gamedev I welcome contributions of everything! [01:20:22 CEST] <koz_> Whoops, wrong channel. [01:36:19 CEST] <lordkrondor> well AIFF is uncompressed, so there you go [01:44:12 CEST] <gallardo> I just can't understand how come libx264 can encode losslessly SO FAST compared to lossless dirac or lossless vp8/9. [01:44:23 CEST] <gallardo> Is there a reason for thatL [01:44:24 CEST] <gallardo> ? [04:13:56 CEST] <xxzz> gallardo: libx265 is much better [04:14:24 CEST] <gallardo> xxzz: how about licences? [04:18:37 CEST] <xxzz> gallardo: its available in pkgsrc so I suppose it can be used [04:18:57 CEST] <gallardo> xxzz: what do you think of vp9? [04:19:24 CEST] <xxzz> I can convert libx265 to mp4 in Linux but can't do that in OpenBSD [04:19:42 CEST] <xxzz> can play HEVC with ffplay on OpenBSD [04:20:04 CEST] <xxzz> have not tried VP9 google item [04:20:41 CEST] <xxzz> openbsd ffmpeg is not built against libx265 [04:21:19 CEST] <xxzz> it was there in freebsd but frogs killed it [04:25:22 CEST] <gallardo> uh [09:39:15 CEST] <keren> I have a few videos possible mov and mp4, and I want to reduce the size of the files to the maximum, I am ok comprimising the quality to upto 80%, I could go all the way to 144px [09:39:26 CEST] <keren> what would be the best output format ? [09:40:07 CEST] <keren> Like a 800 mb video could go all the way to 80mb [10:23:01 CEST] <capradma_> Hi! Since libavformat has a Jackmp depency, I'm not able to deploy release version of my software on macos [10:29:11 CEST] <BtbN> Compile it without jackmp? [12:30:43 CEST] <capradmar> BtbN: How? [12:45:41 CEST] <capradmar> I wonder why no mention of Jackmp exists in ffmpeg source code: https://github.com/FFmpeg/FFmpeg/search?utf8=%E2%9C%93&q=jackmp [12:49:45 CEST] <c_14> capradmar: afaik jack is only an optional indev [12:50:38 CEST] <c_14> https://ffmpeg.org/ffmpeg-devices.html#jack [13:17:41 CEST] <capradmar> c_14: Why is it by default when installed with brew? [13:18:08 CEST] <c_14> Ask whoever manages the brew file. [13:20:47 CEST] <capradmar> I created a PR: https://github.com/Homebrew/homebrew/pull/38633 [13:21:17 CEST] <capradmar> compiling without jack seems to solve the problem [13:28:00 CEST] <JD___> Hi. I'm having some problems with some files that are denied by a quality control software called Baton QC. This is the error: "ref_idx_l0 value was out of range. Found 5. Allowed 0 to 3 Inclusively". [13:28:52 CEST] <JD___> I'm using a Git build from February and the libx264 encoder. Does anyone know how to limit the ref_idx values to within the allowed ranges? [14:28:42 CEST] <Anoia> JD___: from what I can see, the allowed range in the spec is at least 0 to 15 [14:31:00 CEST] <Anoia> (note that the size of the ref_idx_l0 array is 0 to 3) [14:32:02 CEST] <Anoia> the actual limit of the values in it are defined by oither fields int he header [16:04:00 CEST] <JD___> Anola: This target for the files is for hardware decoding equipment which might have limitations beyond the H.264 specification. [16:06:30 CEST] <JD___> Are you saying that if I'm going to limit the range to 0->3, there are other header fields that needs to be adjusted? Which are those? [16:15:23 CEST] <aleb> Hi, if I specify -vcodec libx264 -preset veryfast, where is the "veryfast" preset defined? [16:18:27 CEST] <JEEBsv> aleb: in libx264 itself [16:18:44 CEST] <JEEBsv> libavcodec should just pass that string to libx264 [16:39:26 CEST] <Anoia> hmmm [16:39:53 CEST] <Anoia> does zeranoe usually do Windows builds for the releases? [17:26:25 CEST] <dericed> Hi all, as some of you know, MediaArea applied to http://www.preforma-project.eu/ to work on FFV1, Matroska, and LPCM. Happy to say the proposal was successful and over the next 22 months we'll be working on designing conformance checkers, implementation checkers, and metadata wranglers for these formats. Part of the project includes efforts to make progress at further standardizing FFV1 and Matroska throu [17:26:31 CEST] <dericed> gh IETF. I can post info to ffmpeg-devel soon, but wanted to post this good news to the irc first. Dave [18:42:45 CEST] <sybariten> evening [18:43:34 CEST] <sybariten> i have "mts" files that i want to use in the editing suite Sony Vegas. They are 1080i and i want to get rid of the interlacing so i have been trying to reencvode them with ffmpeg. But the result files arent popular with Vegas, which chokes. [18:43:59 CEST] <sybariten> Now, i've done some test encodes with more or less the same commadn lnie, on MJPEG files and those outputs are OK with vegas [18:44:39 CEST] <sybariten> how do i pinpoint what the difference is , that makes vegas complain? I should mention that the mts files have sound, whereas the MJPEG files do not [19:20:13 CEST] <c_14> ffprobe the input file, the output file, and the files vegas accepts [19:43:17 CEST] <sybariten> c_14: yeah i think i'm on to something now, thanks. Tried a different command line..... [20:15:58 CEST] <sybariten> can someone explain the -c: concept.... more specifically i have this line ffmpeg.exe -i ./00028.mts -y -map 0:0 -c:0 copy -map 0:1 -c:1 aac -b:1 128k -ac:1 2 -cutoff 190 00 -strict experimental -sn -s hd720 $(date +%Y%m%d-%H%M%S)_output.MP4 [20:16:57 CEST] <sybariten> i'm guessing video stream is just copied [20:19:37 CEST] <klaxa> the -c concept is that you use -c (or -codec) to specify a codec for a stream [20:19:59 CEST] <klaxa> you can also use the letters a, v, or s to specify a type rather than a stream-number [20:20:11 CEST] <klaxa> for example -c:v sets the videostream codec [20:20:17 CEST] <klaxa> a is for audio and s for subtitle [20:20:28 CEST] <__jack__> or none to use the same codec for all stream (sample: -c copy) [20:21:11 CEST] <klaxa> right, that's also possible [20:21:41 CEST] <sybariten> okay. I'm guessing -sn in my example here means, avoid any subtitles [20:22:26 CEST] <klaxa> yes [20:22:38 CEST] <sybariten> but, i also have an -s hd720 which doesnt really do anything, presumably since i already had -c:0 copy . What's the prefered way to resize this stream down to 720? Well actually i also want to deinterlace [20:23:14 CEST] <sybariten> i'm guessing i want to use yadif and scale in tandem.. but how [20:23:23 CEST] <kepstin-laptop> sybariten: if you want to change the video, then you have to reencode. [20:23:24 CEST] <__jack__> can you resize without reencode ? [20:23:34 CEST] <sybariten> kepstin-laptop: thats OK with me [20:24:10 CEST] <kepstin-laptop> sybariten: in that cause you'd use e.g. '-vf yadif,scale=hd720' then '-c:v libx264' (or something else), with appropriate encoder quality settings. [20:25:00 CEST] <sybariten> aha, and what about -vcodec ? [20:25:25 CEST] <kepstin-laptop> right now you're reencoding that to aac, dunno if that's what you want. [20:25:44 CEST] <__jack__> -vcodec: same as -c:v, more ugly (for me) [20:26:26 CEST] <kepstin-laptop> oh, for some reason i read vcodec = acodec, lol [20:27:06 CEST] <sybariten> yeah, i do not have vcodec in the lines i've pasted here, but i've seen it in other lines ive used. I was wondering how it relates to -c:v [20:27:17 CEST] <klaxa> it's the same [20:27:39 CEST] <sybariten> ok... i think i'd prefer to do this with as little shorthand as possible [20:27:42 CEST] <klaxa> using -c looks cleaner (to me) though [20:28:20 CEST] <kepstin-laptop> the non-shorthand version of "-c:v" is actually "-codec:v" :) [20:30:12 CEST] <sybariten> okay, thats cool.... and -vf becomes -filter:v ? Also, what about the -map 0:0 i had in my example.... was it there just to make things more confusing? [20:30:59 CEST] <kepstin-laptop> you can use map to select a subset of streams if your file has multiple [20:31:18 CEST] <kepstin-laptop> e.g. if there are two audio tracks but you only want one, then you'd need to use -map to select the streams to keep [20:31:31 CEST] <sybariten> ok [20:31:34 CEST] <sybariten> overkill for this [20:31:40 CEST] <kepstin-laptop> (if you don't use map, ffmpeg copies all streams, if you do use map, ffmpeg copies *only* the mapped streams) [20:31:48 CEST] <sybariten> i see [20:34:35 CEST] <sybariten> ok, so far i have ffmpeg.exe -i ./00028.mts -filter:v yadif,scale=hd720 -c:1 aac -b:1 128k -ac:1 2 -strict experi mental $(date +%Y%m%d-%H%M%S)_output.MP4 but i'm guessing i need to do seomthign about the -c -b and -a settings there [20:38:14 CEST] <kepstin-laptop> sybariten: since you're encoding to mp4, ffmpeg selects some default video codec and options for you, which are h264 (libx264) and a moderate -crf setting. It should just work like that, although you might want to tweak the settings more. [20:39:59 CEST] <sybariten> yeah, hm, it lookied horriblyl broken [20:40:01 CEST] <sybariten> in vegas [20:40:44 CEST] <sybariten> gah... sigh... it actually looked pretty normal with VLC ... so the strange artefacts comes from vegas... [20:40:55 CEST] <sybariten> this video project is killing me... oh well [20:41:20 CEST] <kepstin-laptop> did you accidentally get a build with a 10bit x264? that could cause fun issues. [20:41:41 CEST] <sybariten> hm, dunno, can i try some simpler codec perhaps ? [20:41:53 CEST] <kepstin-laptop> sybariten: pastebin your ffmpeg command output somewhere please. [20:43:38 CEST] <sybariten> OK, this is what i used just now recently. http://pastebin.com/Ugm6eB9S as i said, the video opens in vegas, but it looks much worse there than in a normal video player. At least in its preview window. The video's all "jagged"... not the normal compression artefacts im used to seeing [20:45:24 CEST] <kepstin-laptop> that's just the command line, I want to see the output from when it runs. [20:45:28 CEST] <sybariten> oh [20:47:34 CEST] <sybariten> http://pastebin.com/6YrCXXyZ [20:55:13 CEST] <kepstin-laptop> hmm. that looks all about like I'd expect; the output video should be fine in most players [20:55:20 CEST] <sybariten> I see, ok [20:55:53 CEST] <sybariten> I am following a thread now on videoforum, about these issues (mts/avhcd video, NLE, ffmpeg) and will maybe pick up some tip there [20:56:23 CEST] <sybariten> as it turns out, the avchd format really isnt edited very often [21:03:30 CEST] <JD___> sybariten: Your goal is to deinterlace the 1080i master video file and import the result into Sony Vegas for editing? [21:04:38 CEST] <sybariten> JD___: exactly [21:06:06 CEST] <JD___> I'm not familiar with Vegas import formats, but I would personally have tried a mpeg ts with audio copied and mjpeg image codec at original dimensions [21:07:19 CEST] <sybariten> i am actually trying out some mpeg (2?) solution as we speak [21:07:33 CEST] <sybariten> ffmpeg.exe -i ./00028.MTS -filter:v yadif,scale=hd720 -vcodec mpeg2video -r 29.97 -qscale 4 -qm in 1 -intra -flags +ilme+ildct -top 1 -acodec ac3 -ab 384k -f mpegts test2.m2ts [21:07:59 CEST] <JD___> ffmpeg -i 1080i.mts -vf yadif -c:a copy -c:v mjpeg 1080p.ts [21:08:36 CEST] <JD___> Your command does a lot of work and going to cost you a lot of quality [21:09:03 CEST] <sybariten> aha? [21:09:11 CEST] <sybariten> OK i'll try yours [21:09:26 CEST] <JD___> Try mine and see if Vegas imports it. File size will be huge, but it should be fast and keep quality acceptable [21:09:48 CEST] <sybariten> one thing though.... i think that the bleeding file actually isnt 1920, really ... its a 1440 file with some PAR fuckery [21:10:10 CEST] <sybariten> so i dunno if i'm gonna need an -aspect at some point. But i'll try [21:10:29 CEST] <JD___> Shouldn't matter, you can change that when rendering in Vegas [21:11:44 CEST] <sybariten> ok... [21:12:39 CEST] <sybariten> yeah i dunno why the -r 29.97 option was suggested really... i feel that i wanna mess as little as possible with frame rate [21:13:23 CEST] <sybariten> this is what ffmpeg -i says in my case. Stream #0:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), yuv420p, 1440x1080 [SAR 4:3 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc [21:13:53 CEST] <DragonsLord> Good evening! [21:14:38 CEST] Action: DragonsLord is back (gone 20:36:16) [21:17:15 CEST] <JD___> sybariten: did my command work? Is it encoding? [21:23:58 CEST] <sybariten> nope [21:24:25 CEST] <sybariten> vegas appearently didnt like mjpeg... ? strange, ive used that before. But not what i got from your command [21:25:11 CEST] <sybariten> i just did this an mpeg2video one though, and rendered the vegas project out again... the outcome isnt stellar but quite OK [21:43:21 CEST] <aleb> If I specify "-vcodec libx264 -preset veryslow", what bitrate will be used? How about if I specify "veryfast"? I see x264 does not include a bitrate value for these.. http://git.videolan.org/?p=x264.git;a=blob;f=common/common.c;h=de5d441b9079… [21:43:52 CEST] <aleb> JEEBsv, btw, thanks for your answer earlier :) [21:44:46 CEST] <JEEBsv> aleb: libx264 defaults to its default - which is CRF 23 [21:44:56 CEST] <JEEBsv> that is not a specific bit rate but rather a rate factor [21:55:29 CEST] <JD___> sybariten: Sorry about that. One last hail Mary is that Vegas uses VFW for decoding and that by installing something like ffdshow, you'll be able to import mjpeg and a lot more into Vegas. [22:05:10 CEST] <sybariten> JD___: yeah... you know what i discovered though? (this is kinda embarrassing) [22:06:09 CEST] <sybariten> removing interlacing was just my own instinct from seeing the material with VLC, and it looked like shite with those jagged edges... turns out that if using the source files in vegas and just rendering to 1280x720p, it looks quite alright actually [22:06:45 CEST] <sybariten> seems like my machine handles "HD" files in editing too so i might not need to rescale them beforehand [22:16:43 CEST] <JD___> sybariten: user error = best error [22:16:51 CEST] <sybariten> yeah [22:17:07 CEST] <JD___> Easily corrected [23:26:01 CEST] <DragonsLord> Good night, sleep fine! [23:28:26 CEST] Action: DragonsLord is away: AWAY [23:50:53 CEST] <kyleogrg> Hey. I am converting 29.97i VOB files to 23.976p x264. The VOBs were originally film, so I am doing detelecine. However, as it encodes, I am getting occassional messages saying "Frame X at X is still interlaced." [23:51:38 CEST] <kyleogrg> Is there a best way to detelecine? I am doing fieldmatch and decimate. [23:53:09 CEST] <c_14> Try adding a yadif between the deteceline and the decimate [23:53:44 CEST] <lordkrondor> i've found that if you put the yadif between the fieldmatch & decimate, the video is faster than the audio. I usually have put the yadif after decimate. [23:53:58 CEST] <lordkrondor> or done a pullup,idet,yadif combo [23:54:14 CEST] <kyleogrg> what's idet? [23:54:24 CEST] <lordkrondor> interlacing detection [23:54:53 CEST] <kyleogrg> so what usually delivers the best results? [23:54:57 CEST] <kyleogrg> for you [23:55:30 CEST] <lordkrondor> i've used both, some SD stuff worked better with pullup, the HD stuff i've done looked better with fieldmatch [23:55:37 CEST] <lordkrondor> i'd just play around and see how it works [23:55:49 CEST] <kyleogrg> sure ok [23:55:51 CEST] <kyleogrg> thanks [00:00:00 CEST] --- Wed Apr 15 2015

1 0

ffmpeg-devel.log.20150413
by burek 14 Apr '15

14 Apr '15

[00:08:18 CEST] <BBB> it is [00:08:20 CEST] <BBB> but less [00:08:46 CEST] <BBB> x86-64 allows you to use rip as a register for addressing, whereas on x86-32, you needed to load the ip in an explicit register per function [00:08:56 CEST] <BBB> however, it still means you need the register in addresses [00:09:23 CEST] <BBB> so if you have an expression [base+4*offset], rip doesnt fit in [00:09:28 CEST] <BBB> and you still need an extra instruction [00:09:42 CEST] <BBB> (re: to iive, rcombs and daemon404) [00:13:58 CEST] <iive> isn't the whole point that the base is relative to the rip? and the linker calculates that? [00:16:57 CEST] <iive> it seems like not. [00:22:05 CEST] <BBB> when pic is enabled, yasm changes [base] into [base+rip] [00:22:30 CEST] <BBB> but x86 (and that includes x86-64) allows only a base and an offset as registers in address statements like that [00:22:54 CEST] <BBB> so if your statement has an offset as register, then it doesnt work [01:26:20 CEST] <Timothy_Gu> BBB: the base -> base+rip transformation is `default rel` right? [01:37:41 CEST] <jamrial> release candidate of gcc 5 is out [01:43:20 CEST] <Timothy_Gu> yay? [01:44:07 CEST] <cone-235> ffmpeg 03Himangi Saraogi 07master:0b72a47f5592: rtsp: Check a malloc return value [01:44:08 CEST] <cone-235> ffmpeg 03Michael Niedermayer 07master:4dd0c0712206: Merge commit '0b72a47f5592ecc035edd5bac57558ad9604612d' [01:52:44 CEST] <cone-235> ffmpeg 03Himangi Saraogi 07master:4e5443216445: rtsp: Fix unchecked return value [01:52:45 CEST] <cone-235> ffmpeg 03Michael Niedermayer 07master:818e8897094d: Merge commit '4e5443216445f3a9b8e6cb3fa4f448762e19006e' [01:53:10 CEST] <BBB> Timothy_Gu: I believe so, but Im not 10000% sure [04:11:53 CEST] <cone-235> ffmpeg 03Mark Reid 07master:c919cc61a155: tests/fate: added mxf opatom audio regression tests [04:36:21 CEST] <cone-235> ffmpeg 03Rostislav Pehlivanov 07master:f7f71b5795d7: aacenc: Add support for Perceptual Noise Substitution energy values [07:12:50 CEST] <rcombs> fun fact, ff_all_formats lies when it says "all" [11:32:50 CEST] <cone-632> ffmpeg 03Rodger Combs 07master:a4d58c97d051: lavf/http: handle case where the server returns a redirect during a seek [11:53:07 CEST] <cone-632> ffmpeg 03James Almer 07master:30ce065ffec3: fate: add tta encoder test [12:05:46 CEST] <cone-632> ffmpeg 03Rodger Combs 07master:c69ff12dc3b4: lavf/mpeg: vobsub add an option to specify the .sub's URI [12:26:14 CEST] <wm4> "- 1 x tablecloth: 81,61 ¬" wat [12:26:36 CEST] <BtbN> huh? [12:26:45 CEST] <wm4> must be some sort of tablecloth with a gold printed ffmpeg logo [12:26:52 CEST] <BtbN> Or a hell lot of tables. [12:27:16 CEST] <BtbN> If it was actualy printed with ffmpeg logo, the price doesn't seem too far off though. [13:00:45 CEST] <wm4> spotted an apng in the wild, and ffmpeg is being useless: "apng: Blending with pixel format rgb24 is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented." [13:55:34 CEST] <kierank> so it looks like the web server will be the highest scoring proposal [13:55:36 CEST] <kierank> :( [14:03:16 CEST] <cone-632> ffmpeg 03Michael Niedermayer 07master:d5dcd94630ca: avfilter/vf_scale: Add warning for AVCOL_SPC_YCGCO [14:03:17 CEST] <cone-632> ffmpeg 03wm4 07master:f4f3065c69ce: h264: remove bogus YCgCo warning message [14:04:55 CEST] <kierank> lglinskih: how's it going? [15:19:50 CEST] <kos2dk1> Looking someone for custom FFmpeg Modification, will pay. Contact me with PM [15:22:19 CEST] <kierank> kos2dk1: very specific [15:22:25 CEST] <kierank> you need to go into more detail [15:36:32 CEST] <lglinskih> kierank: fine! dealing with the comments from ML [15:36:48 CEST] <kierank> ok [15:36:56 CEST] <kierank> please let me know if oyu have problems [15:38:55 CEST] <lglinskih> kierank: ok, thanks ^_^ [18:31:50 CEST] <loki_> can somebody explain me, if need set optimization options in ffmpeg configure stage? [18:32:00 CEST] <loki_> or all we be configured best way without me? [18:32:07 CEST] <loki_> for example -march=native [18:32:11 CEST] <loki_> or -Ofast [18:51:51 CEST] <wm4> loki_: no, the configure script adds more than enough rice by default [18:52:22 CEST] <loki_> ok... because when i messed with optimization i recieved even worse result lol [19:05:04 CEST] <jamrial> wm4: not true, configure forces -fno-tree-vectorize [19:05:08 CEST] <jamrial> so that's a lot of rice lost [19:06:26 CEST] <wm4> and a lot of codegen bugs? [19:07:36 CEST] <ramiro> jamrial, gcc's tree vectorize sucks [19:08:37 CEST] <ramiro> I remember massive slowdowns due to that (in 2010, but still =) [19:10:17 CEST] <jamrial> well, gcc has come a long way since then [19:10:25 CEST] <jamrial> i think it was disabled back in the gcc 4.3 days [20:03:57 CEST] <t_than> Hello, all. [20:04:31 CEST] <t_than> I am a student who has applied for Google Summer of Code at FFMpeg. Does anyone know if I can contact someone for some basic information? [20:04:43 CEST] <t_than> Just to clarify some things [20:16:58 CEST] <kierank> t_than: what would you like to know [20:19:34 CEST] <cone-632> ffmpeg 03Vittorio Giovara 07master:73dacabfc9b9: ffv1: Check memory allocations [20:19:35 CEST] <cone-632> ffmpeg 03Michael Niedermayer 07master:fb6dd77000f1: Merge commit '73dacabfc9b9ef1fd2c08105fdab6238ee29c2fc' [20:19:36 CEST] <cone-632> ffmpeg 03Michael Niedermayer 07master:a3963cc8ec3b: avcodec/ffv1: use av_freep() [20:19:37 CEST] <cone-632> ffmpeg 03Michael Niedermayer 07master:d19fc69f15c3: avcodec/ffv1: remove unneeded variable [20:19:38 CEST] <cone-632> ffmpeg 03Michael Niedermayer 07master:4d74c8d55035: avcodec/ffv1: dont leave stale pointers in memory [20:21:08 CEST] <t_than> hi! I was a late applicant, and I've seen that a lot of the projects/qualification tasks have been assigned to different students. I'd like to know if it's too late to start working on some qualification tasks if there are any vacancies. [20:22:49 CEST] <t_than> You see, I got an email from another project today showing interest on my proposal, but truth be told my main interests are on video coding and I'd really prefer to be chosen by FFMpeg for GSoC. I wouldn't like my late application to seem like I wasn't interested enough. [20:23:34 CEST] <Compn> t_than : we dont care about applications or dates, only qualifying task and the quality of your code and ability to work with us :P [20:24:02 CEST] <Compn> but, i guess as you havent started on a task, its hard for us to see your work... [20:24:16 CEST] <Compn> i dont know if its too late, wait around for someone more involed in gsoc [20:24:18 CEST] <Compn> (not me) [20:24:40 CEST] <Compn> and do not pick apng task lol [20:26:23 CEST] <t_than> i think I've seen that on my ffmpeg-devel email subscription... :) [20:26:38 CEST] <philipl> If you filed a GSoC application in time, then you're still eligible. But you need to finish a qualification task by the 27th, when final selections are mode. [20:26:42 CEST] <philipl> So move fast. [20:27:15 CEST] <michaelni> 27 is too late [20:27:20 CEST] <philipl> is it? [20:28:01 CEST] <t_than> but I should start on some qualification task that someone else hasn't picked yet, right? Or else, wouldn't there be a conflict? [20:28:21 CEST] <michaelni> 24 April 07:00 UTC Student acceptance choice deadline [20:28:42 CEST] <michaelni> that means if someone posts apatch on the 24th thats too late [20:28:47 CEST] <philipl> Ah. It was wrong in my mind. [20:29:03 CEST] <philipl> I hope I haven't been saying 27th to people in the past :-P [20:30:20 CEST] <michaelni> 27 April: 19:00 UTC Accepted student proposals announced on the Google Summer of Code 2015 site. [20:34:07 CEST] <wm4> <michaelni> 24 April 07:00 UTC Student acceptance choice deadline <- how nice, they're accounting for students working hard through the night to make the goal [20:35:50 CEST] <BtbN> Isn't that just midnights at the google HQ? [20:36:09 CEST] <wm4> that's a boring interpretation [21:29:50 CEST] <kierank> Anyone know what happened to the FFmpeg dirac encoder gsoc [21:30:09 CEST] <kierank> I seem to recall the student getting fed up with cosmetic reviews [21:30:18 CEST] <pengvado> Timothy_Gu: (if you haven't already got your answer about PIC) [21:30:19 CEST] <pengvado> GOT is only needed when library A calls a function that's implemented in library B. [21:30:22 CEST] <pengvado> x86inc.asm doesn't implement GOT, because I have never used asm for that case. [21:30:25 CEST] <pengvado> it is not needed for references within one library, since references within one library can be resolved at compile-time and thus don't leave any relocation at all in the compiled library. [21:30:40 CEST] <Compn> kierank : theres a page that lists all of the gsoc projects and what happened to them [21:30:42 CEST] <Compn> on the wiki [21:32:31 CEST] <Compn> kierank : http://wiki.multimedia.cx/index.php?title=FFmpeg_/_Libav_Summer_Of_Code#Dir… [21:32:38 CEST] <Compn> Code Status: Decoder committed to FFmpeg, the encoder still needs more work. [21:33:20 CEST] <kierank> Thanks [21:33:33 CEST] <kierank> I might rip the encoder out of ffmpeg [21:34:05 CEST] <kierank> I think there was a second gsoc as well actually [21:35:46 CEST] <Compn> people using dirac now ? :P [21:35:49 CEST] <Compn> besides bbc [21:35:58 CEST] <Compn> didnt bbc also abandon it [21:50:32 CEST] <kierank> It is not abandoned any more [21:52:54 CEST] <j-b> kierank: seriously? [21:53:14 CEST] <kierank> Yes [21:53:36 CEST] <j-b> kierank: they should use VLC to decode and encode :) [21:53:41 CEST] <kierank> http://www.4rfv.com/VKIXRRF1HR2A/snell-and-quantel-demonstrate-realworld-ip… [21:53:49 CEST] <j-b> kierank: can you ask them to clarify their ogg/dirac mapping? [21:54:01 CEST] <kierank> Yes but they are using a simple profile and encoding at a latency of a few lines [21:54:17 CEST] <kierank> I can assure you zero people give a shit about ogg [21:55:13 CEST] <kierank> If they don't use vc-2 they use tico or they use Sony low latency codec which are all patented [21:55:14 CEST] <jamrial> xiph people do, i assume :p [21:56:03 CEST] <jamrial> oh, you meant at bbc [21:56:56 CEST] <kierank> Yes [22:42:04 CEST] <kierank> Compn: any idea where the code is kept? [23:01:59 CEST] <lglinskih> does decoder _have_ to respect the request_channel_layout context field? [23:02:32 CEST] <lglinskih> e.g. FLAC ignores it completely [23:03:28 CEST] <kierank> lglinskih: it doesn't afaik [23:15:16 CEST] <Compn> kierank : there was a git gsoc repo long ago... [23:15:24 CEST] <Compn> lets see [23:17:21 CEST] <Compn> no, no idea [23:17:23 CEST] <Compn> ask luca ? [23:17:38 CEST] <Compn> or maybe michaelni has a copy of it [23:25:47 CEST] <Compn> kierank : you are looking for dirac encoder gsoc code, right ? just to make it easier on whomever has it and reading this. [23:25:54 CEST] <kierank> yes [23:34:47 CEST] <Compn> http://lists.mplayerhq.hu/pipermail/ffmpeg-soc/2007-August/thread.html [23:36:41 CEST] <Compn> it was a svn rpeo [23:36:42 CEST] <Compn> repo [23:43:17 CEST] <Compn> kierank : might be in here > svn://svn.mplayerhq.hu/soc [23:43:38 CEST] <Compn> i make no promises :P [00:00:00 CEST] --- Tue Apr 14 2015

1 0

ffmpeg.log.20150413
by burek 14 Apr '15

14 Apr '15

[01:40:13 CEST] <elgatov> hello guys, i've ripped a couple of blurays and they have quite a big size, i was wondering if removing some audio streams would help, and if i should remove DTS bearing in mind i only have stereo [01:41:25 CEST] <elgatov> i've tried using mediainfo to check the size of each stream but i cant find it [01:42:09 CEST] <c_14> It'll "help", but the biggest part is the video stream. [01:45:45 CEST] <elgatov> c_14, so they aren't that big, thanks [01:46:15 CEST] <elgatov> and for the other part of DTS in stereo, is it multiplexed by the drivers already? [01:46:26 CEST] <elgatov> im quite a noob with multimedia [01:47:03 CEST] <c_14> Most video players should downmix automatically. [01:47:27 CEST] <elgatov> ah it's called downmix, great [01:47:29 CEST] <elgatov> thank you [01:50:59 CEST] <chungy> DTS is lossless iirc, at least re-encoding that to a lossy format can help some [01:51:28 CEST] <c_14> Only DTS-HD MA is lossless [01:53:08 CEST] <chungy> i should get a blu-ray drive at some point, i have Star Trek TOS on the format and it looks great on the TV :P [02:03:28 CEST] <c_14> elgatov: a 1:28:13 DTS HD-MA track is about 2.5GiB [02:03:33 CEST] <c_14> So you can get a feel for the size. [05:19:53 CEST] <ac_slater> hey all. With `libavformat`, how can I "prefill" some stream info so av_get_stream_info() wont fail? I know all of the encoding parameters, rates, etc. [07:32:40 CEST] <Zerowalker> Is libopus limited to 256kbps? [07:44:01 CEST] <BtbN> Propably depends on the profile, or whatever it's called [07:44:06 CEST] <BtbN> but why would you want more? [07:44:27 CEST] <BtbN> opus is even better than aac, which is already "perfect" at around 128~196kbps [07:52:11 CEST] <Zerowalker> BtbN, profile? well i use it for later transcode with youtube, and as space isn't much of an enough i rather use 320kbps [07:52:26 CEST] <Zerowalker> much of an issue* [07:52:50 CEST] <BtbN> Well, if you don't care for space, don't transcode at all? [07:53:30 CEST] <Zerowalker> There is a limit to everything, have thought of using wave of flac, but that's a bit overkill [07:53:51 CEST] <Zerowalker> i just want to be able to fully use opus, i don't see why ffmpeg would limit it [07:54:43 CEST] <BtbN> opus is extremely efficient though [07:54:52 CEST] <BtbN> i'd expect it to be close to lossless at 256kbit/s [07:55:13 CEST] <Zerowalker> indeed that's why i use it, and yeah but it just feels better to have that extra "overhead" if you will [07:55:43 CEST] <Zerowalker> perhaps i need to contact an ffmpeg developer about it, is 256k an universal limit for a:b? [07:55:44 CEST] <BtbN> max bitrate for opus seems to be 512kbps [07:55:58 CEST] <Zerowalker> yes, but i can't get it to work with ffmpeg [07:57:46 CEST] <Zerowalker> "The bit rate 320000 bps is unsupported. Please choose a value between 500 and 256000" [08:01:12 CEST] <BtbN> The bitrate isn't directly given. It's computed from some other values. So it depends on the settings which bitrate is max. [08:02:10 CEST] <BtbN> But why don't you just use vbr with max quality setting? [08:02:52 CEST] <Zerowalker> BtbN, does Opus even have that, was sure it only had bitrate indicators, not quality [08:03:05 CEST] <Zerowalker> what do you mean the bitrate isn't directly given though [08:03:09 CEST] <BtbN> It has a vbr option, that accepts the values on/off/constrained [08:03:34 CEST] <BtbN> The quality/compression is given via compression_level, from 0 to 10 [08:03:36 CEST] <Zerowalker> that simply defines, "variable bitrate" "constant bitrate" "contrained variable bitrate" [08:03:44 CEST] <Zerowalker> yeah and default for that is 10 [08:04:05 CEST] <BtbN> Yes, and generaly vbr is better for quality than cbr. [08:04:31 CEST] <Zerowalker> isn't vbr default as well? [08:04:48 CEST] <BtbN> yes. [08:05:00 CEST] <BtbN> The maximum bitrate seems to be controled by the cutoff option. [08:05:03 CEST] <Zerowalker> so higher bitrate = higher playground [08:05:43 CEST] <BtbN> http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/libopusenc.c;h=78… [08:05:44 CEST] <Zerowalker> but still the bitrate limit decides how much of an average bitrate it has to limit itself to (if it need that much) [08:06:31 CEST] <Zerowalker> if (avctx->bit_rate < 500 || avctx->bit_rate > 256000 * avctx->channels) [08:07:15 CEST] <BtbN> Well, won't be there for fun. [08:07:20 CEST] <Zerowalker> well will make a ticket anyhow, as it's a defect, even if many aren't bothered by it you are prevented from utilizing opus fully [08:07:23 CEST] <BtbN> Maybe 256kbps is the max per channel? [08:07:37 CEST] <Zerowalker> then stereo = 512k [08:07:44 CEST] <Zerowalker> hmm [08:07:51 CEST] <Zerowalker> wait that might be it [08:08:44 CEST] <Zerowalker> BtbN, yeah that was it. I was blind, didn't notice that the file i had been testing on was Mono -_- [08:08:49 CEST] <Zerowalker> thank god i didn't make the ticket;P [12:46:31 CEST] <chungy> Is there any real reason -to is not available as an input option? [13:24:25 CEST] <justinX> "-to time_stop record or transcode stop time" ? [13:25:52 CEST] <justinX> chungy: yeah that looks like it is supposed to be per input file to me.... [13:29:31 CEST] <c_14> chungy: nobody implemented it is the most likely reason [13:37:09 CEST] <chungy> heh yeah. Just seemed silly that -t can be used for input but not -to [15:19:52 CEST] <kos2dk1> Looking someone for custom FFmpeg Modification, will pay. Contact me with PM [15:26:40 CEST] <justinX> kos2dk1: you *should* probably tell something about *what* custom Modifications you want to make [16:34:02 CEST] <d3fault_> how come when i join a video file recorded with raspivid together with an audio file, they get all out of sync randomly and inconsistently (sometimes it catches back up for a bit)? [16:57:40 CEST] <Russ_> I have two videos 1.mp4 and 2.mp4 how to add them using ffmpeg [16:58:59 CEST] <kepstin-laptop> by "add", you mean concatenate so one plays after the other? [17:00:00 CEST] <kepstin-laptop> Russ_: https://www.ffmpeg.org/faq.html#How-can-I-concatenate-video-files_003f [17:00:29 CEST] <kepstin-laptop> if your videos are the same codec, resolution, etc., you might be able to use the concat demuxer; otherwise you'll want to use the concat filter and re-encode. [17:00:35 CEST] <Russ_> kepstin-laptop_ I did that and even file generated but that file imean video is not playing [17:00:46 CEST] <Russ_> ok [17:00:51 CEST] <Russ_> let me try again :) [17:06:34 CEST] <Russ_> it's just showing first video but not second [17:08:44 CEST] <Russ_> k [17:10:30 CEST] <loki_> can somebody explain me one thing... if i build ffmpeg by my own, can i use optimization or all optimizations will be done automaticly? [17:10:57 CEST] <loki_> for example --optflags or something [17:11:34 CEST] <Russ_> it's done thanks guys [17:12:39 CEST] <pron> mm if i have .ts file , wich starts with garbage till the first keyframe ( if i understand correctly ) whats the correct way to remove that grabage ? [19:43:48 CEST] <QuakePhil9000> Can someone please help me with this error? http://pastebin.com/Q7XTZxL1 [19:45:57 CEST] <QuakePhil9000> If I convert them to pngs, it works just fine [19:48:17 CEST] <filo1234> hi guys [19:49:31 CEST] <filo1234> I have a video.mkv that contains two audio trace, Italian and English, so how can I convert it only with an audio trace? [19:56:40 CEST] <filo1234> I have to use -vlang code? [19:57:37 CEST] <kepstin-laptop> filo1234: you'd use the -map option to select tracks, e.g. "-map 0:v:0 -map 0:a:0" to select the first audio and first video track, then use "-c copy" to avoid reencoding. [19:58:57 CEST] <filo1234> kepstin-laptop: ok but how can I see where first trace is mapped? [19:59:16 CEST] <filo1234> kepstin-laptop: I'll paste you some ffmpeg output [19:59:38 CEST] <kepstin-laptop> filo1234: the output of ffmpeg says which tracks are which if you just run "ffmpeg -i inputfile" [19:59:46 CEST] <kepstin-laptop> or "ffprobe inputfile" [20:02:29 CEST] <capradmar> Hi! I wonder why there is a libavformat dependency with Jackmp in the ffmpeg latest version (2.6.1) [20:03:04 CEST] <filo1234> kepstin-laptop: http://paste.ubuntu.com/10817443/ [20:04:37 CEST] <kepstin-laptop> filo1234: ok, then you'd want to use either "-map 0:0 -map 0:1" for italian audio or "-map 0:0 -map 0:2" for english audio. [20:04:51 CEST] <kepstin-laptop> the stuff after the Stream # is the stream id that map can use [20:06:59 CEST] <filo1234> ok kepstin-laptop ok but, for my knowledge, how do you know that map 0:1 is Italian and map 0:2 is English? [20:07:15 CEST] <kepstin-laptop> because it says "ita" and "eng" beside them, which are the language codes [20:07:32 CEST] <filo1234> uhhuh wait [20:07:53 CEST] <kepstin-laptop> on the last 3 lines in your paste [20:08:00 CEST] <filo1234> OMG sorry [20:08:13 CEST] <filo1234> I see that know... :| [20:08:43 CEST] <filo1234> kepstin-laptop: thanks a lot you get my eyes opened [20:12:23 CEST] <QuakePhil9000> I guess for now I can use something gifsicle to hold ffmpeg's hand [20:13:08 CEST] <filo1234> kepstin-laptop: thans again [20:13:09 CEST] <filo1234> bye [21:30:54 CEST] Action: DragonsLord is back (gone 22:20:24) [21:31:12 CEST] <DragonsLord> Good evening to everybody! :) [21:39:36 CEST] <mtcjayne> How do I diagnose a segmentation fault which occurs when combining two filters, but not when using one or the other? [21:40:10 CEST] <mtcjayne> http://pastebin.com/x6A8nnAJ [21:40:43 CEST] <mtcjayne> I would like to get the first scale + the fade combined, but this causes a segmentation fault. [21:44:23 CEST] <mtcjayne> An updated build was available. I am no longer seeing the behavior. [22:09:41 CEST] <Airwave> Hi. I'm trying to decode a g2m file using ffmpeg. It's shown as "Stream #0:2(nob): Video: none (G2M5 / 0x354D3247), 2560x1440, 229 kb/s, 1k tbr, 1k tbn, 1k tbc". From what I see, ffmpeg should have support for G2M5, but it still says "No decoder for stream #0:2, filtering impossible". [22:49:02 CEST] <heiner> I can convert a video file to an animated GIF using: ffmpeg -i f.mp4 -t 10 -r 10 image.gif [22:49:33 CEST] <heiner> Is there also an easy way to create a time lapse GIF instead of a GIF containing all frames? [22:51:00 CEST] <heiner> I search for a way to use e.g. one frame every 20 seconds and create an animated GIF displaying each frame for one second [22:52:33 CEST] <heiner> Currently I create many image files (one per frame) and stitch them together to an GIF using ImageMagick's "convert" [22:55:59 CEST] <heiner> Should I use a different tool for this purpose? [23:01:24 CEST] <gallardo> may someone please recommend a fast encoding lossless video codec [23:01:34 CEST] <gallardo> I am trying dirac but it is so slow [23:17:26 CEST] <loki_> can somebody explain me one thing, why can ffmpeg cosume less cpu when for example i run mplayer? are there some runtime kernel changes or what? [23:18:01 CEST] <loki_> i mean less cpu and equal job done in fps. how it is possible? [23:23:09 CEST] <klaxa> it depends on what you do [23:23:25 CEST] <klaxa> codec copy will consume less cpu than having to decode frames [23:23:39 CEST] <klaxa> and displaying decoded frames also takes some cpu time [23:49:36 CEST] <c_14> gallardo: lossless H.264, ffv1 or huffyuv maybe. [00:00:00 CEST] --- Tue Apr 14 2015

1 0

ffmpeg-devel.log.20150412
by burek 13 Apr '15

13 Apr '15

[00:22:39 CEST] <jamrial> michaelni: ok, thanks [00:22:59 CEST] <cone-750> ffmpeg 03James Almer 07release/2.6:c1c245e1a347: doc: add missing x86 cpuflags to fftools documentation [00:23:00 CEST] <cone-750> ffmpeg 03James Almer 07release/2.6:6f236d37746e: doc: add missing arm cpuflags to fftools documentation [00:23:01 CEST] <cone-750> ffmpeg 03James Almer 07release/2.6:04fd0250e1fd: doc: add aarch64 cpuflags to fftools documentation [00:29:16 CEST] <cone-750> ffmpeg 03Timothy Gu 07master:a202fe00ab9d: fate: Add atrac3+ tests [00:58:29 CEST] <cone-750> ffmpeg 03James Almer 07release/2.2:57f271ce4149: doc: add missing x86 cpuflags to fftools documentation [00:58:30 CEST] <cone-750> ffmpeg 03James Almer 07release/2.2:733cb1c6c59d: doc: add aarch64 cpuflags to fftools documentation [00:58:31 CEST] <cone-750> ffmpeg 03James Almer 07release/2.4:5598d62a0703: doc: add missing x86 cpuflags to fftools documentation [00:58:32 CEST] <cone-750> ffmpeg 03James Almer 07release/2.4:4b817e2fff3a: doc: add missing arm cpuflags to fftools documentation [00:58:33 CEST] <cone-750> ffmpeg 03James Almer 07release/2.4:3a5b4afd911f: doc: add aarch64 cpuflags to fftools documentation [00:58:34 CEST] <cone-750> ffmpeg 03James Almer 07release/2.5:7e05c70bb025: doc: add missing x86 cpuflags to fftools documentation [00:58:35 CEST] <cone-750> ffmpeg 03James Almer 07release/2.5:60f2f332a397: doc: add missing arm cpuflags to fftools documentation [00:58:36 CEST] <cone-750> ffmpeg 03James Almer 07release/2.5:43fdd89a3f07: doc: add aarch64 cpuflags to fftools documentation [01:02:09 CEST] <cone-750> ffmpeg 03Timothy Gu 07n2.6.2:HEAD: fate: Add atrac3+ tests [01:46:25 CEST] <michaelni> Welcome to the FFmpeg development channel | Questions about using FFmpeg or developing with libav* libs should be asked in #ffmpeg | This channel is publicly logged | FFmpeg 2.6.2 has been released! [01:59:06 CEST] <azk> [02:11:30 CEST] <BBB> do we have a way to indicate in e.g. the mpeg12/hevc/h264 decoders to output data in coding instead of display order? [02:16:26 CEST] <Timothy_Gu> yay I setup an FATE instance for Haiku! [02:16:29 CEST] <iive> i think there was such a way. [02:16:41 CEST] <iive> if it haven't been removed. [02:17:31 CEST] <michaelni> theres SLICE_FLAG_CODED_ORDER but its only for ff_draw_horiz_band() [02:18:06 CEST] <michaelni> AVCodecContext.draw_horiz_band() rather [02:24:24 CEST] <kierank> hopefully my opw student can write a draw_horiz_band test [03:17:31 CEST] <cone-750> ffmpeg 03Mark Reid 07master:415f1fab8de7: libavformat/mxfenc: added support mxf opatom audio muxing [03:39:55 CEST] <Timothy_Gu> michaelni: http://fate.ffmpeg.org:8080/report/x86-haiku-gcc-4.8/20150412012810 [03:41:41 CEST] <michaelni> Timothy_Gu, very nice, thanks alot [03:45:09 CEST] <Timothy_Gu> michaelni: also are you gonna merge the old fateserver patch [04:00:41 CEST] <michaelni> Timothy_Gu, applied to the git repos [04:01:02 CEST] <Timothy_Gu> michaelni: thx [04:04:39 CEST] <michaelni> Timothy_Gu, should i update the checkout that is used on fate.ffmpeg.org too ? [04:12:11 CEST] <michaelni> you can update it too, ill leave it to you, dont want to somehow step onto some work there [04:24:42 CEST] Action: michaelni falls asleep [05:48:41 CEST] <Timothy_Gu> michaelni: i'll do it [05:49:04 CEST] Action: Timothy_Gu googles "time in austria" [05:54:50 CEST] <Timothy_Gu> michaelni: done. No work is really being done *there* haha. My brain would explode from all the perl. [12:14:30 CEST] <MelchiorGaspar> anybody alive in here O_o loi ;D [12:16:24 CEST] <phh> ok. [12:16:55 CEST] <MelchiorGaspar> I found a format that FFmpeg does not support though I thought it did [12:17:10 CEST] <MelchiorGaspar> .pss PLaystation 2 (Suikoden 3 and 4) [12:18:07 CEST] <MelchiorGaspar> I extracted from my PS2 DvD discs so I convert to watch (oohh the memories) soI can upload a file if a Dev wants to test it [13:49:20 CEST] <kierank> MelchiorGaspar: open a ticket and upload a sample [13:50:06 CEST] <Compn> MelchiorGaspar : we have samples here http://samples.ffmpeg.org/playstation/ [13:50:10 CEST] <Compn> and in a few other dirs [13:50:13 CEST] <Compn> but sure more samples are good [14:01:20 CEST] <kierank> woah aac encoder patches [14:17:41 CEST] <JEEBsv> finally patches?: o [14:18:41 CEST] <JEEBsv> oh, gsoc [15:31:10 CEST] <MelchiorGaspar> ok thx... I see PLaystation Folder... to reorigize formats I would suggest having a Playstation folder for each version [15:31:28 CEST] <MelchiorGaspar> - PlayStation1 [15:31:31 CEST] Last message repeated 1 time(s). [15:31:31 CEST] <MelchiorGaspar> - PlayStation2 [15:31:33 CEST] <MelchiorGaspar> - PlayStation3 [15:31:38 CEST] <MelchiorGaspar> - PlayStation4 [15:32:23 CEST] <MelchiorGaspar> where you BugTracker located? [15:33:15 CEST] <JEEBsv> http://trac.ffmpeg.org/ [15:33:33 CEST] <MelchiorGaspar> ty. [15:39:10 CEST] <MelchiorGaspar> k tracker account created [15:39:14 CEST] <MelchiorGaspar> I'm Melchior [16:04:21 CEST] <MelchiorGaspar> LOL OMG AUOMATIC POSTING OF BUGTRACKER TICKETS OOH YAH!!!! [16:08:05 CEST] <MelchiorGaspar> FTP access to the http://samples.ffmpeg.org/playstation/ does not seem to work... ;_; because I cannot upload files createer then 2.5MB to the tracker and most of these vid files are easly 50MB+ [16:09:14 CEST] <MelchiorGaspar> is there anyway to get upload permissions to the samples server? [16:12:32 CEST] <michaelni> MelchiorGaspar, http://ffmpeg.org/bugreports.html see "Submitting Sample Media" [16:14:01 CEST] <MelchiorGaspar> ty [16:14:59 CEST] <BtbN> Is ffmpeg supposed to play those but fails, or is that something entirely new? [16:18:00 CEST] <MelchiorGaspar> ok got it I attached a few small files to the bug report as can be seen just above, I have reg'd the upload url to Filezilla I will create a sub folder withthe tarcker id# [16:22:53 CEST] <MelchiorGaspar> the upload server does not like me.... [16:25:27 CEST] <MelchiorGaspar> ok it likes me know I had to set it too one connection only [16:28:35 CEST] <cone-235> ffmpeg 03Paul B Mahol 07master:d2fc702ace34: avfilter: add chorus filter [16:37:35 CEST] <MelchiorGaspar> one set of uploads are done [16:37:43 CEST] <MelchiorGaspar> ./incoming/Ticket-4478_Suikoden3 [16:39:28 CEST] <MelchiorGaspar> ./incoming/Ticket-4478_Suikoden4 <-- Is uploading now , 756MB worth of .pss files [17:01:32 CEST] <MelchiorGaspar> ok uploads are done [17:02:19 CEST] <sfan5> MelchiorGaspar: EV50511.PSS plays fine for me in mpv [17:15:01 CEST] <cone-235> ffmpeg 03Michael Niedermayer 07master:4270a9f547bd: avformat/sierravmd: use av_malloc_array() [17:15:02 CEST] <cone-235> ffmpeg 03Michael Niedermayer 07master:870ec3f69e04: avformat/segafilm: Use av_malloc_array() [17:56:37 CEST] <cone-235> ffmpeg 03Michael Niedermayer 07master:445a02b1ec5e: avformat/rtpdec_asf: Fix potential pointer overflow [17:57:25 CEST] <wm4> Assertion !s->out_ch_layout || out->ch_count == av_get_channel_layout_nb_channels(s->out_ch_layout) failed at libswresample/rematrix.c:446 [17:57:28 CEST] <wm4> what does this mean [17:57:33 CEST] <wm4> it works with libavresample [17:58:16 CEST] <wm4> libswresample unhelpfully prints "Input channel layout has a different number of channels than the number of used channels, ignoring layout" before that, but opening the context succeeds [18:00:18 CEST] <michaelni> how can that be reproduced ? [18:08:56 CEST] <wm4> michaelni: with mpv git master (from moments ago) with a sample that switches from 48000Hz stereo 2ch floatp to 48000Hz 5.1 6ch floatp and -v --speed=1.1 --no-audio-pitch-correction --ao=null [18:09:19 CEST] <wm4> michaelni: it actually triggers reusing the same swresample context for a format change (while normally the context is destroyed and recreated) [18:10:22 CEST] <MelchiorGaspar> [11:02] <sfan5> MelchiorGaspar: EV50511.PSS plays fine for me in mpv <-- I use AvantiGUI w/ FFmpeg [18:11:11 CEST] <sfan5> i use ffmpeg version N-71399-g816bbc1 Copyright (c) 2000-2015 the FFmpeg developers (in case that helps in some way) [18:12:14 CEST] <michaelni> wm4, where can i find the/a input sample for this ? [18:16:58 CEST] <wm4> michaelni: https://0x0.st/Tv.mkv [18:30:44 CEST] <BtbN> What's the best video/image filter to look at for a simple pixel-modification-filter example? [18:36:13 CEST] <Timothy_Gu> how can you check the GPT of an object again? [18:40:18 CEST] <MelchiorGaspar> I get my FFmpeg bin from, http://ffmpeg.zeranoe.com/builds/ [18:40:24 CEST] <MelchiorGaspar> nap time [18:41:00 CEST] <wm4> BtbN: maybe vf_eq? [18:44:42 CEST] <Timothy_Gu> for some reason `-pie` is needed for ffmpeg to link on haiku x86-64 [18:44:48 CEST] <Timothy_Gu> is that the right thing to do? [18:45:58 CEST] <wm4> btw. why care about Haiku at all [18:46:09 CEST] <Timothy_Gu> fun :p [18:46:25 CEST] <wm4> it's 100 times more obscure than using BSD on the desktop, and 1000 times more obscure than using Linux on the desktop [18:46:33 CEST] <sfan5> what about GNU/Hurd [18:46:44 CEST] <wm4> Hurd is actively developed [18:47:04 CEST] <wm4> but there are probably more people caring about Windows 95... [18:47:18 CEST] <Timothy_Gu> pretty sure more people care about haiku than hurd [18:50:13 CEST] <iive> talking about hurd - http://xkcd.com/1508/ [18:54:39 CEST] <cone-235> ffmpeg 03Timothy Gu 07master:7ad27f122156: fate: Include branch information in the payload header [19:19:07 CEST] <Timothy_Gu> fate passes so meh [20:20:30 CEST] <Timothy_Gu> Can someone please explain how `cextern` works in conjunction with shared library? Does it use the GOT? If not how/why? [20:21:08 CEST] <Timothy_Gu> michaelni, jamrial, ubitux, BBB_ ^^ [20:24:12 CEST] <Timothy_Gu> disassembling objects built with PIC seems to reveal that they do not use the GOT and uses R_X86_64_PC32, but http://eli.thegreenplace.net/2011/11/11/position-independent-code-pic-in-sh… says that R_X86_64_PC32 cannot be used with shared libs, so I'm really confused right now& [20:35:18 CEST] <BBB_> Timothy_Gu: I think its just an external reference resolved by the linker [20:35:26 CEST] <BBB_> so no, it doesnt do pic relocation resolution for you [20:35:31 CEST] <BBB_> thats all up to you [20:35:48 CEST] <BBB_> but it allows you to use stuff (e.g. tables) stored in anoher file [20:39:37 CEST] <BBB> Timothy_Gu: youll notice various places in our asm code doing special PIC magic [20:39:42 CEST] <BBB> under %if PIC [20:39:48 CEST] <BBB> or %ifdef PIC [20:39:55 CEST] <BBB> thats how it works [20:45:23 CEST] <Timothy_Gu> BBB: the "external reference" is using R_X86_64_PC32, which fails on Haiku [20:45:24 CEST] <michaelni> wm4, you should be able to workaround this by setting the input/output channel counts ("ich"/"och") but ill push a fix soon and will backport it [20:45:41 CEST] <Timothy_Gu> (and I wonder why it doesn't fail on linux as well) [20:45:50 CEST] <BBB> theres a lot of magic in x86inc.asm [20:46:00 CEST] <Timothy_Gu> who wrote x86inc [20:46:01 CEST] <Timothy_Gu> ? [20:46:38 CEST] <Daemon404> pengvado, [20:46:40 CEST] <Daemon404> did [20:46:45 CEST] <Timothy_Gu> oh [20:46:50 CEST] <Daemon404> a lot of bugmaster too iirc [20:46:58 CEST] <BBB> ok, so [20:47:01 CEST] <BBB> first of all [20:47:03 CEST] <BBB> is PIC defined? [20:47:06 CEST] <Timothy_Gu> yes [20:47:10 CEST] <BBB> in the assembly header? [20:47:16 CEST] <BBB> (config.asm) [20:47:16 CEST] <Timothy_Gu> config.asm? yes [20:47:24 CEST] <BBB> is this a 64bit build? [20:47:26 CEST] <Daemon404> BBB, this is haiku so i bet its just toolchain bugs [20:47:29 CEST] <Daemon404> just sayin. [20:47:38 CEST] <Timothy_Gu> yes [20:47:42 CEST] <BBB> Daemon404: I know, but x86inc.asm has some magic that Ive played with, so we can rule out that [20:47:54 CEST] <Timothy_Gu> or rather PIC is defined in Makefile (-DPIC) [20:49:28 CEST] <BBB> and that is included in ASMFLAGS (not CFLAGS)? [20:49:36 CEST] <BBB> ASFLAGS I guess [20:49:51 CEST] <cone-235> ffmpeg 03James Almer 07master:14edc9f3a392: configure: add missing h263p decoder dependencies [20:49:52 CEST] <Timothy_Gu> YASMFLAGS [20:50:30 CEST] <Timothy_Gu> ok so YASMFLAGS and CPPFLAGS have -DPIC, while CFLAGS and ASFLAGS have -fPIC [20:50:39 CEST] <wm4> michaelni: thanks, then I will wait for the fix [20:52:19 CEST] <Timothy_Gu> Daemon404: probably, but it's fun to figure out why it's a bug in the toolchain so that the next i see this i can say "hey this is OBVIOUSLY a toolchain bug" [20:52:21 CEST] <kierank> what's h263i [20:52:22 CEST] <kierank> ? [20:52:48 CEST] <Timothy_Gu> kierank: i = intel? [20:52:57 CEST] <jamrial> intel, yes [20:53:08 CEST] <kierank> and p is? [20:53:36 CEST] <BBB> Timothy_Gu: and what is the exact error message? [20:53:45 CEST] <j-b> kierank: Plus [20:53:54 CEST] <Timothy_Gu> BBB: http://fate.ffmpeg.org:8080/log/x86_64-haiku-gcc-4.8/20150412162931/compile [20:54:06 CEST] <Timothy_Gu> ld: libavcodec/libavcodec.a(deinterlace.o): relocation R_X86_64_PC32 against symbol `ff_pw_4' can not be used when making a shared object; recompile with -fPIC [20:54:35 CEST] <Timothy_Gu> kierank: "H.263+ / H.263-1998 / H.263 version 2" [21:02:25 CEST] <BBB> Timothy_Gu: is it only deinterlace.asm? [21:02:47 CEST] <Timothy_Gu> nope, everything using the masks [21:03:11 CEST] <Timothy_Gu> BBB: Also if I link with -pie the error goes away [21:04:45 CEST] <BBB> hm ok thats very obscure then [21:04:47 CEST] <Timothy_Gu> BBB: I tried compiling a very simple C program that references a mask, and objdump shows that it's using R_X86_64_GOTPCREL rather than _PC32 [21:05:58 CEST] <BBB> so, my understanding is that were doing all gots directy in our code [21:06:15 CEST] <BBB> so the indicator is kind of unnecessary [21:06:30 CEST] <BBB> but Im not an expert or anything [21:07:27 CEST] <Timothy_Gu> let me boot the 32-bit vm and see what it does there [21:09:28 CEST] <BBB> on 32bit we disable pic [21:09:37 CEST] <Timothy_Gu> oh wait yeah [21:09:40 CEST] <BBB> since pic would cost you one precious register and we dont want that [21:09:46 CEST] <BBB> (you only have 7) [21:09:50 CEST] <Timothy_Gu> x86 [21:10:41 CEST] <Timothy_Gu> i tried using qemu without kvm this morning& [21:57:33 CEST] <iive> i actually wonder why x86_64 is not pic by default. [21:57:48 CEST] <iive> i mean, the ffmpeg configure/build [21:58:18 CEST] <Daemon404> because ricing [21:58:23 CEST] <Daemon404> non-PIC is faster [21:58:39 CEST] <iive> is that true on 64bit too? [21:58:47 CEST] <Daemon404> im not sure actually [22:17:07 CEST] <rcombs> I was under the impression it wasn't on x64 [22:23:58 CEST] <Daemon404> dont call it x64... thats a MS thing [22:28:37 CEST] <cone-235> ffmpeg 03Michael Niedermayer 07master:d7b9cb2f7a51: swresample: Allow reinitialization without ever setting channel counts [22:28:38 CEST] <cone-235> ffmpeg 03Michael Niedermayer 07master:80a28c7509a1: swresample: Allow reinitialization without ever setting channel layouts [22:28:39 CEST] <cone-235> ffmpeg 03Michael Niedermayer 07master:3c77bb5f23b2: swresample: Check channel layouts and channels against each other and print human readable error messages [22:28:40 CEST] <cone-235> ffmpeg 03Michael Niedermayer 07master:ff50b1b13be9: swresample/swresample-test: Randomly wipe out channel counts [22:49:53 CEST] <Timothy_Gu> iive: well having one more register is always a good thing right? [22:54:23 CEST] <iive> Timothy_Gu: well, wasn't the point of x86_64 that it doesn't need one extra register? RIP-relative addressing and stuff? [23:23:11 CEST] <Timothy_Gu> shoot I broke fate [23:27:19 CEST] <Timothy_Gu> there, fixed [23:53:41 CEST] <cone-235> ffmpeg 03Himangi Saraogi 07master:f03dfe118b64: avformat/rtsp: Fix dereference after null check [00:00:00 CEST] --- Mon Apr 13 2015

1 0

ffmpeg.log.20150412
by burek 13 Apr '15

13 Apr '15

[00:04:33 CEST] <jookiyaya> didn't work [00:09:05 CEST] <worried> guess i will need to dump with vlc [00:09:12 CEST] <worried> thanks anyway [00:18:00 CEST] <jookiyaya> anybody know how extract subtitles .idx/.vob from MKV file (i know how to do it from dvd (video_ts)) [02:39:29 CEST] <chungy> When using a DVD as an input source, how do you select titles? [06:08:30 CEST] <jookiyaya> if you had to choice of making ntsc or pal universal, which one would you make universal? [07:11:00 CEST] Last message repeated 1 time(s). [09:22:01 CEST] <DiegoMax> hello everyone [09:22:19 CEST] <DiegoMax> i have a quick question that maybe some of you would be able to answer [09:22:42 CEST] <DiegoMax> is it ok to statically link against an LGPL build of ffmpeg on a closed source application ? [10:25:08 CEST] <JEEBsv> DiegoMax: IANAL, but you have to provide the means for the user to replace the library. In other words, the source code for the FFmpeg, and the object files for the proprietary part if you are doing static linking [10:25:24 CEST] <JEEBsv> which is why most people do shared linking with LGPL :P [10:25:41 CEST] <DiegoMax> ya i guess i will just do that [10:26:20 CEST] <DiegoMax> the source and object files must be included into the app binary or may be a link for the user to get them from a web server as an extra if he wants to ? [10:26:32 CEST] <DiegoMax> i tried to find some specifics about that but i failed :p [10:28:51 CEST] <JEEBsv> That's because it's really not specified (well, other details are specified regarding various things), other than you have to make sure it's available for N years (I think five?). If you want to take it the sure way you could have them be in the installer or whatever and just not install by default, otherwise just have the sources(+ object files etc needed for linking) in an archive linked somewhere [10:29:18 CEST] <DiegoMax> thanks JEEBsv [10:29:36 CEST] <JEEBsv> of course, IANAL and all that jazz [10:29:43 CEST] <JEEBsv> but that's my understanding [10:29:55 CEST] <DiegoMax> and what about h264/265 decoding ? [10:30:02 CEST] <DiegoMax> my app does that, in between other things [10:30:05 CEST] <DiegoMax> is that ok / [10:30:06 CEST] <DiegoMax> ? [10:31:51 CEST] <JEEBsv> patent licensing is a whole separate thing from software licensing (except in the case of fdk-aac f.ex. where it specifically adds requirements regarding that) [10:33:04 CEST] <JEEBsv> (which is why fdk-aac is not compatible with certain licenses and thus the resulting binary isn't distributable at all if you link against it) [10:33:25 CEST] <DiegoMax> im working on a really simple tool to do a movie sheet with thumbnails every x seconds [10:34:41 CEST] <DiegoMax> but im now worried about that [10:34:54 CEST] <DiegoMax> i guess they will not really care unless u make millions, and since its not even a paid app [10:35:03 CEST] <DiegoMax> probably they will not care at all [10:35:35 CEST] <DiegoMax> https://www.dropbox.com/s/7mhdf1vmj5ep043/Screenshot%202015-04-12%2005.32.5… thats really it, plus several export/print options [10:35:38 CEST] <JEEBsv> well, I am not going to be able to assess your needs or your local legislation for obvious reasons; if you really want to be sure you go and check the licensing rules for example with MPEG-LA [10:36:05 CEST] <JEEBsv> I think MPEG-LA only starts wanting money for AVC decoder distribution after N copies, where N was either 10k or 100k :P [10:36:12 CEST] <JEEBsv> HEVC I have no idea [10:36:23 CEST] <JEEBsv> and that number at least with AVC reset every year [10:36:30 CEST] <DiegoMax> thanks for all this info JEEBsv [10:38:07 CEST] <DiegoMax> are you one of FFmpeg's devs JEEBsv, or a user only? [10:38:55 CEST] <JEEBsv> I have at least one file with my name in the copyright field, but I'm not sure if I can call myself "dev" :V [10:39:09 CEST] <DiegoMax> well your API/Libs rock then [10:39:24 CEST] <DiegoMax> i never touched this library before 2 days ago [10:39:30 CEST] <DiegoMax> and in 48 hours the app is up and running [10:39:34 CEST] <DiegoMax> i love it [10:39:44 CEST] <JEEBsv> it's much simpler these days with the examples etc around [10:39:55 CEST] <DiegoMax> yeah i guess [10:40:15 CEST] <JEEBsv> before you had to search for random examples and those generally were years old [10:43:22 CEST] <DiegoMax> i think i have found a few memory leaks, but im not 100% positive yet that they are from libav.... [10:43:49 CEST] <DiegoMax> i would say that im 90%... if i get to be 100% after some more debugging, i will probably report it [10:46:02 CEST] <JEEBsv> valgrind time? [10:46:11 CEST] Action: JEEBsv adores valgrind [10:48:20 CEST] <Rayray> hi guys [10:49:34 CEST] <Rayray> currently running into an issue when compiling Motion, the error comes back to ffmpeg/libavutil. [10:49:41 CEST] <Rayray> ffmpeg/libavutil/time.c:60: undefined reference to `clock_gettime' [10:50:20 CEST] <Rayray> thats what i get when i run make [12:15:54 CEST] <MelchiorGaspar> Anybody alive in here all the Dev's in said channel apper to be lost in a blackhole ;D [13:03:54 CEST] <JamJams> Hi! Does anybody know an easy way to use av_seek_frame to seek to the next I frame? Are libavcodec questions ok here? [13:30:26 CEST] <BtbN> Isn't that what it does anyway? [13:30:32 CEST] <BtbN> "Seek to the keyframe at timestamp." [13:30:52 CEST] <BtbN> So just seek to your current timestamp, and it should seek you to the nearest keyframe. [13:31:49 CEST] <MelchiorGaspar> -------------------------------------------- [13:31:49 CEST] <MelchiorGaspar> I found a format that FFmpeg does not support though I thought it did [13:31:49 CEST] <MelchiorGaspar> .pss PLaystation 2 (Suikoden 3 and 4) [13:31:49 CEST] <MelchiorGaspar> I extracted from my PS2 DvD discs so I convert to watch (oohh the memories) soI can upload a file if a Dev wants to test it [13:31:49 CEST] <MelchiorGaspar> -------------------------------------------- [13:38:13 CEST] <JamJams> BtbN what if I'm at the start of the file? [13:38:26 CEST] <JamJams> I want to just be able to hit a button and it go to the next i frame [13:38:36 CEST] <JamJams> Over and over again if need be [18:18:24 CEST] <obXiDeJSFQ> Question:- When making tiles[0], is it possible for it to automatically choose the amount of tiles required to capture the full video? For instance, if I give it a one hour video at one image per ten seconds, it'll throw me back a 360x10 (Or something) mosaic automatically? [18:18:30 CEST] <obXiDeJSFQ> [0] https://trac.ffmpeg.org/wiki/How%20to%20take%20multiple%20screenshots%20to%… [18:29:18 CEST] <Nitori> I just got a kernel panic while encoding... it's not, per chance, possible to resume a failed encode? :< [18:46:26 CEST] <c_14> mp4 output? [18:48:27 CEST] <c_14> Nitori: ^ [18:48:38 CEST] <Nitori> mkv [18:48:59 CEST] <c_14> Then it should be possible. [18:49:55 CEST] <c_14> Find out how far the output got (ffprobe or whatever) rerun the encoding command with a new output file (and -ss [length you got earlier] as an input option). Then concatenate the 2 files with -c copy and the concat demuxer [18:50:40 CEST] <Nitori> hm okay, thanks [19:57:31 CEST] <flashmaster77> hello there [19:57:57 CEST] <flashmaster77> is anyone of you useing a chromecast with castnow? [20:00:09 CEST] <flashmaster77> i set up castnow on my laptop to cast some .avi video to my chromecast, which didn't work because chromcast can't play .avi files. so i figured out it has to be converted via ffmpeg while streaming, but i can't get it to do that. any ideas? [20:16:06 CEST] <jxself> Hmm. I think I am missing something. I have a video containing WebVTT subtitles: http://dpaste.com/2MARHVM When playing in FFplay I press the lett 't' expecting it to "Cycle subtitle channel in the current program" and that output does show "Switch subtitle stream" in response to my pressing that letter but nothing appears on the screen. I can extract the subtitles and verfiy that they are correct. What am I missing or not doing or whatever? [21:24:58 CEST] <smo_> hi [21:25:04 CEST] <smo_> how can i fix errors like this Oo [21:25:07 CEST] <smo_> Error parsing AAC extradata, unable to determine samplerate [21:25:17 CEST] <smo_> on a twitch stream [21:27:05 CEST] <smo_> full output here [21:27:06 CEST] <smo_> http://paste.ubuntu.com/10810718/ [21:27:10 CEST] <smo_> thanks :p [21:27:41 CEST] <smo_> spawn : -re,-i,pipe:0,-sn,-vf,scale=trunc(iw/2)*2:trunc(ih/2)*2,-c:v,libx264,-preset,ultrafast,-deinterlace,-c:a,copy,-threads,0,-f,matroska,pipe:1 grep stderr: ffmpeg version 2.6.1- http://johnvansickle.com/ffmpeg/ [21:27:48 CEST] <smo_> precision... [21:28:04 CEST] <Guest40174> hi all [21:29:25 CEST] <Guest40174> I use ffmpeg with option -ss (seek). My system kill process ffmpeg. Little memory. what do i do? [21:31:07 CEST] <Guest40174> my options [21:31:10 CEST] <Guest40174> ffmpeg -y -ss 2000 -i "http:// ... etc" -vcodec mjpeg -vframes 1 -an -f image2 -s "1680x1050" "test.jpg" -benchmark [21:31:30 CEST] <Guest40174> Very load my host [21:31:51 CEST] <Guest40174> ok, im sorry [21:33:02 CEST] <smo_> http://paste.ubuntu.com/10810758/ [21:36:28 CEST] <YaMoonSun> How would I convert a 5.1 surround sound AAC audio stream to 48Kbps and 2 channels, OGG? [21:37:46 CEST] <c_14> YaMoonSun: `ffmpeg -i aac -c:a libvorbis -b:v 48k -ac 2 out.ogg' [21:38:12 CEST] <YaMoonSun> Ily, going to give it a shot, thanks. [21:38:12 CEST] <jxself> I think -b:a since it's audio? [21:38:18 CEST] <c_14> eh, right [21:38:33 CEST] <c_14> This is what happens when you don't pay attention to what you're writing. [21:38:55 CEST] <YaMoonSun> Winrar, looks like it's going to work, chears. [21:39:10 CEST] Action: YaMoonSun can't beleive I spelled cheers wrong [21:39:59 CEST] <c_14> smo_: you'll probably have to reencode the audio [21:40:10 CEST] <c_14> Also, -deinterlace is deprecated. Use the yadif (or similar) filter. [21:40:51 CEST] <YaMoonSun> Am making a webm, but I think I should probably do it from the .mkv so I don't transcode any furter. [21:41:08 CEST] <smo_> that s what i m trying c_14 thanks [21:41:10 CEST] <c_14> Guest40174: Get more memory or smaller (resolution) input. Not sure what else I can tell you. [21:42:11 CEST] <Guest40174> c_14, i use 1gb ram [21:42:39 CEST] <c_14> You can also try killing all unnecessary background processes to free up more memory for ffmpeg. [21:44:51 CEST] <Guest40174> dont [21:45:10 CEST] <Guest40174> i small resolution, but killed :( [21:49:47 CEST] <c_14> Guest40174: try adding swap space [21:51:42 CEST] <Guest40174> c_14, how many? [21:51:57 CEST] <c_14> At least 1 preferably 2GiB [22:21:56 CEST] <solrize> i have several files that i want to concatenate (e.g. paste several songs into an album that i can play back as one file). is there a simple way to do that, preferably without transcoding? [22:22:24 CEST] <c_14> https://trac.ffmpeg.org/wiki/Concatenate [22:22:42 CEST] <solrize> aha thanks [22:27:06 CEST] <DragonsLord> c_14, do you know some guidelines in order to use FFmpeg to make backup copies of DVDs? [22:27:35 CEST] <c_14> If all you need is a backup copy, just use dd? [22:27:54 CEST] <DragonsLord> no, I mean ... [22:28:29 CEST] <DragonsLord> ... let's suppose I have VTS_04_0.IFO + VTS_04_0.VOB + VTS_04_1.VOB [22:29:05 CEST] <DragonsLord> FFmpeg is able to read the IFO so tu understand if there is the need to pre-load the audio [22:29:14 CEST] <DragonsLord> or to locate some other stuffs? [22:29:28 CEST] <DragonsLord> and can I "concatenate" at source [22:29:47 CEST] <DragonsLord> I mean tell FFmpeg to encode VTS_04_0.VOB + VTS_04_1.VOB together? [22:29:56 CEST] <DragonsLord> (sorry for my bad English, once more) [22:30:19 CEST] <Mavrik> Honestly, ffmpeg is probably not the best tool for that [22:30:32 CEST] <Mavrik> dedicated DVD conversion tools do that better [22:31:45 CEST] <DragonsLord> they failed [22:31:56 CEST] <DragonsLord> at least in terms of A/V syncronization [22:32:04 CEST] <DragonsLord> I would like to give FFmpeg a try [22:43:24 CEST] <c_14> DragonsLord: you can try concatenating the vobs, but ffmpeg doesn't have a dedicated dvd demuxer [22:44:20 CEST] <YaMoonSun> DVDDecrypter + Handbrake = winrar [22:50:21 CEST] <RobotsOnDrugs> mplayer does a decent job of turning dvd files into complete mpeg streams of chapters or titles [22:51:22 CEST] <RobotsOnDrugs> there are better tools for it, but afaik, they're all gui-based and don't really do automation (or they do and they cost money) [22:55:54 CEST] <DragonsLord> thank u [23:10:01 CEST] <DragonsLord> Good night & sweet dream to everybody ... chat to you tomorrow :) [23:10:28 CEST] Action: DragonsLord is away: AWAY [00:00:00 CEST] --- Mon Apr 13 2015

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ffmpeg-devel.log.20150411
by burek 12 Apr '15

12 Apr '15

[00:00:18 CEST] <wm4> how does git LFS work? (yeah I'm totally too lazy to look it up) [00:00:40 CEST] <Daemon404> its github's thing [00:00:43 CEST] <Daemon404> google it [00:01:12 CEST] <llogan> he can't. too lazy. [00:02:02 CEST] <Daemon404> it works by running some code and doing some operations [00:02:04 CEST] <Daemon404> presumably on file.s [00:05:12 CEST] <wm4> so it basically works around git's database for file contents [00:05:30 CEST] <wm4> and stores the actual contents separately [00:05:47 CEST] <wm4> reminds me of git submodules, I hope its not as bad [00:14:33 CEST] <jamrial> a sample collection per release branch is surely a good idea. files currently can be added but not removed because old versions will still need them to run fate [01:30:00 CEST] <cone-166> ffmpeg 03Timothy Gu 07master:5faca08cafc1: texi2pod: Handle @verbatim [01:30:01 CEST] <cone-166> ffmpeg 03Timothy Gu 07master:3d6069d01c8b: Use @verbatim instead of @example for ASCII arts [02:02:43 CEST] <cone-166> ffmpeg 03James Almer 07master:410c93cfd5ab: doc: add missing x86 cpuflags to fftools documentation [02:02:44 CEST] <cone-166> ffmpeg 03James Almer 07master:666ec9bd0972: doc: add missing arm cpuflags to fftools documentation [02:02:45 CEST] <cone-166> ffmpeg 03James Almer 07master:9fc45681e0c4: doc: add aarch64 cpuflags to fftools documentation [02:27:02 CEST] <cone-166> ffmpeg 03Martin Storsjö 07master:4f373a5111f9: vfwcap: Unbreak building after c201069fa [02:27:03 CEST] <cone-166> ffmpeg 03Michael Niedermayer 07master:d4a6c9404601: Merge commit '4f373a5111f900af54301907132942f95276285c' [02:44:28 CEST] <cone-166> ffmpeg 03Andrey Utkin 07master:7f64a7503b19: rtpenc_jpeg: handle case of picture dimensions not dividing by 8 [03:01:22 CEST] <Timothy_Gu> ls [03:01:26 CEST] <Timothy_Gu> oops [04:13:26 CEST] <Timothy_Gu> OK I think i set up a haiku vm correctly. let's see how it rols. [05:41:09 CEST] <jamrial> Timothy_Gu: that's a weird error your latest edit generated in the wiki [06:15:14 CEST] <Timothy_Gu> jamrial: I was trying to use syntax highlighting provided by trac& [06:15:30 CEST] <Timothy_Gu> jamrial: it works for some languages like C but not shell [06:15:38 CEST] <jamrial> just pointed it out in case you missed it [06:16:55 CEST] <Timothy_Gu> also for some reason I need to explicitly define HAVE_6REGS to 0 on Haiku& [06:18:06 CEST] <jamrial> do both ebx_available and ebp_available configure checks succeed? [06:18:23 CEST] <Timothy_Gu> jamrial: no, ebx didn't but ebp did [06:18:35 CEST] <Timothy_Gu> and gcc is complaining about impossible constraints [06:19:51 CEST] <jamrial> as long as one of those succeeds HAVE_6REGS is set to true [06:20:17 CEST] <Timothy_Gu> jamrial: yes but compilation fails even though ebp is clearly available [06:20:23 CEST] <jamrial> are you using the same compiler as michaelni? [06:20:42 CEST] <jamrial> could be a gcc bug [06:21:08 CEST] <Timothy_Gu> michaelni's haiku box died a long time ago [06:21:19 CEST] <Timothy_Gu> and no i don't think so [06:21:35 CEST] <Timothy_Gu> (i thin his was using 4.4 or something while im using 4.8) [06:21:51 CEST] <Timothy_Gu> i'll check [06:21:54 CEST] <jamrial> ah, then you can open a ticket in gcc's bugzilla [06:22:08 CEST] <jamrial> you may be lucky and it will get fixed before they close the 4.8 branch :p [06:22:39 CEST] <jamrial> which they will do after 4.8.5 is released, i guess [06:22:41 CEST] <Timothy_Gu> jamrial: they are unlikely to fix anything related to haiku :d [06:23:13 CEST] <jamrial> well, assuming there is a haiku maintainer, he's unlikely to have many bug reports to deal with :p [06:24:01 CEST] <Timothy_Gu> actually quite the opposite :p [06:24:23 CEST] <Timothy_Gu> anyway, micahel's box was using 4.3.3, and it's on x86-64 [06:24:30 CEST] <Timothy_Gu> while mine is 4.8.* on x86 [06:25:22 CEST] <Timothy_Gu> 4.8.4 (2014_12_21) [06:33:10 CEST] <Timothy_Gu> "PIC register clobbered by '%ebx'" [06:38:27 CEST] <Timothy_Gu> jamrial: seems like Haiku builds EVERYTHING with -fPIC, which gives FFmpeg a performance hit so big they had to explicitly disable it in their build: https://github.com/mirror/haikuports/blob/master/media-video/ffmpeg/ffmpeg-… [06:39:04 CEST] <Timothy_Gu> or rather they don't build with -fPIC but rather hardcode it in their GCC build& [06:57:02 CEST] <Timothy_Gu> jamrial: seems to be this issue: https://gcc.gnu.org/bugzilla/show_bug.cgi?id=47602 it was filed 4 years ago and was fixed about 6 months ago [06:58:32 CEST] <jamrial> doesn't look like it was backported [06:59:31 CEST] <Timothy_Gu> no it wasn't [07:00:53 CEST] <Timothy_Gu> even on my ubuntu machine it's broken [07:01:09 CEST] <Timothy_Gu> now the question is why configure didn't catch that [07:06:15 CEST] <Timothy_Gu> wait, the issue is about ebx [07:06:24 CEST] <Timothy_Gu> ugh time to get some sleep [13:59:30 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07master:e0d8ff5ef105: avformat/nsvdec: Use av_malloc_array() [13:59:30 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07master:e33355213daf: avformat/wtvenc: Use av_realloc_array() [13:59:30 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07master:d6bcdf0dcdd7: avformat/xwma: Use av_malloc_array() [14:52:27 CEST] <BBB> so, theres this interesting discussion in libav-devel where they want to change some unsigned int to uint32_t [14:52:33 CEST] <BBB> can we please not ever merge that sort of stuff? [14:53:13 CEST] <kierank> why do they want to do that [14:53:28 CEST] <BBB> I cant possibly imagine any situation where its warranted [14:53:36 CEST] <BBB> but its diego and a new contributor [14:53:37 CEST] <BBB> so... [14:54:26 CEST] <kierank> ah [14:55:15 CEST] <j-b> Unfortunately, sometimes this makes sense [14:56:44 CEST] <Compn> BBB : bring it up on ffmpeg-devel [14:56:50 CEST] <Compn> otherwise will be forgotten on irc [14:57:06 CEST] <Compn> quote or reply from libav-devel thread too for maximum effect [14:57:23 CEST] <Compn> or at least ping michaelni so he sees this request here :P [14:58:30 CEST] <j-b> technically, unsigned int could be only 0-65535 range [14:58:46 CEST] <j-b> while uint32_t is guaranteed to be at least 32bits [14:59:10 CEST] <michaelni> posix gurantees (unsigned) int to be 32bit at least [14:59:15 CEST] <BBB> posix means 32bit [14:59:17 CEST] <BBB> right [14:59:23 CEST] <BBB> so, no, unsigned int is >= 32 bit [14:59:37 CEST] <kierank> http://comments.gmane.org/gmane.comp.gdb.patches/106689 [14:59:40 CEST] <BBB> but adjusts to nativity of the system running on, i.e. its not guaranteed to be 32bit on systems where 64bits ay be more performant [14:59:40 CEST] <kierank> can we do this please too [14:59:57 CEST] <BBB> so unsigned int for non-array types where the specific scope/size makes no difference, is better [15:06:49 CEST] <Compn> kierank : apply the patch to gdb or introduce av_16bit_something ? [15:12:42 CEST] <wm4> kierank: also, POSIX requires chars to be 8 bits [15:15:15 CEST] <kierank> "This explains how unsigned arithmetic wraparound works, not what happens [15:15:16 CEST] <kierank> when you assign negative values to unsigned variables." [15:15:16 CEST] <kierank> lol'd [15:15:39 CEST] <kierank> oh well bbq time [15:20:51 CEST] <wm4> haha nicolas gets repeatedly accused of trolling [15:20:56 CEST] Action: wm4 gets popcorn [16:21:33 CEST] <lglinskih> hi! what does the `align` parameter of av_frame_get_buffer do? [16:22:34 CEST] <lglinskih> the doc says "buffer size alignment". What is the alignment of a size? [16:25:47 CEST] <wm4> lglinskih: it's the byte alignment of the data [16:26:19 CEST] <wm4> lglinskih: so for video, it means the pointers to each 1st pixel of a line is aligned to this value [16:26:29 CEST] <wm4> if you don't know what it means, just set it to 32 [16:31:01 CEST] <lglinskih> wm4: that's what I did=) [17:59:50 CEST] <cone-750> ffmpeg 03Michael Kostylev 07master:7a9b764c0737: libdc1394: Unbreak build after c201069fa [17:59:51 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07master:6530e2ff0cd0: Merge commit '7a9b764c0737f42cf2458c3c5378b0df216e14a2' [18:12:43 CEST] <cone-750> ffmpeg 03James Almer 07master:bbdb50d7a8a9: libx265: print supported presets and tunes on error [18:12:44 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07master:816bbc1cf532: Merge commit 'bbdb50d7a8a91f38188fd15080d7f45f1540b3ac' [19:22:37 CEST] <kierank> OK now Nicholas wants an http server [19:26:28 CEST] <Daemon404> kierank, it as his gsoc ide.a [19:26:32 CEST] <Daemon404> was* [19:30:24 CEST] <wm4> what's the http server for? [19:31:14 CEST] <j-b> ponr [19:35:40 CEST] <Compn> mongoose project is a simple http server that offers whatevers in the active dir , http://code.google.com/p/mongoose/ [19:35:44 CEST] <Compn> got a libmongoose too :P [19:37:19 CEST] <wm4> there are probably hundreds of http servers [19:47:47 CEST] <kierank> wm4: nih [20:08:28 CEST] <rcombs> this whole Perseus thing would be very interesting if there was any evidence it actually exists [20:13:09 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07master:f944aeb07aff: avformat/rtpproto: Move dscp into context & AVOptions [22:12:31 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07release/2.6:f597b9f04e69: avutil/pca: Check for av_malloc* failures [22:12:32 CEST] <cone-750> ffmpeg 03Rainer Hochecker 07release/2.6:7689fe5cfd28: h264: avoid unnecessary calls to get_format [22:12:33 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07release/2.6:fa538f1a8c33: Revert "avcodec/exr: fix memset first arg in reverse_lut()" [22:12:34 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07release/2.6:f1b4a71ddfcd: avcodec/h264: Fail for invalid mixed IDR / non IDR frames in slice threading mode [22:12:35 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07release/2.6:9ee7fcdcd087: avcodec/h264_refs: Do not set reference to things which dont exist [22:12:36 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07release/2.6:9bff35abde67: ffmpeg: Fix extradata allocation [22:12:37 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07release/2.6:ac07ab7db78d: avformat/utils: avoid discarded streams in av_find_default_stream_index() [22:12:38 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07release/2.6:d3f96c1e3cc2: avcodec/h264: Fix race between slices where one overwrites data from the next [22:12:39 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07release/2.6:3550d239a6cb: avcodec/h264: finish previous slices before switching to single thread mode [22:12:40 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07release/2.6:7d5908d5c834: avcodec/h264_slice: Dont reset mb_aff_frame per slice [22:12:41 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07release/2.6:8be177e048ce: avcodec/h264: reset the counts in the correct context [22:12:42 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07release/2.6:bcc4c360aadf: avcodec/aacdec: Fix storing state before PCE decode [22:12:43 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07release/2.6:e6d9094fd3c6: avcodec/h264: Be more tolerant to changing pps id between slices [22:12:44 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07release/2.6:b4bfbbfb95ac: avcodec/h264_ps: Move truncation check from VUI to SPS [22:12:45 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07release/2.6:05b448082ae0: avcodec/h264: Do not fail with randomly truncated VUIs [22:37:09 CEST] <cone-750> ffmpeg 03Michael Niedermayer 07release/2.6:369f46aae3f2: Update for 2.6.2 [23:12:40 CEST] <cone-750> ffmpeg 03Stephan Holljes 07master:b51027fd18f6: libavformat/http.c: Make http-listen work as an input stream. [23:56:35 CEST] <jamrial> michaelni: are documentation commits ok to backport? [23:56:44 CEST] <michaelni> sure [00:00:00 CEST] --- Sun Apr 12 2015

1 0

ffmpeg.log.20150411
by burek 12 Apr '15

12 Apr '15

[00:04:21 CEST] <ChocolateArmpits> DragonsLord: yes however it will most likely not be distributed evenly across the file as compared with other files [00:05:27 CEST] <DragonsLord> whould you be so kind to elaborate? [00:05:47 CEST] <DragonsLord> What happens with AVIdemux is that, for some files, [00:06:01 CEST] <DragonsLord> the average bitrate seems to be not respected [00:06:06 CEST] <DragonsLord> only for some files [00:06:16 CEST] <DragonsLord> I mean 1500 kbps avg bitrate [00:06:26 CEST] <DragonsLord> and the resulting file is something like 2800 :O [00:06:39 CEST] <DragonsLord> of course also the file size is completely different [00:07:02 CEST] <DragonsLord> if compared with the other episodes/files with the same duration [00:07:03 CEST] <DragonsLord> :/ [00:07:32 CEST] <ChocolateArmpits> it probably uses some other ratecontrol rate than the 2 pass method [00:07:57 CEST] <ChocolateArmpits> 2 pass will always result in aproximately the same file size [00:08:43 CEST] <iive> you don't need color key, you need alpha channel [00:09:36 CEST] <DragonsLord> so, what could I do ChocolateArmpits ... in order to have the same filesize for all the episodes? [00:09:40 CEST] <DragonsLord> one more pass? [00:09:50 CEST] <ChocolateArmpits> what? use 2 passes [00:10:01 CEST] <ChocolateArmpits> for variable encoding [00:10:30 CEST] <ChocolateArmpits> or if you want to save time then constant bitrate by setting b:v maxrate and bufsize [00:11:04 CEST] <DragonsLord> I am already using 2-passes VBR [00:11:10 CEST] <ChocolateArmpits> well ok [00:11:38 CEST] <DragonsLord> but, as mentioned before, for some files the "codec" seems to be not able to respect the average bitrate I set [00:11:38 CEST] <Kolizer> there are Russians? [00:13:29 CEST] <ChocolateArmpits> DragonsLord: sorry, can't comment on this, but that shouldn't be ever happening using 2 pass [00:14:48 CEST] <ps-auxw> DragonsLord: What encoder are you using? It seems to be a broken one if it can't respect your bitrate. Or you are encoding perfect noise, maybe. :) [00:15:33 CEST] <DragonsLord> ps-auxw, thank you in advance ... I am using libxvid [00:15:50 CEST] <ps-auxw> That *should* have decent rate control. [00:16:03 CEST] <ps-auxw> Are you sure you are doing both passes? [00:16:08 CEST] <DragonsLord> it respect my bitrate 90% of the times ... but sometimes no :( [00:16:17 CEST] <DragonsLord> YES, I am 100% sure [00:16:19 CEST] <ps-auxw> Okay. [00:16:55 CEST] <DragonsLord> I am going to try also with FFmpeg [00:17:26 CEST] <ps-auxw> You could check the quantizer limits somewhere in xvid's settings. Maybe it can't use high enough quants to reach the target... [00:18:01 CEST] <ps-auxw> Or try switching to x264. It'll probably do better and has much better quality. [00:18:10 CEST] <DragonsLord> I should have set 1 as a min and 31 as a maximum [00:18:24 CEST] <ps-auxw> That sounds okay. [00:18:24 CEST] <DragonsLord> I know but there is not an option [00:18:32 CEST] <DragonsLord> I need XviD or DivX [00:20:17 CEST] <ps-auxw> DragonsLord: How about resolution? If you are encoding HD or something, you might be better off downscaling with XviD. [00:20:27 CEST] <DragonsLord> 704x400 [00:20:47 CEST] <DragonsLord> c_14 & ChocolateArmpits, y=[pixels to cut from the top] this is right? [00:20:59 CEST] <ChocolateArmpits> DragonsLord: yes [00:21:41 CEST] <DragonsLord> y should not also start from 0,0? [00:21:58 CEST] <DragonsLord> so I thought it was from the bottom [00:22:39 CEST] <DragonsLord> looking at the x,y Cartesian coordinate system, wasn't it? [00:22:49 CEST] <ChocolateArmpits> x,y 0,0 is top left [00:23:00 CEST] <ChocolateArmpits> 1919,1079 is bottom right [00:23:32 CEST] <DragonsLord> Well, so it's not the Cartesian coordinate system [00:23:39 CEST] <DragonsLord> :) [00:28:30 CEST] <DragonsLord> ps-auxw, I love your nickname :) [00:28:34 CEST] <DragonsLord> are you still there? [00:42:38 CEST] <ps-auxw> DragonsLord: :) [00:42:48 CEST] <ps-auxw> I'm still here. Got a bit distracted. [00:43:35 CEST] <ps-auxw> I also kind of ran out of ideas. ;) [00:44:37 CEST] <ps-auxw> I suppose you could try adding in some kind of denoising filter to make things more compressible. [00:44:53 CEST] <DragonsLord> I don't really know :| [00:45:20 CEST] <DragonsLord> here is an example of what I am doing [00:45:20 CEST] <DragonsLord> http://pastebin.com/D1FRhPfh [00:46:40 CEST] <ps-auxw> Hmm. [00:48:18 CEST] <ps-auxw> And your commandline for the second pass is the same (except for -pass 1/2 of course)? [00:49:01 CEST] <DragonsLord> yes + the audio of course [00:51:52 CEST] <ps-auxw> Maybe try to see if you can find a setting for hqdn3d or owdenoise filters that doesn't ruin your video but still makes xvid happier. That should go into the -vf part. [00:54:00 CEST] <DragonsLord> I have just tried to perform the conversion with FFmpeg [00:55:00 CEST] <DragonsLord> and it's seems to be that ... [00:55:14 CEST] <DragonsLord> Average Bitrate is ... [00:55:15 CEST] <DragonsLord> 1499.789828 kbps [00:55:20 CEST] <DragonsLord> 100% respected [00:55:21 CEST] <DragonsLord> :D [00:55:23 CEST] <ps-auxw> Oh... So it worked. :) [00:55:30 CEST] <DragonsLord> only with FFmpeg [00:55:34 CEST] <ps-auxw> I wonder what went wrong before... [00:55:44 CEST] <DragonsLord> for some reason (unkonwn to me) [00:55:56 CEST] <DragonsLord> with AVIdemux and Xmedia Recode not [00:56:03 CEST] <ps-auxw> Well, as long as it works now... :) [00:56:20 CEST] <DragonsLord> please I'm curiuous ... tell me :))) [01:00:17 CEST] <ps-auxw> I have no clue either. ;) [01:00:59 CEST] <ps-auxw> Maybe I could see something from a paste of a failed second pass log. [01:01:32 CEST] <DragonsLord> But with ffmpeg it works [01:01:50 CEST] <DragonsLord> and with the other software I don't really know how to produce a log [01:01:52 CEST] <ps-auxw> Maybe avidemux failed to correctly pass all arguments. [01:02:01 CEST] <ps-auxw> Oh well. [01:02:13 CEST] <DragonsLord> thank you for your time, ps-auxw [01:02:25 CEST] <ps-auxw> Sure, you're welcome. :) [11:44:54 CEST] <luc4> Hello! Does anybody know if, starting the playback of a video using ffplay perfectly at the same time on different computers, I should expect a perfect sync? [11:45:35 CEST] <luc4> I know from past experiences that perfect sync is very difficult& but what can explain this? [12:28:38 CEST] <rjp421> send the absolute timecode from the source? [12:29:55 CEST] <iive> luc4: mplayer had an option to sync using udp packets. [12:30:19 CEST] <iive> just running players won't do it, even if they start at the same moment [12:30:52 CEST] <iive> usually audio playback is used as reference clock and video is synced to it [12:31:18 CEST] <luc4> iive: yes, I know, but mplayer is commonly useless for me as I work on embedded. I would just like to know first why this happens, Im pretty curious. [12:31:44 CEST] <iive> however many audio cards have something called frequency drift, where the sampling is a percent off by the 44.1kHz. [12:32:34 CEST] <luc4> iive: ok, so this frequency drift causes video to accumulate somehow a deloy, correct? [12:32:36 CEST] <iive> it could be caused by e.g. temperature of the oscillator. [12:33:06 CEST] <luc4> iive: which means the clock is slightly different, which causes the delay to increase? [12:33:12 CEST] <iive> it would simply mean that all players would play at different speed. [12:33:38 CEST] <iive> you won't notice it at first, but the longer it play, the more they would drift. [12:33:44 CEST] <luc4> iive: very interesting& and so mplayer, or whatever other implementation, keeps this updated by re-seeking? [12:33:55 CEST] <luc4> iive: perfectly reasonable. [12:34:13 CEST] <iive> e.g. STB players have a way to fine-tune the audio sampling freqency. [12:34:50 CEST] <iive> luc4: they have a master server and they hold frames until it says them it is ok to show them. so one leads, the other follows. [12:35:44 CEST] <luc4> iive: oh& so whichever is leading slows down? [12:36:17 CEST] <iive> it's more of a hack... i don't remember if the followers had audio... [12:37:17 CEST] <luc4> iive: I have to say I implemented something similar in the past, I was more interested in knowing why this happened and of how other projects managed to handle these problems. It is very interesting& thanks for your time!! [12:37:34 CEST] <iive> :) [12:40:10 CEST] <iive> i've heard that vlc could resample the audio by software, however the sinc method that is usually used is quite troublesome when conversion frequencies are close. still, it is an option. [12:44:15 CEST] <luc4> iive: interesting is that I noticed this lack of sync even in perfectly identical devices. [12:47:14 CEST] <iive> as i said, it depends on the oscillator, that's analog tech and it always have some tolerance of differences. [12:48:30 CEST] <iive> one way to make it more precise is to use very high frequency oscillator (e.g. like the cpu one) that runs in GHz and divide it down to kHz. [12:48:47 CEST] <iive> since it is embedded device, it would be practical. [12:49:09 CEST] <iive> it would be interesting to see if the CPU speed also differs a little. [12:50:40 CEST] <luc4> iive: yes, it would be very interesting... [12:53:33 CEST] <iive> but hey, I remember that even the first multicore AMD cpu's had synchronization problems. [12:55:05 CEST] <iive> (there is a counter based on clock clicks that is accessible by cpu instruction. however there was slight drift and programs got confused when moved from one core to the other). [16:29:43 CEST] <floholl> Hi all, I've noticed that ffmpeg, by contrast to avconv, supports fades to/from alpha (rather than to/from black), e.g., fade=out:99:25:alpha=1. Is this a fairly recent feature in ffmpeg? Would it justify a switch from avconv to ffmpeg? [16:41:27 CEST] <c_14> 2011 afaict [16:42:34 CEST] <c_14> Whether or not that's worth switching is up to you. [16:43:32 CEST] <floholl> c_14: Thanks [18:45:15 CEST] <floholl> leave [23:03:31 CEST] <worried> hello [23:03:47 CEST] <worried> i need some help [23:04:11 CEST] <worried> i'm trying to record a restricted stream [23:05:18 CEST] <worried> it's detecting ffmpeg and blocking it [23:05:45 CEST] <worried> it works fine with ffplay [23:07:32 CEST] <c_14> What's it using to block it? User agent? Username/password? IP? [23:09:12 CEST] <worried> user agent block [23:09:42 CEST] <worried> i did specified a user agent, but it gives me 404 [23:09:51 CEST] <worried> but not with ffplay [23:10:15 CEST] <worried> so it detected ffmpeg and blocked it [23:10:52 CEST] <worried> how to make the stream thinks i'm using ffplay [23:11:12 CEST] <worried> so i can record [23:15:06 CEST] <c_14> ffmpeg and ffplay use the same libraries. They send the exact same http headers by default [23:17:48 CEST] <worried> then why the stream is working only with ffplay [23:18:08 CEST] <c_14> no clue [23:22:40 CEST] <worried> http://pastebin.com/ashHvbPX [23:22:58 CEST] <worried> stream is censored, as i'm not allowed to share it with general public [23:23:52 CEST] <c_14> user-agent should be an input option [23:23:56 CEST] <c_14> (before -i) [23:24:51 CEST] <worried> same error [23:29:08 CEST] <c_14> Can you pastebin the command and output with ffplay that works? [23:31:45 CEST] <worried> http://pastebin.com/d4mZD5EA [23:35:00 CEST] <c_14> Does just `ffmpeg -user-agent Lavf53.32.100 -i url' give the same 404? [23:35:23 CEST] <worried> yes [23:42:22 CEST] <c_14> No clue. Can't really think of anything they do differently that would cause that. [23:54:56 CEST] <jookiyaya> anybody know how extract subtitles .idx/.vob or .srt from MKV file [23:55:25 CEST] <jookiyaya> anybody know how extract subtitles .idx/.vob or .srt from MKV file (i know how to do it from dvd (video_ts)) [23:55:56 CEST] <c_14> ffmpeg -i mkv -c:s copy -map 0:s:0 out.srt [23:57:02 CEST] <jookiyaya> c_14 what about .idx/.vob [23:57:36 CEST] <c_14> Not entirely sure how ffmpeg handles that internally. Would need to see an ffprobe. [23:57:54 CEST] <jookiyaya> c_14 do you know the difference between .idx/.vob and .srt ? [23:58:31 CEST] <c_14> srt is a text subtitle format. idx/vob is picture [23:58:40 CEST] <jookiyaya> exactly [23:58:46 CEST] <jookiyaya> are you sure "ffmpeg -i mkv -c:s copy -map 0:s:0 out.srt" will work? since srt will require OCR [23:59:10 CEST] <c_14> That command only works for extracting srt subtitles from mkv. [23:59:31 CEST] <jookiyaya> let me try. i doubt it will work though [00:00:00 CEST] --- Sun Apr 12 2015

1 0

ffmpeg-devel.log.20150410
by burek 11 Apr '15

11 Apr '15

[00:07:13 CEST] <gnafu> llogan: no [00:11:57 CEST] <Timothy_Gu> Rewriting a fate server should be a gsoc or outreachy task [00:13:17 CEST] <llogan> gnafu: ok!!! [00:14:24 CEST] <llogan> Timothy_Gu: feel free to add it to some Wiki page. maybe a generic tasks ideas page would be use(ful|less) [00:15:42 CEST] <Timothy_Gu> llogan: no I already finished writing it [00:16:18 CEST] <llogan> ok. i don't pay attention to stuff apparently. [00:16:31 CEST] <Timothy_Gu> llogan: currently serving at http://fate.ffmpeg.org:8080/ (baptiste hasn't set up proper nginx stuff yet) [00:18:01 CEST] <Timothy_Gu> speaking of which [00:18:17 CEST] <llogan> Timothy_Gu: would you qualify as a student for these various projects? [00:19:51 CEST] <Timothy_Gu> llogan: nope (i'm too young) [00:20:11 CEST] <Timothy_Gu> (and I'm not female or transexual/genderqueer) [00:21:08 CEST] <nevcairiel> I think outreachy is just weird, isnt it a kind of discrimination to only accept certain kind of people =p [00:24:36 CEST] <llogan> looks nice so far. that's quite a brigt yellow though. maybe something like FFEE70 would be nice [00:43:17 CEST] <Prelude2004c> hello everyone [00:44:10 CEST] <Prelude2004c> still looking for a solution.. can someone help point me in the right direction.. i am trying to understand if there is something wrong with ffmpeg or something else. ... ffmpeg -i <1 input > -c:v output1 -c:v output 2, etc etc ... i get a max of 120fps [00:44:40 CEST] <Prelude2004c> if i run all 5 outputs i get only 120fps .. if i run 5 seperate ffmpeg instances i get full fps 60fps * 4 ( 300 fps ) [00:44:45 CEST] <Prelude2004c> does anyone know why thats possible [00:44:49 CEST] <kierank> yes probably an ffmpeg issue [00:45:02 CEST] <kierank> you've asked about 50 times now [00:46:21 CEST] <Prelude2004c> i know hopping someone new here will have an answer :) [00:46:39 CEST] <Prelude2004c> can anyone help disagnose it.. i am not a programmer so.. not sure how to fix it [00:46:43 CEST] <llogan> did you try ffmpeg-user mailing list? [00:47:11 CEST] <kierank> Prelude2004c: open a ticket [00:47:14 CEST] <jamrial> nevcairiel: i ask again in case you missed the question earlier today, are you ok with the libdcadec patch in the ml? [00:49:29 CEST] <baptiste> Timothy_Gu, sorry about the delay [01:01:37 CEST] <iive> Prelude2004c: most likely ffmpeg runs all streams in the same thread and you are not using encoder that is thread optimised. [01:02:19 CEST] <Daemon404> jamrial, is there no api bump needed for that dcadec flag? [01:02:26 CEST] <Daemon404> i mean teh dcadec ver requirement [01:03:37 CEST] <jamrial> no, it's been there since the beginning [01:03:50 CEST] <jamrial> also, the library has no version defines at all :p [01:03:57 CEST] <Daemon404> ah ok. [01:04:03 CEST] <Daemon404> lo lreally? [01:04:05 CEST] <Daemon404> not even in the .pc? [01:04:33 CEST] <jamrial> only an arbitrary 0.0.0 Timothy_Gu came up with [01:05:55 CEST] <jamrial> https://github.com/foo86/dcadec/issues/6 it's on the author's to-do list [01:06:24 CEST] <Daemon404> ic [01:06:39 CEST] <Daemon404> that foo86 guy really came out of nowhere [01:07:32 CEST] <Prelude2004c> isn't nvenc thread optomized ? [01:08:06 CEST] <Prelude2004c> can't be nvenc because i tried with just h264 cpus and it was fine too [01:08:10 CEST] <Prelude2004c> i mean sorry not fine [01:08:18 CEST] <Prelude2004c> so its not the encoder because 2 different ones do the same thing [01:11:17 CEST] <kierank> http://www.mremoteng.org/ [01:11:19 CEST] <kierank> this is quite good [01:12:36 CEST] <Daemon404> in inherently mistrust freeware [01:16:20 CEST] <kierank> ? [01:16:22 CEST] <kierank> it's open source [01:16:30 CEST] <kierank> https://github.com/rmcardle/mRemoteNG [01:17:03 CEST] <Daemon404> oh [01:17:11 CEST] <Daemon404> i clicked contribute and it asked me for bitcoins. [01:17:36 CEST] <Daemon404> apparently im an idiot and missed 'source code' [01:29:31 CEST] <iive> Prelude2004c: nvenc is quite recent, external and I remember something about 2 streams limitation. [01:36:28 CEST] <polygraphed> i have an xbmc machine running my television. but this isn't the right software for what i want. i want for the tv machine to represent its queue as a filesystem having files like 'currently_playing'. this way copying data to this host and telling it which data to play can be the same action. does any of this sound familiar? [01:44:22 CEST] <polygraphed> i guess what i'm asking is: can i do something like ffmpeg -i ./ -o /dev ? [01:51:36 CEST] <Timothy_Gu> nevcairiel: same [01:52:21 CEST] <Daemon404> Timothy_Gu, i am surprised you did not apply for gsoc [01:52:25 CEST] <Daemon404> surely you qualify? [01:52:43 CEST] <Timothy_Gu> llogan: looks good [01:57:23 CEST] Action: kierank wonders whether to resurrect hall of shame [01:59:39 CEST] <Timothy_Gu> Daemon404: no im too young [02:01:19 CEST] <Daemon404> o [02:01:37 CEST] <Timothy_Gu> post-secondary student developers ages 18 and older stipends im neither [02:54:52 CEST] <Prelude2004c> we got around with the 2 stream limit by using the k4000 series stuff [02:55:07 CEST] <Prelude2004c> that's not the issue. i think its a code issue on ffmpeg [03:02:07 CEST] <selsta> can someone help me writing to memory with avio_open() instead of to disk? do I need to do anything else apart from allocating my own AVIOContext? [03:02:43 CEST] <selsta> reading from memory with avformat_open_input() works fine [03:03:25 CEST] <Daemon404> selsta, you dont need to [03:03:29 CEST] <Daemon404> there isa built in func [03:03:57 CEST] <selsta> Daemon404: I dont need to do what exactly? [03:04:05 CEST] <Daemon404> http://ffmpeg.org/doxygen/trunk/avio_8h.html#adb5259ad07633518173eaa47fe657… [03:04:20 CEST] <selsta> oh nice [03:10:28 CEST] <selsta> is the useage then the same as with a file from disk? avformat_write_header() segvaults, even tho avio_open_dyn_buf returns no error [03:15:56 CEST] <selsta> found the error: I did not set ofmt_ctx->oformat [03:25:35 CEST] <selsta> yay, my program finally works. [03:27:45 CEST] <cone-803> ffmpeg 03Ferdinand Oeinck 07master:eff72a6c7375: libavcodec/hqx: multi threading support [03:38:30 CEST] <cone-803> ffmpeg 03James Almer 07master:3553b815f66c: avcodec/libdcadec: honor AVCodecContext bitexact flag [04:28:26 CEST] <Timothy_Gu> goddam nginx caching is FAST (on an SSD that is) [08:30:59 CEST] <Timothy_Gu> michaelni: your aarch64 machine is dead: http://104.131.148.213/report/aarch64-linux-qemu-ubuntu-gcc-4.8/20150409171… [09:44:49 CEST] <thardin> https://www.thebroadcastbridge.com/content/entry/2472/sky-and-ebu-endorse-n… what do you guys make of this? [09:45:16 CEST] <nevcairiel> is that the perseus thing? [09:45:22 CEST] <thardin> yes [09:46:05 CEST] <nevcairiel> i dont think anyone is going to oppose a new codec if it can deliver what it promises, but many companies made similar claims before...... :) [09:46:22 CEST] <j-b> and none delivered [09:46:37 CEST] <j-b> although, some good ideas in HEVC were not merged into the standard [09:47:20 CEST] <nevcairiel> from what I understand they were worried about increasing computational complexity too much? [09:49:37 CEST] <thardin> on2 comes to mind [09:50:19 CEST] <j-b> nevcairiel: sure, but someone could have picked some ideas [09:50:57 CEST] <thardin> the claim that mpeg2 stbs can decode it is a bit iffy [09:51:02 CEST] <nevcairiel> didnt perseus also claim that it decodes faster? [09:51:46 CEST] <kierank> you can find their patents if you look hard enough [09:52:00 CEST] <thardin> near the end of the article there's a bit that implies it may have explicit contrast/lightness coding (like CELT, but elsewise) [09:52:07 CEST] <nevcairiel> i only hope they also open the spec if it really comes to being used by big commercial media [09:52:24 CEST] <thardin> which daala is doing as well (I think) [09:56:47 CEST] <TimNich> I thought the word "novel" implied a work of fiction. [10:06:40 CEST] <TimNich> where do you stuble across these kierank ? [10:07:39 CEST] <kierank> TimNich: people telling me usually [10:09:12 CEST] <j-b> thardin: haha, that's a joke, right? [10:10:03 CEST] <thardin> dunno? [10:10:09 CEST] <j-b> the mpeg2 part [10:10:38 CEST] <thardin> that's what one of our sales people said they said at some convention [10:10:48 CEST] <kierank> http://companycheck.co.uk/company/07888208/V-NOVA-INTERNATIONAL-LTD/group-s… [10:11:00 CEST] <kierank> https://www.google.co.uk/search?q=LUCA+ROSSATO+patents&oq=LUCA+ROSSATO+pate… [10:11:01 CEST] <j-b> "can generate a single stream that incorporates different levels of encoding" [10:11:07 CEST] <kierank> not rocket science to find out what they are doing [10:11:23 CEST] <thardin> but that may have been a miscommunication [10:12:05 CEST] <kierank> s/a miscommunication/marketing [10:12:16 CEST] <kierank> don't let facts get in the way of good marketing [10:12:21 CEST] <kierank> wm4: ping [10:12:31 CEST] <saste> rcombs, I suppose your patch is supposed to help with mkv segmentation, right? [10:12:38 CEST] <saste> do you have a command to test here [10:12:55 CEST] <saste> something which doesn't work fine without the patch [10:14:02 CEST] <wm4> kierank: pong? [10:14:03 CEST] <rcombs> saste: ffmpeg -i <input> -f segment -segment_format matroska -segment_time 1 -segment_header_filename matroska_test/header.mkv -c copy matroska_test/segment-%05d.mkv [10:14:20 CEST] <kierank> wm4: so what are the priorities for api fate tests [10:14:27 CEST] <kierank> see the flac test the student wrote [10:15:04 CEST] <wm4> kierank: what do you mean? which APIs should be tested? [10:15:15 CEST] <kierank> which ones should be written first [10:15:22 CEST] <kierank> h264 imo [10:15:22 CEST] <rcombs> saste: I was just looking at your segment patches on the ML yesterday, and wondered what the use-case for the never-split mode was? [10:15:28 CEST] <wm4> I'll look at that patch closer once I finished writing my troll post [10:15:54 CEST] <saste> rcombs, my use case was to be able to split by size or by chapter [10:16:17 CEST] <saste> rcombs, i wanted to rip a CD using a single command, and putting each chapter/song in a single track [10:16:30 CEST] <saste> without to rely on scripting [10:17:08 CEST] <saste> the libcdio demuxer treats each track as a separate chapter [10:17:16 CEST] <rcombs> ah, interesting [10:17:24 CEST] <rcombs> I saw that patch as well but didn't make the connection [10:22:46 CEST] <saste> rcombs, the logic behind the never-split mode is: with -segment_expr I'm using an expression to detect when to split, and I don't want it to conflict with the -segment_time option [10:22:57 CEST] <saste> so I need some way to disable -segment_time from splitting [10:23:21 CEST] <saste> the problem is that it breaks backward compatibility, so it's a bit controversial (that's why I didn't commit it yet) [10:23:50 CEST] <ubitux> damn i'm missing the drama around libquvi [10:23:57 CEST] <saste> rcombs, about your patch, how do I use segments and header to get a playable file? [10:23:57 CEST] <ubitux> even though i'm the author of it... [10:23:59 CEST] <ubitux> sorry guys. [10:24:01 CEST] <ubitux> :D [10:24:13 CEST] <saste> ubitux, why libquvi is still working? [10:24:30 CEST] <ubitux> i dunno, i can't build it anymore anyway [10:24:43 CEST] <rcombs> saste: concatenate the header and any consecutive sequence of segments [10:24:44 CEST] <saste> even now that bigG disabled most downloader plugins? [10:24:53 CEST] <ubitux> since my distro is distributing 0.9 exclusively [10:25:01 CEST] <ubitux> and ffmpeg support the 0.4 (before license change) [10:25:05 CEST] <ubitux> (iirc) [10:25:27 CEST] <ubitux> i'll send a patch to drop libquvi support when i'm back [10:25:39 CEST] <nevcairiel> ooh more drama! [10:25:41 CEST] <rcombs> saste: any particular reason to change the default? [10:25:45 CEST] <ubitux> (in about 10 days, wait for 2 weeks) [10:25:48 CEST] <rcombs> saste: other than that, looks good [10:27:24 CEST] <saste> rcombs, suppose that i want to use an expression to split, now I have to disable the segment_time, right now this is impossible [10:27:38 CEST] <saste> having a default of "never split" helps with -segment_expr [10:27:56 CEST] <nevcairiel> that doesnt have to be the default though [10:28:00 CEST] <saste> also it's not very intuitive to have an arbitrary value of 2 seconds as default [10:28:27 CEST] <rcombs> saste: I'd have something like `if (segment_expr) seg->time = -1;` [10:28:31 CEST] <rcombs> saste: yeah, that is pretty weird [10:29:14 CEST] <rcombs> I don't have a strong opinion either way, since the default is probably not used much [10:29:18 CEST] <saste> rcombs, anyway, I'm testing your patch and seems sane, but I think we must work some documentation so people know how to use it, and which use case it covers [10:29:21 CEST] <saste> now i'll review the code [10:29:28 CEST] <rcombs> cool, thanks [10:30:47 CEST] <rcombs> oh, or just put `if (seg->segment_expr)` first [10:32:32 CEST] <saste> rcombs, but, I also want to allow an OR logic, so if I specify -segment_time 10 -segment_expr EXPR the muxer will split IF segment time is 10 OR the expression is true [10:32:49 CEST] <rcombs> ah, interesting [10:33:11 CEST] <nevcairiel> shove that logic into your expression? =P [10:33:28 CEST] <saste> nevcairiel, not very user friendly, but yes that's a possibility [10:34:58 CEST] <rcombs> saste: so, the patch doesn't do that now, but could in the future? [10:35:17 CEST] <saste> rcombs, what patch? [10:35:31 CEST] <rcombs> yours [10:35:32 CEST] <rcombs> [WIP] add support to segmentation expression [10:35:33 CEST] <saste> rcombs, with the complete patchset it will work as I described [10:36:01 CEST] <saste> the other thing which I'm not sure about is the copy chapter metadata to segment [10:36:12 CEST] <saste> probably a dedicated option should be used instead [10:36:51 CEST] <rcombs> yeah, an option would be good there [10:39:29 CEST] <rcombs> looks to me like segment_time is ignored when segment_expr is present? [10:41:41 CEST] Action: rcombs removes the earwax^Hlow-pass filter from his earbuds [11:00:04 CEST] <ubitux> michaelni: i'll review the sami/srt thing patches, please don't apply before i review [11:07:38 CEST] <rcombs> wm4: well said, re: ML post [11:57:48 CEST] <wm4> kierank: so, uh, you have no clear idea yet how exactly API tests should work? [11:58:03 CEST] <kierank> well yes I want them to go into fate [11:58:07 CEST] <kierank> and produce the same framecrc stuff [11:58:09 CEST] <kierank> api_h264 [11:58:19 CEST] <kierank> api_flac [11:58:19 CEST] <kierank> etc [11:58:24 CEST] <wm4> ok [11:58:43 CEST] <kierank> I think it's ok for the student to reuse framecrc [11:59:28 CEST] <kierank> wm4: but I would personally say h264 is the one where the student should start [12:00:00 CEST] <kierank> I also guess the codec could be passed as a command line or something [12:00:03 CEST] <kierank> ./api_test h264 [12:00:05 CEST] <kierank> or whatever [12:00:57 CEST] <wm4> kierank: the student is in hurry to finish the task tomorrow? [12:01:02 CEST] <wm4> or even today [12:01:05 CEST] <wm4> crazy deadline [12:01:13 CEST] <kierank> no i don't think so [12:01:20 CEST] <kierank> they can continue until the 27th afaik [12:01:24 CEST] <kierank> because 27th is the due date [12:01:26 CEST] <wm4> oh, good then [12:01:49 CEST] <wm4> was that another student yesterday or am I just making shit up? [12:01:53 CEST] <kierank> the capability to just write a simple test with the api is more than all the applicants [12:02:02 CEST] <kierank> wm4: name? [12:02:57 CEST] <wm4> vibr [12:03:07 CEST] <kierank> haven't heard from him/her [12:03:42 CEST] <loki_> what you are taking about? what students? there is unvirsity of ffmpeg or something? [12:05:08 CEST] <kierank> gsoc and outreachy [12:06:28 CEST] <wm4> I already feel sorry for the student [12:06:35 CEST] <wm4> the API examples are somewhat buggy [12:08:03 CEST] <wm4> and I wish claws-mail would stop corrupting its own database [12:50:40 CEST] <cone-166> ffmpeg 03Michael Niedermayer 07master:599dc8fee18f: avcodec/hqx: Use av_clip_uintp2() [13:13:12 CEST] <cone-166> ffmpeg 03Shivraj Patil 07master:8af3ce53786d: configure: add support for mips32r5, p5600 cpu and msa [13:13:13 CEST] <cone-166> ffmpeg 03Shivraj Patil 07master:076bfe96128f: configure: add support for mips64r6 and i6400 cpu [13:13:14 CEST] <cone-166> ffmpeg 03Shivraj Patil 07master:7fdd31421cfe: configure: add support for 74kf cpu [13:13:15 CEST] <cone-166> ffmpeg 03Shivraj Patil 07master:578d99e7c6d1: avutil/mips/intreadwrite: build fix for mips64r6 (instruction 'lwl' not supported) [13:50:32 CEST] <cone-166> ffmpeg 03Carl Eugen Hoyos 07master:7b39d853b88e: lavf/flac: Autodetect raw flac files. [13:50:33 CEST] <cone-166> ffmpeg 03Michael Niedermayer 07master:5d0f836f6270: Merge remote-tracking branch 'cehoyos/master' [14:26:25 CEST] <cone-166> ffmpeg 03Himangi Saraogi 07master:aae9f52c4efa: avformat/rtsp: Fix unchecked return value [14:53:48 CEST] <cone-166> ffmpeg 03Himangi Saraogi 07master:8d15de7eb265: ffmdec: Check return value of ffm_append_recommended_configuration [17:41:19 CEST] <kierank> "I for one wish we had this functionality in FFmpeg [at least basic HTTP server]." [17:42:45 CEST] <BtbN> Throw it into a cgi script and invoke it via apache/lighttpd. ;) [17:43:02 CEST] <Daemon404> NO WE NEED IT IN LIBAVGRABGEDUMP [17:43:06 CEST] <Daemon404> garbage* [17:43:32 CEST] <BtbN> Hm, libavmisc wouldn't be that bad for stuff like this. [17:44:02 CEST] <michaelni> BtbN, how should hls and http client code be tested ? [17:44:35 CEST] <BtbN> http client is perfectly fine like it is [17:45:16 CEST] <michaelni> yes but its being tested in fate [17:45:23 CEST] <michaelni> its NOT being tested [17:45:30 CEST] <wm4> the holy scriptures demand that all be in a single repo, with libavcodec, libavformat, libswscale (... skipping to paragraph 10 ...), and no other repos or libs shall exist, for the libs for holy [17:45:35 CEST] <wm4> s/for/are [17:46:39 CEST] <BtbN> I'm confused [17:47:04 CEST] <kierank> anyway if ffmpeg gsoc involves a http server I think I will fork [17:47:47 CEST] <nevcairiel> that is one of the tasks which a student works on [17:48:07 CEST] <kierank> I am aware [17:48:14 CEST] <kierank> "I for one wish we had this functionality in FFmpeg [at least basic HTTP server]." is from melange [17:49:18 CEST] <BBB> its not worth forking over [17:49:21 CEST] <BBB> just make it disableable [17:49:28 CEST] <BBB> you compile with --disable-httpserver [17:49:34 CEST] <BBB> or the people that want it compile with --enable-xyz [17:49:36 CEST] <BBB> and were all happy [17:49:42 CEST] <kierank> yes but it's bloat [17:49:58 CEST] <kierank> and a waste of gsoc resources [17:52:36 CEST] <BBB> I dont think we can tell other people what they should like to work on [17:52:42 CEST] <BBB> if people want a http server, so be it [17:53:03 CEST] <BBB> some people might claim hevc or vp9 is a waste of time [17:53:14 CEST] <BBB> (depending on where in the political spectrum you lie) [17:53:18 CEST] <kierank> they can work on it but should it really be merged... [17:53:36 CEST] <Daemon404> i gotta say, i dont think "commit ALL the garabge as logn as you can disable it" is a pretty stupid ideal. [17:53:43 CEST] <BBB> anyone here familiar with cocoa? [17:54:01 CEST] <BBB> Daemon404: dont, or do? [17:54:04 CEST] <BBB> sounds like you do [17:54:10 CEST] <Daemon404> er [17:54:18 CEST] <Daemon404> yeah i think it is a stupid ideal. [17:54:23 CEST] <BBB> hehe gotcha [17:59:16 CEST] <wm4> <kierank> and a waste of gsoc resources <- weren't these paid student things always waste of resources until now? [17:59:23 CEST] <wm4> at least according to Daemon404's judgement [18:00:15 CEST] <Daemon404> it worked once or twic ein 10 years [18:05:40 CEST] <BBB> I think it worked several times, but utsually succes is predicted by the student being a prior contributor to ffmpeg, or a prior gsoc participant [18:05:45 CEST] <BBB> which isnt a good thing, I guess [18:05:49 CEST] <BBB> but weve had several good ones [18:06:04 CEST] <Daemon404> BBB, well i mean success as in attracted new contribs [18:06:05 CEST] <BBB> and its not nice to say all their work is wasted, theres a lot of good stuff coming out of it [18:06:08 CEST] <Daemon404> not "project passed" [18:06:12 CEST] <BBB> hm, fair enough [18:08:27 CEST] <BBB> where do I find cocoa experts [18:08:53 CEST] <gnafu> BBB: Switzerland? [18:09:02 CEST] <Daemon404> he means hipster cocoa [18:09:12 CEST] <Daemon404> so probably up their own butts in the valley [18:09:29 CEST] <kierank> BBB: j-b probably knows some [18:09:38 CEST] <gnafu> Daemon404: I am always so delighted with your abundant charm. [18:09:39 CEST] <gnafu> ;-D [18:10:10 CEST] <BBB> oo good idea [18:10:10 CEST] <BBB> j-b: pokey! [18:10:31 CEST] <BBB> he was right though [18:10:38 CEST] <BBB> its indeed hipster-cocoa Im having issues with [18:11:16 CEST] <j-b> yes [18:12:09 CEST] <BBB> j-b: are you familiar with autolayout? [18:12:37 CEST] <j-b> yes [18:12:44 CEST] <BBB> wanna help me debug a really odd issue? [18:12:56 CEST] <BBB> I can push my test code to github if that helps [18:13:00 CEST] <BBB> I can also pastebin it [18:13:04 CEST] <BBB> whichever is easier for you [18:13:09 CEST] <j-b> now, really not [18:13:12 CEST] <j-b> but query feepk [18:13:22 CEST] <BBB> -enothere [18:15:14 CEST] <j-b> query :) [18:15:22 CEST] <BBB> yup, working on it [18:15:39 CEST] <BBB> hes away I think, Ill wait for him to get back [18:26:31 CEST] <cone-166> ffmpeg 03Michael Niedermayer 07master:4d0f6d3fb421: avdevice/vfwcap: revert header reordering from c201069fac9a76e6604f9d84d76a172434d62200 [18:57:36 CEST] <cone-166> ffmpeg 03Michael Niedermayer 07master:bc48c88918f7: avcodec/h264: Do not fail with randomly truncated VUIs [20:23:53 CEST] <BBB> Timothy_Gu: why does the cabac test need a REF? [20:24:10 CEST] <BBB> isnt the idea that these tests are refless and pass purely based on exit code? [20:25:49 CEST] <Timothy_Gu> BBB: that's why REF=/dev/null [20:26:02 CEST] <BBB> I meant, why do we need that line at all [20:26:03 CEST] <Timothy_Gu> not sure if it is needed but everything else has it [20:26:13 CEST] <BBB> I wonder if we can remove it? [20:26:25 CEST] <BBB> (Im not a makefile expert) [20:26:32 CEST] <BBB> nor do I care much, I guess [20:26:43 CEST] <BBB> and +100 for putting that test in fate, thank you [20:27:22 CEST] <Timothy_Gu> BBB: seems like it's not (at least it's not used in run() in fate-run.sh) [20:27:46 CEST] <Timothy_Gu> np [20:51:45 CEST] <Timothy_Gu> jamrial: so do you want me to remove the timer stuff? it doesn't matter for me either way [20:52:09 CEST] <jamrial> if nobody complains yes, in a separate patch [20:52:24 CEST] <Timothy_Gu> is this patch good? [20:58:15 CEST] <Timothy_Gu> jamrial: ^^ [20:58:58 CEST] <jamrial> Timothy_Gu: yeah, the first two are good [20:59:52 CEST] <jamrial> comment the timer lines out instead of removing them, so people can easily uncomment them if they want to benchmark the cabac code [21:04:07 CEST] <Timothy_Gu> jamrial: i'll do it after these two are applied if you don't mind. nobody likes merge conflicts. [21:04:17 CEST] <jamrial> sure [21:48:12 CEST] <cone-166> ffmpeg 03James Almer 07master:61090db29a3e: avcodec/libx265: print supported presets and tunes on error [21:57:08 CEST] <cone-166> ffmpeg 03Timothy Gu 07master:744594685e81: cabac-test: Return 1 if there are any errors [21:57:09 CEST] <cone-166> ffmpeg 03Timothy Gu 07master:28e2bf90b93e: Add cabac test into fate [22:29:05 CEST] <cone-166> ffmpeg 03Timothy Gu 07master:1a562adb0101: tests: Do not include stdout/stderr or diff if the test passed [22:34:04 CEST] <wm4> who has write access to the fate samples? [22:56:19 CEST] <Daemon404> wm4, why? [22:56:26 CEST] <wm4> just curious [22:56:48 CEST] <wm4> the samples are not in any way "publicly" managed, unlike a source code repo [22:57:07 CEST] <Daemon404> i dont know hwo ffmpeg does it [22:57:11 CEST] <Daemon404> but at libav i scp'd it once [22:57:15 CEST] <Daemon404> (and long forgoten how to) [22:57:39 CEST] <Daemon404> er sftp [22:59:53 CEST] <rcombs> maybe this is a job for git LFS [23:00:16 CEST] <rcombs> now that that's a thing [23:02:06 CEST] <Daemon404> its not even special [23:02:13 CEST] <Daemon404> it's like git annex but less open [23:02:19 CEST] <Daemon404> and rsync is more than good enough [23:02:27 CEST] Action: Daemon404 doesnt get the "put everything in git" obsession [23:54:19 CEST] <BBB> Daemon404: I think its about the ability to branch and see changelogs [23:54:39 CEST] <BBB> Daemon404: e.g. who uploaded that file or what version of the samples do we use for the 0.x release branch [23:54:49 CEST] <Daemon404> hmm perhaps [23:54:49 CEST] <BBB> (you could even branch/tag the samples collection for a release branch) [23:54:54 CEST] <BBB> rsync doesnt do that [23:55:08 CEST] <Daemon404> im still unconvinced by git LFS though [23:55:11 CEST] <BBB> its a little complicated but theres some okish reasons for it [23:55:14 CEST] <Daemon404> like i said, its like git annex [23:55:16 CEST] <Daemon404> but more closed [23:55:42 CEST] <BBB> Im not advocating for any specific implementation, but the idea of version controlling samples is not terrible [23:56:05 CEST] <Daemon404> right. [00:00:00 CEST] --- Sat Apr 11 2015

1 0

ffmpeg.log.20150410
by burek 11 Apr '15

11 Apr '15

[00:00:41 CEST] <DragonsLord> I asked this because I still have a black line on the left [00:00:47 CEST] <DragonsLord> and I am not able to crop it [00:02:17 CEST] <justinX> are there way to encode in a video that its size is some pixels smaller than the data really is? I mean so it crops the video say 3 pixel left and 7 bottom while playing, without having to change video data itself. [00:02:59 CEST] <justinX> perhaps something in the container format or such... [00:03:50 CEST] <llogan> justinX: ffplay -vf crop=w:h:x:y input [00:03:53 CEST] <justinX> for all the people that thinks multiple is 8 is booring :-D [00:03:56 CEST] <justinX> ok nice! [00:04:41 CEST] <llogan> ffplay lets you take a crop [00:04:50 CEST] <llogan> on your video [00:05:21 CEST] <DragonsLord> llogan, it seems to be not working for me :( [00:05:21 CEST] <justinX> but ffplay? that would only be when you play the video with that right? I want something that tells every player (vlc, webpages etc) how to crop the video when they play it. [00:05:51 CEST] <llogan> justinX: oh. i don't know. [00:06:29 CEST] <llogan> DragonsLord: sorry, but i'm not sure what you're doing exactly. [00:08:01 CEST] <DragonsLord> llogan, crop=w=iw-118:h=ih-4:x=52:y=4,scale=w=640:h=480:flags=bilinear [00:09:28 CEST] <DragonsLord> I also received this "Warning: data is not aligned! This can lead to a speedloss" [00:10:12 CEST] <DragonsLord> at the end [00:10:13 CEST] <DragonsLord> I still have a black line on the left [00:10:18 CEST] <DragonsLord> and I am not able to crop it [00:12:53 CEST] <DragonsLord> :( [00:13:36 CEST] <llogan> you can ignore that warning [00:14:39 CEST] <sg90> Adding to my previous post regarding segments and timecode, I've actually noticed the second segment at t=30 doesnt have the correct timecode, its supposed to be 00:00:30:00 but it is just an int - 42384379 - and never increments [00:15:31 CEST] <DragonsLord> llogan, OK but ... at the end I continue having a black line of about 2 pixels on the left [00:15:33 CEST] <DragonsLord> :( [03:23:04 CEST] <jookiyaya> is there big difference between AC3 and DTS? [04:59:51 CEST] <noobert> http://i.imgur.com/279K7DC.png [04:59:56 CEST] <noobert> pls halp [05:40:04 CEST] <pentanol> noobert could you make grep -ri nvEncode ./ in yours ffmpeg sources folder [05:40:19 CEST] <brotherBox> he's gone [05:41:20 CEST] <pentanol> really, thanks [05:41:38 CEST] <relaxed> if he comes back he needed --extra-cflags, and not cxxflags [12:51:32 CEST] <sybariten> uh ok hai [12:52:15 CEST] <sybariten> would anyone be so kind to give me a short crash course in what presets are, where ffmpeg on win expects to find them, and how i should refer to them on the command line [12:53:02 CEST] <sybariten> I'm suspecting this is all being complicated by the fact that i'm trying to run ffmpeg from bash, also ... should maybe go with cmd32 [12:54:52 CEST] <c_14> sybariten: do you mean the ffpresets/avpresets? [12:55:23 CEST] <sybariten> c_14: yes i think so. [12:56:00 CEST] <sybariten> "file for preset 'normal' not found" [12:56:07 CEST] <c_14> https://ffmpeg.org/ffmpeg.html#Preset-files [13:01:38 CEST] <sybariten> i can do a hard refer using -fpre but then i get another error: invalid option or argument: directpred=3 [13:04:05 CEST] <c_14> That doesn't look like a valid ffmpeg option to me. (isn't listed in ffmpeg -h full) [13:04:50 CEST] <sybariten> ok [13:04:57 CEST] <sybariten> i tried a simpler command lnie now anyhow [13:05:05 CEST] <sybariten> but fuck if i dont get other errors... sigh [13:15:38 CEST] <sybariten> " -sameq was removed , use -qscale 0 " [14:52:15 CEST] <Naminor> Hello there [14:52:33 CEST] <Naminor> I'm facing an issue with ffmpeg. I'll explain the situation [14:53:46 CEST] <Naminor> We need for our client to make screencast of webpages. So we decided to create servers (Ubuntu) put ffmpeg on it and with an API system call the ffmpeg command to screencast all the screen and the audio system [14:54:09 CEST] <Naminor> so it records the web page and the sound of the webpage. [14:54:46 CEST] <Naminor> Also ffmpeg is used to both stream and record locally the screencast. [14:55:20 CEST] <Naminor> But we are facing random audio/video desync [14:55:38 CEST] <Naminor> I tried several parameters and even power upthe serversz [14:55:43 CEST] <Naminor> servers* [14:56:08 CEST] <Naminor> we are now with a 6 Ghz Quad Core Xeon for one screencast agent [14:56:19 CEST] <Naminor> and some time, it is out of sync [14:56:31 CEST] <Naminor> could you help me with this problem ? [15:00:46 CEST] <Venti> check that the liquid nitrogen hasn't run out, because 6GHz is kind of a lot [15:06:41 CEST] <Naminor> it actualy run on a vmware esx [15:06:50 CEST] <Naminor> we rent these servers [15:07:35 CEST] <Naminor> an on full charge it consumed about 6400MHz of CPU [15:20:59 CEST] <Naminor> here is my ffmpeg command if you want to take a look: ffmpeg -f x11grab -s 1280x720 -r 25 -i :0+0,0 -f pulse -i default -c:v h264 -vb 1024k -preset ultrafast -tune zerolatency -pix_fmt yuv420p -async 1 -acodec libmp3lame -ar 22050 -ac 2 -ab 50k -r 25 -f flv 'rtmp://XXXXXXXXXXXX/live/XXXXXXX' [15:21:30 CEST] <Naminor> and ffmpeg version 1.2.6-7:1.2.6-1~trusty1 [15:23:46 CEST] <Naminor> any idea ? [15:24:35 CEST] <c_14> Naminor: what's the first line of the output of `ffmpeg -version' ? [15:24:57 CEST] <Naminor> ffmpeg version 1.2.6-7:1.2.6-1~trusty1 [15:26:20 CEST] <c_14> First of all, you should probably update your version of ffmpeg. That version is very old. [15:26:45 CEST] <c_14> Second, have you tried without -async 1? [15:27:20 CEST] <Naminor> I will try without -async 1 [15:27:41 CEST] <Naminor> yeah but we are on ubuntu and there are no recent repository (or should i compile it from source ?) [15:29:23 CEST] <c_14> If you can compile from source, please do. There have been lots of bug fixes and general improvements since 1.2.6. [15:30:59 CEST] <Naminor> Okay [15:31:25 CEST] <Naminor> i've removed async, it's a little better but the desync happen later [15:33:39 CEST] <c_14> Other things you could try are changing the first -r to -framerate. And if that doesn't help adding -af aresample=async=1000 [15:37:59 CEST] <Naminor> it is worse with framerate so i add -af aresample=async=1000 and keep framerate or do i rollback to -r ? [15:38:32 CEST] <c_14> It's worse with -framerate? Weird. Try both. [15:38:41 CEST] <c_14> Updating will probably be your best bet though. [15:38:49 CEST] <Naminor> framerate + -af ? [15:39:02 CEST] <c_14> Both -framreate and -af and -r and -af [15:39:06 CEST] <c_14> *framerate [15:45:39 CEST] <Naminor> no it is worse in the both cases [15:48:15 CEST] <Naminor> i think i'm gonna go with the sources but i never did it before, i cloned the git so i just have to compile it with gcc ? [15:49:22 CEST] <c_14> https://trac.ffmpeg.org/wiki/CompilationGuide/Generic https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu [15:49:45 CEST] <Naminor> thank you ! [16:40:49 CEST] <karika200> hi [16:41:09 CEST] <karika200> I'm restreaming an rtsp source to rtmp push with ffmpeg [16:41:20 CEST] <karika200> in the rtsp source i have sometimes broken frames i think [16:41:31 CEST] <karika200> how can i drop this corrupted frames with ffmpeg? [16:43:29 CEST] <synthecypher> Is there a quick way to split a video into 30 second chunks? [16:44:59 CEST] <karika200> ffmpeg -ss 00:00:00 -i video.mp4 -to 00:30:00 -c copy cut.mp4 ? [16:45:24 CEST] <synthecypher> will take take a 3 minute video and split it into 30 chunks? [16:45:31 CEST] <synthecypher> or just the first 30 seconds? [16:46:10 CEST] <karika200> just first 30, but with minimal scripting you can step forward the whole video [16:47:40 CEST] <Prelude2004c> join #ffmpeg-dev [16:47:53 CEST] <Prelude2004c> oops [16:48:06 CEST] <c_14> synthecypher: look at the segment muxer [16:48:15 CEST] <synthecypher> c_14: kk [16:48:16 CEST] <Prelude2004c> :) good morning everyone.. using up a new nvenc with ffmpeg.. now they changed form fast , very fast, etc etc to hq hp , etc [16:48:34 CEST] <Prelude2004c> anyone know a link that tells me which is considered slow or normal or veryfast etc ? [16:48:41 CEST] <Prelude2004c> should be comparable right? [16:58:50 CEST] <synthecypher> c_14: I'm using this "ffmpeg -i scenarios.mp4 -codec copy -map 0 -f segment -segment_times 30,60,90,120,150,180 out%03d.mp4" the first segment is fine but the others don't play. [17:02:38 CEST] <c_14> Just tested on my system with an mp4 I had lying around. Works perfectly for me. What version are you running? [17:19:22 CEST] <synthecypher> c_14: Are all of the segments playable in a player like VLC? [17:22:38 CEST] <c_14> I used mpv, but yes they're all playable. [17:22:42 CEST] <Timster> Hey, guys. I have some trouble with concatenating files together - there are files recorded on two different computers, but reencoded with the same software again. Media Info shows, that everything is exactly the same but still, when there is a switch between files of different origins I get choppy video, artifacts and glitches - audio seems just fine. Any idea what this could be? [17:23:00 CEST] <synthecypher> c_14: must be something wrong with my mp3 [17:23:02 CEST] <synthecypher> c_14: must be something wrong with my mp4* [17:23:16 CEST] <Timster> Sure, c_14 - one sec. [17:26:00 CEST] <Timster> http://pastebin.com/4ws0QKR4 [17:27:32 CEST] <Timster> Anything wrong there? Need a moment to change the setup to replicate the error again. [17:27:32 CEST] <c_14> Are you sure all the input files have the same settings? Can you pastebin an ffprobe of all of them? [17:27:57 CEST] <Timster> Of course, if you tell me how I use ffprobe on all flv's in a folder [17:28:00 CEST] <Timster> Well [17:28:25 CEST] <Timster> The only difference I am aware of is, before the reencode with the software, that one origin uses bilinear filtering, the others lanczos [17:28:56 CEST] <c_14> for file in dir/*flv; do ffprobe $file; done |& curl -F 'sprunge=<-' http://sprunge.us [17:29:02 CEST] <c_14> Or redirect into a file. [17:29:44 CEST] <Timster> Could you tweak it so it does log into a file, please? [17:29:58 CEST] <c_14> for file in dir/*flv; do ffprobe $file; done > ffprobe_logs [17:30:06 CEST] <c_14> eh [17:30:06 CEST] <Timster> Thanks, one sec [17:30:09 CEST] <c_14> &> [17:30:23 CEST] <c_14> Though the usefull info is on stdout anyway iirc. [17:30:35 CEST] <Timster> for file in dir/*flv; do ffprobe $file; done &> ffprobe_logs [17:30:37 CEST] <Timster> ? [17:31:34 CEST] <Timster> May I query you about this? [17:32:06 CEST] <c_14> ye [17:55:32 CEST] <Prelude2004c> hey, with a quadro card... Gpu : 8 % [17:55:32 CEST] <Prelude2004c> Memory : 2 % [17:55:32 CEST] <Prelude2004c> Encoder : 60 % [17:55:32 CEST] <Prelude2004c> Decoder : 0 % [17:55:46 CEST] <Prelude2004c> any way to use up the GPU too as well as the nvenc hardware ? [17:56:02 CEST] <Prelude2004c> wonder if there is some way to increase encoding process to GPU as well as the new hardware accel [18:01:02 CEST] <BtbN> Prelude2004c, the nvenc encoder is a dedicated piece of hardware on the cards. The GPU is not involved at all. [18:14:14 CEST] <Scorp1us> Hi all, I was wondering if there was a way to use ffmpeg to divide a video file up inso segments based on I and P frames? [18:15:00 CEST] <Scorp1us> Example Video= [IPPPPIPPPIPPP] -> [IPPP], [IPPP], [IPPP] [18:16:43 CEST] <loki_> Prelude2004c: for what you need utilize gpu core? [18:17:36 CEST] <loki_> Prelude2004c: and in what terminal program you check your video card utilization? [18:52:25 CEST] <crossf> hi guys, could you please help me to understand this error message: Failed to configure output pad on Parsed_concat_0 [18:53:14 CEST] <spaam> if you show us the commandline you trying to use. we might can help you. [18:54:56 CEST] <crossf> hmmm, not sure how to extract it from this wrapper c#..., I have tried with native ffmpeg and args and I had other problems, let me paste it for you [18:57:09 CEST] <crossf> my args: " -i Intro_31bf475c-ba8f-48de-bd59-83197fbcdea5 -i BodyPart_da822a64-2941-4317-b111-1c2c2a5ecd3f -filter_complex 'movie=Intro_31bf475c-ba8f-48de-bd59-83197fbcdea5, scale=1920:1080, setsar=1:1 [v1] ; movie=BodyPart_da822a64-2941-4317-b111-1c2c2a5ecd3f, scale=1920:1080, setsar=1:1 [v2] [v1] [v2] concat [outv]' -map '[outv]' intvid74127b8a-df2f-46ea-b818-302e25d6e574-c5a447a5-f42e-4475- [18:57:10 CEST] <crossf> 8138-9d26e3786673" [18:57:39 CEST] <crossf> Stream specifier ':1' in filtergraph description [0:0] [1:1] concat=n=2:v=1:a=1 [v] [a] matches no streams. [18:58:06 CEST] <crossf> basically I would like to concat videos [18:59:17 CEST] <crossf> guid are not really readable sorry [19:00:35 CEST] <crossf> spaam is that help? [19:06:07 CEST] <debianuser> crossf: It seems you miss a comma between "setsar" and "concat" filters in the "... setsar=1:1 [v2], [v1] [v2] concat ..." part. [19:06:20 CEST] Action: debianuser isn't sure about movie=... syntax [19:11:54 CEST] <c_14> semicolon [19:11:56 CEST] <c_14> not comma [19:12:15 CEST] <c_14> Like so [v2];[v1][v2] [19:16:38 CEST] <jehar_work> Heyo [19:16:55 CEST] <jehar_work> any map_channel gurus around today? [19:17:04 CEST] <jehar_work> Specifically, prores with 16 channels on a single audio track (fucking blackmagic), pulled out a wav, similarly with 16 channels on one track. [19:17:13 CEST] <jehar_work> However, any attempt to use map_channel 0.0.1 or whatever to produce a single-channel wav just gives me a blank file, of the correct length. [19:17:20 CEST] <jehar_work> I know I'm accessing the correct channels, because 0.0.0-0.0.15 are fine, but anything above 15 is rejected [19:27:29 CEST] <Roberth1990> I see that chrome uses ffmpeg, do they ship it builtin? [19:28:32 CEST] <jehar_work> Full outputs from my tests: http://pastebin.com/z3HaHQM4 [19:30:11 CEST] <llogan> your ffmpeg build is old [19:30:58 CEST] <jehar_work> Granted, will compare with a newer build [19:31:42 CEST] <llogan> http://johnvansickle.com/ffmpeg/ [19:32:43 CEST] <kepstin-laptop> Roberth1990: yes, chrome builds a custom copy of ffmpeg included with the browser. I dunno the details of how they link it. [19:33:24 CEST] <Roberth1990> kepstin-laptop: well that doesn't matter I just began to wonder why chrome continued to work without restart after going from libav to ffmpeg [19:50:31 CEST] <debianuser> jehar_work: according to https://trac.ffmpeg.org/wiki/AudioChannelManipulation it should work. Have you tried it on that 16ch .wav file, e.g. `ffmpeg -i all.wav -map_channel 0.0.0 1.wav` ? [19:55:03 CEST] <kyleogrg> I am converting a dvd vob file (non square pixels) to a 16:9 mkv video. For -vf I (bascially) have yadif,mcdeint,framestep,crop black border,scale to square pixels,scale to resize. Is this the right order of video filters? [19:57:59 CEST] <RobotsOnDrugs> looks reasonable, though i wouldn't bother with squaring the pixels myself [19:58:43 CEST] <jehar_work> debianuser, llogan was correct. the 2013 build still had those bugs. [19:59:32 CEST] <kyleogrg> ok [21:20:40 CEST] <blert> Is it possible to use ffmpeg with Cisco's OpenH264 binaries? I'd like to be able to drop the binary DLL into a folder and have ffmpeg recognize the binary, load it, and use it for decoding and encoding h264 streams. Thanks. [21:21:23 CEST] <klaxa> i don't think openh264 does encoding [21:21:50 CEST] <klaxa> oh wait [21:21:53 CEST] <llogan> --enable-libopenh264 enable H.264 encoding via OpenH264 [no] [21:21:57 CEST] <klaxa> yes [21:22:45 CEST] <blert> So, does it work "out of the box"? Or do I need to rebuild ffmpeg with those flags enabled? (btw, thanks for th quick response!) [21:23:24 CEST] <klaxa> you will probably have to compile it, since it is disabled by default [21:23:27 CEST] <blert> (by "out of the box", i mean the pre-built ffmpeg binaries.) [21:23:34 CEST] <blert> ok. thanks! [21:24:48 CEST] <blert> this may be OT: does anyone here know what the difference is between ffmpeg and Chrome's "ffmpegsumo" dll? [21:30:38 CEST] <BtbN> It's every libav* library in one binary iirc [21:31:08 CEST] <blert> oh. that makes sense. thanks!! [22:08:18 CEST] <DragonsLord> Hello to everybody ... I have a cropping problem [22:08:23 CEST] <DragonsLord> May someone help me? [22:13:59 CEST] <ChocolateArmpits> DragonsLord: I can help [22:14:09 CEST] <DragonsLord> Thank you so much, ChocolateArmpits [22:14:36 CEST] <ChocolateArmpits> DragonsLord: I'm all ears [22:14:50 CEST] <DragonsLord> I am proceeding with Pass-1 [22:15:01 CEST] <DragonsLord> I will pastebin the log [22:17:03 CEST] <DragonsLord> Report is too big [22:17:03 CEST] <DragonsLord> :| [22:17:35 CEST] <ChocolateArmpits> Can you possibly trim it? [22:18:47 CEST] <DragonsLord> What could I delete? [22:19:16 CEST] <ChocolateArmpits> What is exactly in your log? Is it full verbose output? [22:20:05 CEST] <DragonsLord> Here it is > http://pastebin.com/D1FRhPfh < [22:20:25 CEST] <DragonsLord> The troubles I am experiencing are: [22:22:17 CEST] <DragonsLord> 1) "Warning: data is not aligned! This can lead to a speedloss" (not visible probably because I set 00:02:00.000, as duration) [22:23:19 CEST] <DragonsLord> 2) Also increasing NN in crop=w=iw-NN ffmpeg seems to be not able to crop a 2 pixels black line on the left [22:23:28 CEST] <DragonsLord> am I doing something wrong? [22:25:40 CEST] <DragonsLord> ChocolateArmpits, should I produce a different report? [22:25:58 CEST] <ChocolateArmpits> DragonsLord: Nah, if this is enough [22:26:20 CEST] <DragonsLord> The fact that it's not working is enough for me ;) [22:29:03 CEST] <ChocolateArmpits> DragonsLord: You should try cropping to multiples of 16 [22:29:19 CEST] <DragonsLord> O_o [22:29:27 CEST] <DragonsLord> so, is it a bug? [22:29:31 CEST] <ChocolateArmpits> However does it crash at the warning error ? [22:30:09 CEST] <DragonsLord> no, it simply seems to keem -56 from the left and -52 from the right [22:30:31 CEST] <DragonsLord> also if i increase the NN number in crop=w=iw-NN [22:30:57 CEST] <DragonsLord> c_14, told me that it should be crop from the left + crop from the right true? [22:31:32 CEST] <ChocolateArmpits> As far as I understand the x y coordinate starts at 0,0 placement [22:31:56 CEST] <DragonsLord> probably [22:32:11 CEST] <ChocolateArmpits> so x=52 means the horizontal crop will start at pixel 52 [22:33:04 CEST] <ChocolateArmpits> "The horizontal position, in the input video, of the left edge of the output video. It defaults to (in_w-out_w)/2. This expression is evaluated per-frame. " [22:33:37 CEST] <ChocolateArmpits> So in your case with x=52 it will crop 52 pixels from left and 56 from right [22:33:49 CEST] <ChocolateArmpits> leaving 612 pixels horizontal [22:33:53 CEST] <DragonsLord> true [22:34:01 CEST] <DragonsLord> exactly [22:34:37 CEST] <DragonsLord> but I have 2 pixels still present on the left [22:34:52 CEST] <DragonsLord> so I increase 108 to 110 e.g. [22:35:00 CEST] <DragonsLord> and the didn't disappear [22:36:02 CEST] <ChocolateArmpits> hmm [22:36:51 CEST] <ChocolateArmpits> I'll try testing it [22:37:02 CEST] <DragonsLord> as you can see [22:37:11 CEST] <DragonsLord> I am also upscaling the frame [22:38:00 CEST] <DragonsLord> from 612x572 up to 640x480 [22:38:47 CEST] <ChocolateArmpits> Insert format=yuv420p right before the crop filter [22:38:55 CEST] <ChocolateArmpits> That should clear up the speedloss error [22:39:09 CEST] <ChocolateArmpits> wait [22:39:15 CEST] <ChocolateArmpits> not yet [22:39:29 CEST] <DragonsLord> ok [22:40:07 CEST] <DragonsLord> I don't know if it's important ... but i run FFmpeg on WindowsOS :( [22:40:29 CEST] <ChocolateArmpits> me too, however I didn't have any cropping related problems before [22:40:48 CEST] <DragonsLord> wait wait, ChocolateArmpits [22:40:58 CEST] <DragonsLord> I did a test with AVIdemux [22:41:27 CEST] <DragonsLord> 52 L - 56 R [22:41:46 CEST] <DragonsLord> the 2 pixels line on the left should not be present [22:42:08 CEST] <DragonsLord> and I am basically doing the same using swsResize also [22:44:25 CEST] <DragonsLord> and, as far as I know, AVIdemux should use LibAV, too [22:44:45 CEST] <DragonsLord> just performed the same test with xmedia recode [22:44:50 CEST] <DragonsLord> the result is the same [22:45:09 CEST] <DragonsLord> 52 L - 56 R <- no crop left [22:45:35 CEST] <DragonsLord> I cannot understand why FFmpeg is acting in this way [22:45:41 CEST] <DragonsLord> (sorry for my bad English) [22:47:31 CEST] <ChocolateArmpits> hmm problem cropping 2 pixels on my side [22:48:32 CEST] <DragonsLord> :) [22:48:38 CEST] <DragonsLord> I am almost happy [22:48:43 CEST] <ChocolateArmpits> wait [22:48:49 CEST] <ChocolateArmpits> sorry, wrote wrong [22:48:58 CEST] <ChocolateArmpits> *No problems here cropping 2 pixels [22:49:31 CEST] <DragonsLord> Did you start with my same source [22:49:38 CEST] <DragonsLord> in terms of size and so on? [22:49:50 CEST] <ChocolateArmpits> I found a 720x576 image [22:49:58 CEST] <ChocolateArmpits> and applied same cropping settings [22:50:17 CEST] <DragonsLord> to an image [22:50:28 CEST] <DragonsLord> not to a video [22:50:31 CEST] <llogan> blame your player [22:50:44 CEST] <DragonsLord> hello, llogan [22:50:45 CEST] <ChocolateArmpits> I'm testing the cropping with ffplay [22:51:00 CEST] <ChocolateArmpits> you can paste your vf there [22:51:06 CEST] <ChocolateArmpits> and see how the output looks [22:52:57 CEST] <DragonsLord> encoding [22:53:22 CEST] <DragonsLord> llogan, if you were referring to my player ... this is not the case [22:54:01 CEST] <DragonsLord> the same video, interlaced, cropped and resized with both AVIdemux and Xmedia Recode is fine with the same player [22:54:04 CEST] <DragonsLord> (VLC) [22:55:43 CEST] <DragonsLord> [swscaler @ 000000000033c3a0] Warning: data is not aligned! This can lead to a speedloss [22:56:09 CEST] <llogan> ignore that [22:58:20 CEST] <DragonsLord> here is the image: [22:58:22 CEST] <DragonsLord> http://postimg.org/image/xmh8dofhh/ [22:58:46 CEST] <ChocolateArmpits> You mean that dark line on the left ? [22:58:57 CEST] <DragonsLord> yep [22:59:34 CEST] <ChocolateArmpits> Is this the output of the command line you shared in the pastebin? [23:00:01 CEST] <DragonsLord> yes [23:00:05 CEST] <DragonsLord> oh [23:00:07 CEST] <DragonsLord> not really [23:00:11 CEST] <DragonsLord> that was pass-1 [23:00:23 CEST] <DragonsLord> this is of course the output for pass-2 [23:00:24 CEST] <ChocolateArmpits> ok, but the filter settings are the same right? [23:00:28 CEST] <DragonsLord> yes [23:00:31 CEST] <DragonsLord> exactly the same [23:00:35 CEST] <llogan> you should provide a short input sample [23:00:39 CEST] <ChocolateArmpits> Did you try setting crop x to 54 ? [23:01:47 CEST] <DragonsLord> yes, increasing number 108 e.g. to 118 [23:02:31 CEST] <ChocolateArmpits> The first setting ("w") controls how wide the cropped image will be [23:02:48 CEST] <ChocolateArmpits> It doesn't tell where the cropping should start [23:03:12 CEST] <ChocolateArmpits> That's what "x" setting does for cropping width [23:03:29 CEST] <DragonsLord> so probably I misunderstood something [23:04:08 CEST] <DragonsLord> crop=w=iw-108:h=ih-4:x=52:y=4 [23:04:36 CEST] <ChocolateArmpits> here "w" is 720-108=612, so the width of the output will be 612 pixels [23:04:53 CEST] <ChocolateArmpits> the width will be cropped starting from" x=52" [23:05:07 CEST] <DragonsLord> OH [23:05:08 CEST] <DragonsLord> so [23:05:19 CEST] <DragonsLord> -52 if from the left of the frame [23:05:23 CEST] <DragonsLord> or from the right [23:05:24 CEST] <ChocolateArmpits> exactly [23:05:26 CEST] <ChocolateArmpits> LEFT [23:05:40 CEST] <ChocolateArmpits> 0,0 denotes upper left corner of the frame [23:06:05 CEST] <DragonsLord> however [23:06:12 CEST] <DragonsLord> I put these setting into AVIdemux [23:06:16 CEST] <DragonsLord> -52 left [23:06:25 CEST] <DragonsLord> -56 right [23:06:30 CEST] <DragonsLord> -4 up [23:06:34 CEST] <DragonsLord> 0 down [23:07:59 CEST] <DragonsLord> and ... [23:08:11 CEST] <DragonsLord> here is the resulting and corresponding frame: [23:08:12 CEST] <DragonsLord> http://s10.postimg.org/97rdvla3t/vlcsnap_2015_04_10_23h07m16s64.png [23:08:21 CEST] <DragonsLord> as you can see [23:08:37 CEST] <DragonsLord> non black vertical line on the left, right? [23:09:08 CEST] <ChocolateArmpits> will it crops slightly more than in your ffmpeg example [23:09:12 CEST] <ChocolateArmpits> not just the line [23:10:09 CEST] <DragonsLord> I don't really understand what you mean [23:10:11 CEST] <DragonsLord> however [23:10:17 CEST] <DragonsLord> the basic issue is that [23:10:26 CEST] <DragonsLord> from my point of view setting are the same [23:10:31 CEST] <DragonsLord> and the result is different [23:10:34 CEST] <DragonsLord> furthermore [23:11:16 CEST] <DragonsLord> I am trying to set crop=w=iw-160 [23:14:43 CEST] <DragonsLord> ChocolateArmpits, it can be that ffmpeg is trying to crop up to the next multiple of 16 so 112? [23:15:12 CEST] <ChocolateArmpits> That is not possible [23:15:22 CEST] <DragonsLord> I agree [23:15:23 CEST] <ChocolateArmpits> It's working with uncompressed data [23:15:27 CEST] <DragonsLord> I agree [23:15:31 CEST] <DragonsLord> but, at the end [23:15:41 CEST] <DragonsLord> this is what it's happening [23:16:29 CEST] <DragonsLord> ChocolateArmpits!!! [23:16:48 CEST] <DragonsLord> increasing NN in crop=w=iw-NN [23:16:57 CEST] <DragonsLord> and leaving x-crop untouched [23:17:10 CEST] <DragonsLord> ffmpeg should crop more on the left, right? [23:17:17 CEST] <ChocolateArmpits> on the right [23:17:27 CEST] <ChocolateArmpits> actually [23:17:33 CEST] <ChocolateArmpits> it should crop LESS on the right [23:17:44 CEST] <ChocolateArmpits> as you're expanding the output width leaving less room for the right portion [23:18:17 CEST] <ChocolateArmpits> left will be left intact [23:19:31 CEST] <DragonsLord> OK [23:19:40 CEST] <DragonsLord> this is my misunderstanding [23:20:32 CEST] <DragonsLord> just to recap [23:20:44 CEST] <DragonsLord> x= 52 52 is cropping from the left? [23:21:23 CEST] <ChocolateArmpits> YES [23:22:05 CEST] <DragonsLord> <c_14> A "simple" formula is w=iw-[total width to cut]:h=ih-[total height to cut]:x=[pixels to cut from the right]:y=[pixels to cut from the top] [23:22:09 CEST] <ChocolateArmpits> Well to be absolutely correct, that's where the first collumn of pixels will be passed [23:22:31 CEST] <ChocolateArmpits> so 52nd pixel will be in the output [23:22:40 CEST] <DragonsLord> c_14, stated x=[pixels to cut from the right] [23:22:41 CEST] <ChocolateArmpits> while 51st no [23:22:54 CEST] <c_14> left [23:23:02 CEST] <ChocolateArmpits> well 0 is a pixel too :) [23:23:09 CEST] <c_14> I think I corrected myself. I hope I corrected myself. [23:23:17 CEST] <DragonsLord> hello c_14 [23:23:45 CEST] <DragonsLord> But, in this case I expect to have the same result of AVIdemux [23:24:34 CEST] <DragonsLord> and furthermore [23:25:13 CEST] <DragonsLord> setting w=iw-[total width to cut]=iw-160 and leaving x=[pixels to cut from the LEFT] untouched [23:26:03 CEST] <ChocolateArmpits> Well considering avidemux presents the information differently than the barebones crop filter I think that there could be some different interpretation of the values [23:26:04 CEST] <DragonsLord> OK [23:26:06 CEST] <DragonsLord> I got it [23:26:19 CEST] <DragonsLord> I set -56 left in AVIdemux [23:26:38 CEST] <DragonsLord> while I set -52 left within ffmpeg [23:26:39 CEST] <DragonsLord> :| [23:26:44 CEST] <ChocolateArmpits> see [23:27:11 CEST] <DragonsLord> sorry for causing you some lost time [23:27:15 CEST] <DragonsLord> re-encoding [23:27:26 CEST] <DragonsLord> in italy we would say "prova del 9" [23:27:29 CEST] <DragonsLord> :-P [23:27:36 CEST] <ChocolateArmpits> what does that mean? [23:28:24 CEST] <DragonsLord> uhm [23:28:37 CEST] <DragonsLord> can you remember when you were at the primary school? [23:28:47 CEST] <DragonsLord> more or less when you were 5-6 years old? [23:29:19 CEST] <ChocolateArmpits> that was kindergarten for me at that age [23:29:23 CEST] <ChocolateArmpits> some parts [23:29:43 CEST] <DragonsLord> the teacher was trying to teahc you additions [23:30:11 CEST] <DragonsLord> OK [23:30:16 CEST] <DragonsLord> in english it should be [23:30:17 CEST] <DragonsLord> casting out nines [23:30:33 CEST] <DragonsLord> Casting out nines is a sanity test to ensure that hand computations of sums, differences, products, and quotients of integers are correct. By looking at the digital roots of the inputs and outputs, the casting-out-nines method can help one check arithmetic calculations. The method is so simple that most schoolchildren can apply it. [23:30:40 CEST] <DragonsLord> Wikipedia helps a lot ;) [23:31:24 CEST] <DragonsLord> can you understand what I meant? [23:31:29 CEST] <ChocolateArmpits> sorta [23:31:34 CEST] <DragonsLord> hi Kolizer [23:31:44 CEST] <DragonsLord> sorta? [23:31:52 CEST] <Kolizer> hello, is there a function executing ignoring colors like chrome key? [23:34:10 CEST] <BtbN> No chroma key. [23:35:40 CEST] <DragonsLord> ChocolateArmpits, test completed successfully :))) [23:35:53 CEST] <DragonsLord> crop was performed perfectly [23:36:05 CEST] <DragonsLord> thank you very much for your time and your patience [23:36:16 CEST] <DragonsLord> the same also "applies" for c_14 and llogan [23:36:16 CEST] <ChocolateArmpits> DragonsLord: no probs [23:36:19 CEST] <DragonsLord> :-P [23:36:42 CEST] <DragonsLord> ChocolateArmpits, YES ... there was a problem: myself XD [23:36:58 CEST] <Kolizer> That is, I won't be able to overlay an image with a green background and make the background transparent? [23:39:24 CEST] <ChocolateArmpits> Kolizer: not with libav filters it seems. If you are working with files then try the Keying node in Blender or Chroma Key node in Natron, if you need a scripted solution maybe Avisynth, if streaming from an outside source then CasparCG suite. [23:39:50 CEST] <ChocolateArmpits> all these are open source [23:40:59 CEST] <DragonsLord> ChocolateArmpits, I am going to smoke ... when I will be back could I share another dubt with you? [23:41:03 CEST] <Kolizer> I want tools ffmpeg [23:41:09 CEST] <BtbN> Chroma/Color key is one major usefull feature that's just plain missing from ffmpeg. [23:41:59 CEST] <ChocolateArmpits> Avisynth scripts can be fed into ffmpeg [23:42:27 CEST] <Kolizer> cool =) [23:44:31 CEST] <Kolizer> Thanks for the great answers ) [23:46:35 CEST] <Kolizer> I hope someday will add this feature [23:49:27 CEST] <BtbN> It's not too complex, just nobody has done it. [23:49:34 CEST] <Kolizer> I just stream using ffmpeg and I need real time to do the transparency [23:49:56 CEST] <BtbN> Why not use something like OBS for that? [23:50:14 CEST] <Kolizer> Oh no) [23:50:32 CEST] <BtbN> It's way better suited for interactive live streams. [23:50:38 CEST] <ChocolateArmpits> obs has a bad keyer [23:50:51 CEST] <BtbN> a bad what? [23:50:53 CEST] <Kolizer> There is also no such feature [23:51:06 CEST] <BtbN> No such feature? OBS definitely does support chroma key. [23:51:54 CEST] <ChocolateArmpits> well it's called color key [23:51:59 CEST] <BtbN> No it's not. [23:52:07 CEST] <BtbN> It's called color-key for image source. [23:52:13 CEST] <BtbN> Video-Capture-Sources have chroma key. [23:52:34 CEST] <ChocolateArmpits> monitor capture has color key [23:52:44 CEST] <ChocolateArmpits> as does window capture [23:52:44 CEST] <BtbN> Yes, because it makes much more sense for that. [23:53:01 CEST] <BtbN> You get bit exact color captures from it. [23:53:06 CEST] <BtbN> So color key is what you usualy want. [23:59:22 CEST] <DragonsLord> Guys, if I have multiple videos and the duration is the same [23:59:31 CEST] <Kolizer> I would have written the functionality of the color key for ffmpeg, if I was a programmer =) [23:59:46 CEST] <DragonsLord> if I set the 2-pass medium bitrate the same [00:00:00 CEST] <DragonsLord> I should obtain files with the same size, right? [00:00:00 CEST] --- Sat Apr 11 2015

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