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October 2013

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ffmpeg-devel.log.20131020
by burek 21 Oct '13

21 Oct '13

[00:15] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:62533eab6fb1: ffv1enc: use 64bit in maxsize calculation [00:15] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:9c0fe487c755: avcodec/h264_parser: fix order of operations [01:15] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:4c67ed870503: avcodec/hevc: fix EOB/EOS check [01:15] <cone-239> ffmpeg.git 03Mickaël Raulet 07master:92a97d1168cc: hevc: inline cabac in residual coding(cherry picked from commit 17d7a880445b72feb36d684ae1f0597195811e97) [01:15] <cone-239> ffmpeg.git 03Mickaël Raulet 07master:b5d197a38b64: hevc: inline cabac in hls_mvd_coding(cherry picked from commit ad387195ad04e8a005a1bfd509e9e4f827e68fa9) [01:15] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:acecd6b4d7fd: avcodec/hevc_cabac: trivial simplifications [01:15] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:f2eca8d06060: avcodec/hevc: do not dereference pointer before null check in verify_md5() [01:15] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:23d69b158af4: avcodec/hevc_refs: fix h/vshift calculation in ff_hevc_output_frame() [01:54] <michaelni> BBB / ubitux coverity also found things in vp9 (didt check if they are real or false positives) [01:55] <BBB> ffvp9? [01:55] <BBB> feel free to send to me [01:55] <BBB> I have no time to check coverity [01:55] <BBB> (or anything) [02:02] <michaelni> ffvp9, yes, not sure how i could send something from coverity [02:06] <michaelni> for example: "cond_const: Checking "i < 10" implies that the value of "i" is 10 on the false branch" and then it thinks "3315 s->f = s->fb[i];" can overread [02:06] <BBB> is there a link to some web report? [02:10] <michaelni> AFAIK yu need to have an account, i just send you an invite through the web interface to @gmail.com [02:10] <michaelni> theres a good chance it will end in your spam folder [02:13] <BBB> hm... [02:13] <BBB> if you feel there's anything real, feel free to file a bug or so, or copypaste it to my email [02:13] <BBB> else there's a lot chance I'll look at it [02:13] <BBB> as for the overread, it can't overread, vp8 has an assert there [02:13] <BBB> I remved the assert for vp9 because it's obvious [02:14] <michaelni> ok ill mark that overread as false positive [02:17] <michaelni> BBB, when you click on the left side on FILES/with outstanding issues and then vp9... from the list then you can see exactly the ones about vp9 [02:17] <michaelni> i see 1 in vp9 and 6 in vp9dsp only [02:19] <michaelni> the 6 all look related [02:21] <iive> the 3315 one does look fishy. [02:22] <michaelni> either way, ill look at the vp9 onces eventually in case you have no time [02:23] <michaelni> but there are a few others left that fall closer into my responsibility which ill look at first [02:43] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:6338f1b3c095: avcodec/tiff: remove byte based bpp special case [02:43] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:2e9b79fc003f: avcodec/wavpackenc: fix uninitialized ret [09:32] <mraulet> michaelni: I have fixed the latest sequence [11:25] <cone-729> ffmpeg.git 03Mickaël Raulet 07master:c841d02c056f: hevc: fix PPS_A_qualcomm_7(cherry picked from commit 2af177a8761c88eb477a658eebcf4264068aa773) [11:25] <cone-729> ffmpeg.git 03Mickaël Raulet 07master:1b5a52f425d9: hevc: pretty print(cherry picked from commit 64a4b623b7d66dfc0f3883e5f1d9125c00c3b18c) [12:17] <cone-729> ffmpeg.git 03Reimar Döffinger 07master:c9a22d69afc7: hevc: Initialize sample aspect to valid value. [12:20] <ubitux> BBB: iiuc, with pmaddwd, i will need to mix things inputs and vp9 cos constants together; is it appropriate to use the punpck* instr for such purpose? [12:22] <ubitux> well i guess so. [13:26] <BBB> ubitux: sure [13:26] <BBB> ubitux: anything is fine if it's faster :) [13:46] <BBB> michaelni: both in vp9 are false [13:46] <BBB> michaelni: the ones in vp9dsp are from ubitux' code, I think, he should probably look at them [13:47] <BBB> michaelni: the vp9.c ones can be "solved" with a simple assert (assert(pred < 8); and assert(i < 10);) [13:49] <ubitux> i'll have a look [14:07] <burek> hi all :) [14:07] <burek> can anyone help me how to answer this question (about v 0.5.x) http://ffmpeg.gusari.org/viewtopic.php?f=16&t=1114 [14:09] <michaelni> marked the vp9.c one as false positive [14:21] <BBB> ubitux: I think line 775 and 777 should be size - j - 1 in the memcpy, instead of size - j [14:21] <BBB> ubitux: memset overrides the size - j - 1 position [14:21] <BBB> ubitux: that should fix them [14:21] <BBB> the rest seems clean [14:22] <ubitux> i didn't look yet, i'm still in the asm [14:22] <BBB> ok [14:22] <BBB> michaelni: you then [14:22] <ubitux> i'll look, no worry :) [14:22] <BBB> michaelni: and feel free to apply the asserts tothe codebase, it's somewhat clean to document [14:22] <BBB> ubitux: and 32x32 asm is steadily progressing, it's just a lot of it, as you'd know, you wrote the c [14:22] <BBB> :) [14:22] <michaelni> BBB ok, will do [14:23] <ubitux> BBB: yeah i can guess so :p [14:26] <BBB> michaelni: ty - poke me if there's any new issues [14:26] <BBB> the asm one is a nice catch [14:26] <BBB> er... dsp [14:33] Action: Daemon404 logs nto coverity for the first time ever [14:37] <Compn> took me 10 minutes thinking how to crop an mpeg2 video [14:37] <Compn> which program supports copying audio and video without messing anything up [14:37] <Compn> finally just dd'd it :D [14:47] <cone-729> ffmpeg.git 03Michael Niedermayer 07master:f198efb179cb: avcodec/vp9: Add asserts to help source code analyzers [14:47] <cone-729> ffmpeg.git 03Michael Niedermayer 07master:78e6f83ce027: avformat/au: add assert to help source code analyzers [15:46] <ubitux> gdb not unrolling macros sadness. [16:54] <cone-729> ffmpeg.git 03Michael Niedermayer 07master:baab248c499a: avformat/network: check for fcntl() failure in ff_socket() [16:54] <cone-729> ffmpeg.git 03Michael Niedermayer 07master:b294a4beec24: avformat/oggparsecelt/celt_header: fix memleak [17:03] <cone-729> ffmpeg.git 03Michael Niedermayer 07master:2c1e075308e1: avformat/oggparseflac: check ff_alloc_extradata() return code [17:03] <cone-729> ffmpeg.git 03Michael Niedermayer 07master:eb5cc8febc6c: avformat/oggparsespeex: Check for extradata allocation failure [17:42] <BBB> ubitux: michaelni: did you guys fix the vp9dsp one or do you need a patch? [17:42] <ubitux> i didn't look yet, will do later today probably [17:43] <ubitux> i might be done with the first dim for idct soon [17:43] <ubitux> ...but it looks like i've already some rounding problem :( [17:44] <BBB> easy to fix [17:44] <BBB> goodluck debugging asm :) [17:45] <BBB> michaelni: and yeah the asserts look good thanks [17:56] <j-b> BBB: I ffvp9 is higher priority. But some free distributions do not wnat libavcodec, but they would welcome libvpx [17:57] <Compn> 'free distributions' ? [17:57] <j-b> yeah fedora [17:58] <Compn> lol [17:58] Action: Compn afk [17:58] <j-b> Compn: yeah :) [18:00] <wm4> so vlc compiles without libav*? [18:01] <kierank> probably without libavcodec [18:02] <j-b> wm4: yep [18:02] <j-b> wm4: the biggest issue is swscale, you know :) [18:03] <wm4> fascinating [18:03] <wm4> the result can't be awfully useful, or do you hook into gstreamer? [18:04] <j-b> what would be the use of that? [18:04] <BBB> i bet it can decode webm and ogg/theora [18:04] <BBB> and wav [18:05] <wm4> gstreamer has licensed h264 decoders (provided by external companies) [18:05] <BBB> the dream of freedom lovers [18:05] <kierank> wm4: why can't they licence libavcodec [18:05] <BBB> wm4: no, those companies have licensed h264 decoders [18:05] <BBB> gstreamer doesn't own or distribute them [18:05] <wm4> right, wrong wording [18:06] <BBB> happens... useful thing to have [18:06] <BBB> chrome has such a license also [18:06] <BBB> they happen to use it to ship libavcodec's implementation [18:07] <BBB> I'm happier that way, since at least it's free software [18:07] <BBB> it's better than licensing crap along with close source software [18:07] <Compn> is it still faster than the other closed options ? [18:07] <BBB> libavcodec's? unlikely, it's probably about as fast, within a few % error margin [18:07] <BBB> but I never measured [18:08] <beastd> probably faster to fix if needed :) [18:09] <j-b> wm4: licensed? lol. [18:09] <j-b> wm4: we have a gstreamer module, never merged. [18:09] <j-b> wm4: no, it's more to play the "free" stack, aka Ogg/Vorbis/Theora/FLAC/Opus+Webm1/2 [18:10] <Compn> luckily there are no multimedia patents that cover those codecs :) [18:11] <j-b> right :) [18:11] <wm4> anyway, they could just ship a crippled libavcodec... [18:12] <j-b> of course, that would be what makes sense... [18:12] <j-b> but sense is not really shared with some of our colleagues [18:16] <ubitux> BBB: i want to go from dw A,B,A,B to -B,A,-B,A; should i transform the first (assuming stored in a reg) into the second, or just load it? [18:16] <ubitux> basically; pw_X: dw A,B,A,B; pw_Y: -B,A,-B,A vs loading only pw_X and transform it into pm_Y when necessary [18:18] <ubitux> i might be able not to do the shuffling though, and find a way negating all both A and V [18:18] <ubitux> s/V/B/ [18:45] <michaelni> hi saste, do you have time to look at some ffprobe coverity issue(s?) [18:45] <saste> michaelni, yes nail me if you see no patch from me in a few days [18:46] <cone-729> ffmpeg.git 03Michael Niedermayer 07master:88d4ff4b5f4c: avformat/utils: Check av_packet_new_side_data() return before using it [18:46] <cone-729> ffmpeg.git 03Michael Niedermayer 07master:ad28fe35c55b: avutil/file_open: Print debug message if setting close on exec fails [18:46] <cone-729> ffmpeg.git 03Michael Niedermayer 07master:2d8ccf0adcae: avutil/opt: initialize ret [19:22] <cone-729> ffmpeg.git 03Lukasz Marek 07master:d1f383341fda: lavd/pulse_audio_enc: add support for flushing [19:24] <cone-729> ffmpeg.git 03Lukasz Marek 07master:36d0b15b4e57: MAINTAINERS: add lavd/fbdev_enc entry [20:44] <BBB> ubitux: whichever is easier. transforming might suck because it means extra instructions, but whichever is faster and simpler and smaller code [20:44] <BBB> (in that order of relevance) [20:44] <ubitux> yes but transforming a register might be faster than loading from memory [20:45] <ubitux> basically, 1 load + transform vs 2 load [20:45] <ubitux> i was wondering about the cost of such transform [20:45] <ubitux> but well, i'd better get something working first anyway [20:45] <ubitux> i'm still derping with various random issues right now [20:45] <BBB> good question, not sure [20:46] <BBB> what you're saying makes sense but I'm not sure [20:46] <BBB> I tend to just multiply straight from memory [20:46] <BBB> i.e. pmaddwd mX, [pw_...] [20:46] <BBB> as opposed to mova mY, [pw_...]; pmaddwd mX, mY [20:46] <BBB> but for 4x4 your version may be faster because you have enough registers [20:46] <ubitux> i don't have enough registers actually :( [20:47] <BBB> oh [20:47] <BBB> then multiply from memory [20:47] <ubitux> i'm limiting myself to 8 reg though [20:47] <ubitux> also, i'm doing the add mul in 2 steps [20:48] <ubitux> so i exhaust the register a bit faster [20:49] <BBB> you have to do 8 reg, it's mmx [20:49] <BBB> not xmm [20:49] <BBB> 16 regs is xmm only [20:49] <BBB> what is add-mul in 2 steps? [20:49] <BBB> you're just doing punpcklwd; pmaddwd; padddx2; psradx2; packssdw, right? [20:50] <BBB> actually punpckl/hwd, (pmaddwd, paddd, psrad)x2, packssdw [20:50] <BBB> anyway [20:50] <ubitux> yes, that ^ [20:50] <BBB> yeah that's fine [20:50] <ubitux> on the 8 reg, i have 2 already used to store t0 and t1 [20:51] <ubitux> so i have 6 reg remaining, and i'm trying to avoid reloading from memory the coef and various other saves [20:51] <ubitux> anyway, i still need some time :p [20:52] <BBB> ok [20:52] <BBB> I'm not there yet either [20:52] <BBB> just make sure you submit a patch to fix that coverity issue [20:54] <ubitux> yeah, i'll look at this in about 10m [21:04] <ubitux> 14:21:06 <@BBB> ubitux: I think line 775 and 777 should be size - j - 1 in the memcpy, instead of size - j [21:04] <ubitux> 14:21:42 <@BBB> ubitux: memset overrides the size - j - 1 position [21:04] <ubitux> confirmed [21:04] <BBB> ty [21:16] <cone-729> ffmpeg.git 03Marton Balint 07master:b2d9790c2ba7: lavc: make avcodec_decode_subtitle2 more robust [21:17] <cone-729> ffmpeg.git 03Ronald S. Bultje 07master:fed483f188b3: avcodec/vp9dsp: fix overwrite by 1 in vert_left pred. [21:17] <ubitux> BBB: thx ^ [21:17] <BBB> back to simd now [21:17] <BBB> I still hope to have a full 32x32 in a few weeks [21:17] <BBB> that would be a good 10-20% speedup [21:17] <ubitux> yeah well, now that i'm in coverity i'll fix one more elsewhere [21:18] <ubitux> michaelni: i don't remember, do i need to mark them as fixed in coverity or they will disappear by themselves? [21:19] <BBB> anything vp8-related there? [21:19] <BBB> I didn't check very closely [21:20] <ubitux> i don't see anything [21:21] <ubitux> there are not that much issues remaining [21:29] <cone-729> ffmpeg.git 03Clément BSsch 07master:4189fe11ffcb: avformat/vobsub: fix invalid sub queue access while seeking. [21:30] <ubitux> ok, now back to simd [21:48] <michaelni> ubitux, i think they will disappear on their own but if you mark them as fix submited or something then others know before that already that they dont need to look at that one anymore [21:48] <ubitux> ok [21:52] <Compn> ZMK4 ? another xvid fourcc ? hmm [00:00] --- Mon Oct 21 2013

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ffmpeg.log.20131020
by burek 21 Oct '13

21 Oct '13

[01:13] <kriskropd> does ffmpeg have some built in hotkey to pause a running encode process? if not, is it safe to send 'kill -STOP pid' and 'kill -CONT pid' to pause adn resume a running ffmpeg process? [01:17] <zybil_> hi+ [04:58] <ferrouswheel> Hi all - anyone have any tips on how to automate conversion of old codecs to newer ones? I have some old videos that don't play on my RPi. Ideally I'd scan all my media, identify what files are Div3, and convert to something newer. [05:17] <sacarasc> If you can, you'd be better off starting with the source material, as Div3 is pretty poor, and you'll only lose quality encoding from it, [05:19] <sacarasc> But, if you can script a little bash, you could use ffprobe to see what codec it's using. [05:19] <ferrouswheel> Yeah - it's pretty bad, but there are videos that would be hard to track down again. [05:19] <ferrouswheel> Thanks :-) [05:19] <ferrouswheel> I'd started investigating ffprobe [05:20] <ferrouswheel> Using xargs and grep to find out just how many Div3 I have. Hopefully not many ;-) [07:40] <rudolfovich> I'm trying to implement a real-time video streaming via udp. [07:40] <rudolfovich> At this moment, I implemented a simple communication protocol, where each coded video frame is wrapped into a NAL unit and divided into pieces no larger than MTU. [07:40] <rudolfovich> This works fine in LAN, but when I test over internet, i have a lot of undelivered packets and forced to drop whole frame. [07:40] <rudolfovich> It is obvious that putting in NAL entire frame - it's a bad idea. [07:40] <rudolfovich> But I do not know what part of the encoder work result, is best to wrap into NAL to ensure minimum loss. [13:38] <Ulfalizer> sorry if this is a bit tangential. i'm thinking of adding a video recording feature to an emulator i'm working on. it would need to take the raw audio and video streams and encode and stream them to a file on disk. easy switching of encoding formats would be a plus. would libavcodec be a good match for that? any possibly simpler alternatives? i have very little experience when it comes to video. [13:39] <JEEB> libavcodec and libavformat probably are the simplest way to provide multiple alternatives [13:39] <JEEB> libavcodec for the actual audio/video streams being encoded, and then libavformat to write those streams into a container [13:39] <Ulfalizer> how about gstreamer? or is it more tailored to streaming over a network and the like and hence overkill? [13:40] <JEEB> gstreamer really isn't about streaming, it's meant to be a multimedia framework like DirectShow on Windows [13:41] <JEEB> and it generally is a mess and using libraries straight out in general usually ends up being a better idea [13:42] <Ulfalizer> ok, i'll look into libavcodec and libavformat. thanks! [13:43] <Ulfalizer> from some quick reading up this might be a politically loaded questions, but how do the ffmpeg and libav versions of libavcodec compare? :) [13:43] <Ulfalizer> which one is more commonly used? pros and cons? [13:44] <JEEB> well, ffmpeg tends to merge more stuff in while libav generally merges only things deemed important enough and if someone pokes them with stuff [13:44] <JEEB> generally the usage should be similar enough, except for some derpy cases [13:45] <Ulfalizer> switching from one to the other should be relatively painless? [13:46] <JEEB> in general, yes [13:47] <JEEB> when you start adding support for various versions of things and such it tends to get a bit more painful [13:47] <Ulfalizer> ok, i'll try the ffmpeg version first [19:06] <zennist> Hi anyone knows about problem with decoding AAC audio stream? [19:06] <zennist> I've got a code snippet that decodes almost all formats correctly using libavcodec, but it fails on AAC streams and instead gives back noise [19:09] <zennist> But the command line ffmpeg can still convert the files correctly; so I'm thinking it's a problem in my own code, but I don't know what's wrong [19:09] <zennist> Here's the code http://pastebin.com/VfkU1kVi [19:11] <JEEB> zennist, could it be that you're expecting 16 bit integer pcm or something? [19:11] <JEEB> didn't check the code, but that's the most common error [19:11] <zennist> Yes, I'm playing the audio with libao, so I guess that's 16bit pcm [19:12] <zennist> JEEB: does the decoded frame not contain data as 16bit pcm? [19:12] <JEEB> with lossy formats it's generally float [19:13] <JEEB> only lossless formats will have integer audio [19:13] <JEEB> you can use libavresample or libswresample for conversion for output [19:13] <zennist> JEEB: okay..so I have to use resample to convert it pcm correct? [19:14] <JEEB> it's still pcm :P [19:14] <JEEB> it's just 32bit float [19:14] <JEEB> not 16 bit integer or whatever you're expecting [19:15] <JEEB> so yes, use one of the resampling libraries to make the output be what you need [19:15] <zennist> JEEB: cheers! I would make some changes and see whether it works out [20:19] <Ulfalizer> can avcodec_encode_video() do format conversions? e.g. for encoding ARGB frames to mpeg, which seems to use YUV. [20:20] <JEEB> generally no [20:20] <JEEB> you would have to use libswscale for that, or create your own conversion routines [20:21] <Ulfalizer> ok [20:27] <Ulfalizer> what's the deal with AVFrame.data being an array of uint8_t* by the way? is it more common for video to use planar rather than packed representations? [20:29] <Ulfalizer> and how would something like ARGB appear there (if it ever does)? will it always be split? [22:01] <zennist> Just added resampling to my program; and swr_convert is returning 0 all the time. Any ideas? [22:02] <zennist> I put src_sample_fmt as AV_SAMPLE_FMT_FLTP and dest_sample_fmt as AV_SAMPLE_FMT_S32 [22:07] <JEEB> Ulfalizer, yes -- esp. with YCbCr video it's more often planar than packed [22:07] <JEEB> Ulfalizer, also I think you should be able to handle both packed and planar ARGB [22:08] <JEEB> zennist, unfortunately I can't say much more than look at the format you're getting from the decoder and stare at the doxygen :s [22:11] <Fresh> hello, would this be the correct place to ask a noob usage question? [22:13] <Fresh> I am trying to create an http video stream of my desktop so I can view it on my tv, but i cannot figure out how to get the segementer to intergrate with ffmpeg. If someone can tell me the missing link, I would appreciate it [22:14] <Fresh> I have downloaded a precompiled segmenter, with these expected parameters. USAGE: segmenter -i input-MPEG-TS-file -d seconds-per-segment [-o segment-file-prefix] -x output-playlist-m3u8 [-p http-prefix] [-w live-segments] [--watch-for-kill-file] [--strict-segment-duration] [22:15] <Fresh> Here is my ffmpeg input ffmpeg -f dshow -i video="UScreenCapture" -r 10 -vcodec mpeg4 -q 12 -f mpegts <OUTPUT HERE> [22:15] <Fresh> Here is the segmenter input segmenter -i <INPUT HERE> -d 10 -x test.m3u8 [22:15] <Fresh> how can I end up streaming the stuff to http://localhost:8080/ instead of having it on the local disk ? [22:15] <Fresh> thank you if to anyone, who can maybe clear up the missing step. [22:15] <Fresh> I have searched google for quite a while now [22:16] <zennist> JEEB: mhh, the frame I got from the decoder does indicate the format is still AV_SAMPLE_FMT_FLT though.. [22:38] <viric> Hi! [22:39] <viric> I've a video reported as: Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc [22:39] <viric> But *it is interlaced*. [22:39] <Mavrik> and? [22:39] <viric> I want yadif to provide 50fps [22:39] <viric> yadif=1 doesn't work with it. I think because it's marked as non-interlaced... [22:40] <viric> yadif alone deinterlaces fine, though. [22:40] <viric> (it's a direct recording from DVB) [22:41] <viric> Do you have any idea how can I tell ffmpeg that the image comes from two fields? [22:41] <viric> (for what I understand, it's interlaced video encoded as progressive) [22:41] <viric> Is it an usual thing? Do I understand it right? [22:42] <viric> ah, damn it. only PART of the video is interlaced... I guess the broadcaster messed all up [22:44] <Mavrik> hmm, I mean your question is wierd [22:44] <Mavrik> if you're deinterlacing the video, you're merging the fields [22:44] <Mavrik> at least that's the primary use case [22:44] <viric> yes [22:45] <viric> but if I have 25fps interlaced video, that means 50 fields per second. yadif can give me either 25fps or 50fps progressive, right? [22:46] <viric> in this case, the video media is '25fps progressive', but I see in the pictures that it is interlaced. So it means that the broadcaster sent interlaced video as progressive. [22:46] <viric> Or I'm completely wrong? where ffprobe tells if it is interlaced or not? :) [22:48] <cbsrobot> did you add -r 50 ? [22:48] <viric> hm no [22:52] <viric> maybe I'm just checking things bad. [22:53] <viric> In this video, I see parts of the picture interlaced, and parts clearly not interlaced. [22:53] <Mavrik> there's no real reason why WHOLE video should be interlaced :) [22:53] <viric> hehe [22:53] <viric> but there should be some flags in the file, whether it has one or two fields, no? [22:54] <Fresh> I was wondering if anyone could point me in the direction of how to setup a http video stream [22:54] <viric> Fresh: ffserver comes with examples [22:54] <Fresh> i will take a look over there now, thank you [22:55] <viric> Hm in fact I don't remember ever a program noticing that a file is interlaced, and so, automatically deinterlacing it. [22:55] <viric> Mavrik: your last sentence has very valuable information, it seems [22:57] <Mavrik> viric, I'm not entirely sure about MPEG-2 [22:57] <Mavrik> but in H.264 you can't know if frame is interlaced without actually decoding it [22:59] <Fresh> viric, is it impossible under windows? It seems like ffserver is only for linux, yet ffmpeg has been able to act as a udp,rdp, etc server [22:59] <viric> Mavrik: but once decoding it... is it clear? [22:59] <Mavrik> well [22:59] <Mavrik> not really :) [23:00] <viric> ha [23:00] <viric> what a mess [23:01] <viric> Btw, when I shot video, and I keep only 25fps (instead of the '50fps interlaced'), it seems to me as the movement is much less soft. [23:02] <viric> (when I pan the camera horizontally, for example) [23:02] <viric> does this mean that I should keep 50fps, that I should not move the camera so fast, or something else? [23:02] <viric> I don't see 50fps videos very spread... so I guess it's the movement thing. I move too fast. [23:05] <Mavrik> viric, if you want to keep smoothness, leave it interlaced [23:05] <Mavrik> anything else you do will be a mess [23:05] <viric> aha... [23:05] <viric> interesting [23:06] <viric> how do *encoders* know that it's interlaced? [23:07] <viric> Imagine I want to reencode from mp4 to webm, both interlaced. If players don't know about it being interlaced, how do encoders know? [23:12] <viric> I imagine encoders care, whether the input is interlaced or not [23:14] <Mavrik> hmm, good question [23:17] <viric> I'm afraid vp8 doesn't support "interlaced mode" [23:18] <viric> but I guess I've to tell the encoder [23:20] <viric> the x264 program allows the interlaced mode [23:26] <viric> Mavrik: maybe the 'il' filter? [23:27] <viric> hm no [23:33] <Ulfalizer> JEEB: okay [23:44] <viric> hm neither VP8 or Theora supports interlaced video. [23:45] <viric> Isn't there any free codec I can use for interlaced video? [00:00] --- Mon Oct 21 2013

1 0

ffmpeg-devel.log.20131019
by burek 20 Oct '13

20 Oct '13

[00:01] <cone-893> ffmpeg.git 03Lukasz Marek 07master:f5695926235c: lavd/pulse_audio_enc: fix error check [00:10] <saste> http://linux-fbdev.sourceforge.net/ -> status [00:37] <cone-893> ffmpeg.git 03ÞSÊ 07master:4bfdd021c71b: configure: remove \r from cc_ident, [01:00] <cone-893> ffmpeg.git 03Michael Niedermayer 07master:12e66f205669: MAINTAINERS: add Timothy to documentation [02:17] <BBB> ubitux: I think it's because the multiplier was 16 bit (i.e. >=32768) so I wasn't sure if it would work correctly (sign-wise) with pmaddwd, which expects signed input [02:18] <BBB> ubitux: so to fix that, I *2 the input and /2 the other input to have it work out [02:18] <BBB> since a*x is the same as (a+a)*(x/2) if x is even, or (a+a)*(x/2)+a if x is odd [02:18] <BBB> I think x was even [02:19] <BBB> so then it's simple [03:41] <mathstuf> ok...so i found out where the STRINGS_METADATA goes when a codec attaches it [03:42] <mathstuf> (av_packet_merge_side_data in avpacket.c) [03:43] <mathstuf> though i dont see how the app is supposed to get that data back easily... [03:43] <mathstuf> i guess test the last 64 bits for FF_MERGE_MARKER? [03:44] <mathstuf> ah, av_packet_split_data [04:40] <sirloin1> Hi guys I think I stumbled upon a bug in ffmpeg, but I was wondering if I could get someone's opinion. It has to do with RTSP session ID. I noticed it doesn't seem to handle the full charset from the RFC [04:41] <sirloin1> oops not a ffmpeg bug i think... sorry [11:13] <cone-239> ffmpeg.git 03Luca Barbato 07master:2b72f8ac320c: wtv: Seek by sector properly [11:13] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:5d2a83571b0a: avformat/wtv: rename to wtv_common [11:13] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:2a5f4c1ea2c7: Merge remote-tracking branch 'qatar/master' [11:41] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:fec6d547cc27: avformat/wtvdec: drop SHIFT_SECTOR_BITS() macro [12:37] <ubitux> michaelni_: i though the purpose of this was the int64_t cast [12:37] <ubitux> otherwise it's 32-bit wide and fails with big files [12:38] <michaelni> ubitux, yes [12:38] <ubitux> but i guess 1<<12 is ok, and sector is already int64 mmh [12:50] <wm4> why do some source files have an av prefix, and some don't? is this just a general inconsistency? [15:14] <Daemon404> michaelni, ping pthread patch when you have time [15:20] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:a6388616e826: doc/developer: Add a policy item about updating the MAINTAINERs file [16:49] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:d5ec8ba7f2cb: Do not leave positive values undefined when negative are defined as error [16:52] <ubitux> this will break some applications randomly if you start changing the behaviour [17:04] <wm4> ubitux: does packet side data need to be verified or not? [17:05] <wm4> tried to get an answer about this out of the Libav devs, but couldn't get a clear answer [17:05] <ubitux> verified? [17:05] <ubitux> verify what? [17:07] <wm4> the contents... my question is whether side data is always internally generated and stored (so you can be sure it's in the form it's required to be), or if it could come from external files and has to be checked in order to avoid security issues [17:09] <ubitux> side data content is sometimes extracted as "raw" from the input [17:09] <ubitux> typically identifier or settings from webvtt [17:09] <wm4> which means? [17:09] <wm4> so it depends on the side data type? [17:09] <ubitux> so if you plan to printf(side_data), i could generated a webvtt with a "%s" in it and make you crash [17:10] <wm4> so it's mixed? [17:10] <wm4> what about AV_PKT_DATA_STRINGS_METADATA specifically? [17:10] <ubitux> likely controlable by the user as well [17:11] <ubitux> some side data should be safe, but not all of them [17:11] <wm4> comical [17:11] <ubitux> but it should be properly formatted [17:11] <ubitux> if that's what you mean by verifying [17:12] <ubitux> there should be no direct control on the layout by the user [17:12] <wm4> user=? API user, application user? [17:12] <ubitux> application user [17:12] <ubitux> for api user i would guess they can do whatever they want anyway [17:13] <ubitux> and i was thinking of crafted files [17:13] <wm4> the question is whether they're supposed to... [17:13] <ubitux> i don't think an input file can cause the generation of a side data not following the layout [17:13] <ubitux> but note that in our code base we actually do test if the size is the expected one for instance [17:14] <wm4> another question, why is side data not just memcpy() of a struct? [17:14] <wm4> (in general, of course it wouldn't work with AV_PKT_DATA_STRINGS_METADATA) [17:14] <ubitux> endianess maybe [17:14] <wm4> lu_zero has been saying something about serializing [17:14] <wm4> different endianess on the same machine, same process? [17:15] <wm4> while we're at it, what does av_packet_get_side_data do? it's undcoumented [17:15] <ubitux> what if we avio_write the side data to an output file [17:15] <wm4> man [17:15] <ubitux> (in a muxer context) [17:15] <wm4> so what if you avio_read it [17:16] <wm4> do you verify it after reading? [17:16] <wm4> or does the API user have to do that? [17:16] <ubitux> i can't tell from memory [17:16] <wm4> and if it's per side-data, it should be documented [17:17] <wm4> well, _especially_ in this case [17:20] <wm4> I still don't get this side data split/merge stuff [17:20] <wm4> must be something completely hysteric [17:21] <wm4> if the packet data contains a magic value (FF_MERGE_MARKER) at some random looking position, the packet is considered to contain side data? [17:21] <wm4> if (!pkt->side_data_elems && pkt->size >12 && AV_RB64(pkt->data + pkt->size - 8) == FF_MERGE_MARKER){ [17:22] <wm4> (so, at the end of the packet) [17:24] <wm4> and whether to use this shit is decided by the "keepside" avoption flag [17:24] <wm4> wat [17:28] <wm4> ok so it seems libavcodec always calls av_packet_split_side_data() on each packet [17:28] <wm4> so at least libavcodec itself has to verify side data always [17:29] <michaelni> ubitux, do you suggest that master:d5ec8ba7f2cb should be reverted or something else or i misunderstood ? [17:30] <wm4> and I found the commit which added this shit: 94eadee7efc2c5d19ecfe92d36f0556663468080 [17:30] <wm4> with no explanation whatsoever [17:30] <wm4> so, does anyone even know why av_packet_split_side_data() is called? [17:30] <wm4> and I fully expect the answer to be "nut" [17:31] <wm4> while possibly subtly breaking some other formats, or adding security issues [17:31] <wm4> unless I'm hopefully misunderstanding this [17:32] <nevcairiel> the only possible breakage would be if some format contains this 64-bit marker at a specific position, right? [17:33] <wm4> nevcairiel: at the end of the packet [17:33] <michaelni> software that isnt avcodec based likely doesnt use AVPackets and alot of it only transports data/size from demuxer to decoder, encoder to muxer, demuxer to muxer, so putting sidedata in data simplifies integration alot for such apps [17:33] <wm4> #define FF_MERGE_MARKER 0x8c4d9d108e25e9feULL [17:34] <wm4> so that software is supposed to use av_packet_merge_side_data() on the demuxer side? [17:34] <michaelni> also it should allow generic containers to preserve side data if they just store and load packets [17:34] <nevcairiel> hitting exactly that 64-bit marker seems rather unlikely, but i agree that its a terribly ugly thing to do, especially if you only use avformat and not avcodec, you can end up with junk data in your packets that you feed to external decoders - if you don't disable it [17:34] <wm4> ok so my conclusion of this mess is: be as defensive as possible when touching side data [17:35] <michaelni> apps that dont want this merge stuff can set AVFMT_FLAG_KEEP_SIDE_DATA [17:36] <wm4> isn't that default [17:37] <michaelni> no [17:37] <Daemon404> hey nevcairiel is bac [17:39] <wm4> so is this a security bug? http://git.videolan.org/?p=ffmpeg.git;a=blob;f=libavcodec/utils.c;h=3832b81… [17:39] <wm4> it basically calls strlen() on packet data [17:40] <wm4> you'd just have to prepare an evil file with FF_MERGE_MARKER at the end of packet data, and then you can at least get ffmpeg to crash? [17:40] <wm4> or am I getting this wrong [17:44] <michaelni> any demuxer that supports loading generic side data would allow the same, you dont need FF_MERGE_MARKER, but yes this code looks problematic [17:45] <michaelni> side data should not be trusted more than data [18:00] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:240fd8c96f59: avcodec/avpacket/av_packet_split_side_data: ensure that side data padding is initialized [18:00] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:838f461b0716: avcodec/utils: add some saftey checks to add_metadata_from_side_data() [18:08] <nevcairiel> Daemon404: kinda, still need to sleep like 16 hours to recover from the flight =P [18:32] <kierank> nevcairiel: are you at gsoc con as well? [19:10] <ubitux> michaelni: no i'm not saying it should be reverted since it should not be wrong currently, but i think it's risky because if you are more likely to change the behaviour ("it's ok since it now matches the doc") [19:17] <michaelni> yes, what do you suggest ? [19:22] <ubitux> being careful not to change the behaviour [19:22] <ubitux> even if it matches the doc& [19:23] <beastd> at least for new functions we should always document it that way (regardless of the behaviour if the function only returns 0 for success at that time) [19:23] <ubitux> sure [19:23] <beastd> libav* callers should not check the result of those kind of functions with if (rc == 0)... [19:24] <beastd> ubitux: ack, to being careful to not change functions that were (initially) documented that way [19:27] <beastd> if the desire arises to change one of those we should probably add a new symbol and retain the old one with the return-0-on-success behaviour [19:48] <kierank> Daemon404: lol vp9 presenting at smpte [19:59] <ubitux> smjd: ping [19:59] <smjd> o_o [19:59] <smjd> pong [19:59] <ubitux> smjd: i looked at the gif thing [19:59] <ubitux> are you the guy from the ticket? [19:59] <smjd> no [19:59] <ubitux> ok [20:00] <ubitux> anyway, i have a working patch here [20:00] <ubitux> but i wonder how you can have correct timing with that method [20:00] <ubitux> (even before the changes) [20:01] <ubitux> well anyway, i'll send a patch soon so you can test if you are interested [20:02] <smjd> I can test it, but I don't need to do any GIF stuff now [20:02] <smjd> does it need testing? [20:03] <smjd> by now, I mean at the moment [20:03] <ubitux> probably, dunno [20:04] <ubitux> "soon" :p [20:04] <ubitux> smjd: how do you set the timing with gif sicle? [20:04] <ubitux> do you know if it's using the delay at the end of each gif when available? [20:07] <smjd> ubitux: the man page says --delay <ms> [20:09] <ubitux> smjd: http://b.pkh.me/0001-avformat-image2-allow-muxing-gif-files.patch [20:09] <ubitux> feel free to try this out [20:09] <ubitux> still a WIP, i need to do proper testing (it's mostly untested currently) [20:10] <ubitux> ./ffmpeg -i <input> -frames:v 150 -c:v gif -f image2 'out%d.gif' [20:15] <smjd> I'll test it in a few hours [20:16] <BBB> ubitux: how's 4x4idct for vp9? [20:17] <ubitux> BBB: didn't progress today, this gif thing is considered a regression (of which i'm responsible) [20:52] <ubitux> ok, back to vp9 now :p [21:40] <Daemon404> kierank, fun [23:01] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:6838e1f547d4: avformat/oggdec: favor av_freep() over av_free() [23:24] <michaelni> ubitux, want to take a look at CID1108581 (ffprobe potential strchr(NULL)) ? [23:25] <ubitux> later maybe, i really want to get done with vp9 [23:26] <michaelni> sure no hurry, ill ask sate or do it myself [23:33] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:d0812f91c810: avcodec/exr: favor av_freep() over av_free() for saftey [23:33] <cone-239> ffmpeg.git 03Michael Niedermayer 07master:80b1e1c03d26: avcodec/exr: fix null pointer dereference [00:00] --- Sun Oct 20 2013

1 0

ffmpeg.log.20131019
by burek 20 Oct '13

20 Oct '13

[00:20] <teratorn> plm: ffmpeg -f lavfi -i testsrc=size=320x240 -t 10 test.mp4 [01:33] <TSM2> has anyone run benchmarks with the new ffmpeg with opencl? [01:42] <buu> uh [01:42] <buu> How do I pass codec options? [01:42] <buu> -vbr on; ? [05:02] <buu> Has anyone see an issue where encoding with -codec opus to a .ogg file ignores the specified -b:a setting? [05:02] <buu> I specify -b:a 64k; and get 93k encodings [06:48] <kriskropd> Can anyone offer a suggestion to work around an error like this? '[mpeg1video @ 0xae8660] MPEG1/2 does not support 10/1 fps' - I'm trying to convert .flv to .mpg [06:54] <relaxed> kriskropd: it means you need to change the framerate [06:54] <relaxed> try -r 25 [06:55] <relaxed> actually it says, Supported framerates: 24000/1001 24/1 25/1 30000/1001 30/1 50/1 60000/1001 60/1 15/1 5/1 10/1 12/1 15/1 [06:56] <kriskropd> relaxed: I expected I needed to rencode themt o something else at first :x - thanks, that worked [14:48] <brontosaurusrex> converting I-frames only video to I-frames only output, is it possible to close encoding every N minutes, those generating multiple files, if yes, how? [14:49] <brontosaurusrex> (The idea is to start editing the converted video while conversion is still in process) [14:54] <brontosaurusrex> I'll be happy with "not possible" as well ... [17:05] <MarcWeber> WHich is the best way to get commercial support on how to setup ffserver to distribute a mpegts stream? [17:06] <MarcWeber> Everything is fine watching from vlc, but a dreambox 800 cannot view it whereas the original stream works fine. So I guess that I didn't setup ffserver properly yet. [18:23] <izemize> hy [18:24] <izemize> i use this command: /usr/bin/avconv -loglevel verbose -i "rtmp://127.0.0.1/deejay/2" -c:v libx264 -profile:v main -force_key_frames "expr:gte(t,n_forced*1)" -c:a libfaac -ac 2 -ar 44100 -f flv "rtmp://0.0.0.0:1935/mpd/2" [18:24] <izemize> but no video output [18:24] <izemize> anyone have good idea / solution? [18:25] <sacarasc> #libav is the channel for avconv. [18:25] <izemize> sorry [18:29] <beastd> izemize: if you get and try with ffmpeg (if possible latest git master) you can of course come here for support [20:10] <SirCmpwn> how do I control the number of cores ffmpeg can use? [20:13] <buu> SirCmpwn: -threads ? [20:14] <SirCmpwn> that's what I want [20:14] <SirCmpwn> thanks [23:35] <brontosaurusrex> converting I-frames only video to I-frames only output, is it possible to close encoding every N minutes, those generating multiple files, if yes, how? [23:35] <brontosaurusrex> (The idea is to start editing the converted video while conversion is still in process) [00:00] --- Sun Oct 20 2013

1 0

ffmpeg-devel.log.20131018
by burek 19 Oct '13

19 Oct '13

[08:57] <Neel> hello [14:12] <durandal_1707> avio_seek cant use SEEK_END? [15:18] <cone-893> ffmpeg.git 03Paul B Mahol 07master:a6881765358b: avformat/flvdec: use avpriv_request_sample() [15:49] <cone-893> ffmpeg.git 03Carl Eugen Hoyos 07master:5b9f39860d83: Support 4k AVC-Intra in mov. [16:53] <cone-893> ffmpeg.git 03Carl Eugen Hoyos 07master:10c240a08593: avcodec: Add little-endian G726 decoder [16:53] <cone-893> ffmpeg.git 03Carl Eugen Hoyos 07master:640c70ddc2d9: avformat: Use G726LE decoder for Sun AU files [17:20] <Compn> the great g726 patch has been applied! [17:20] <Compn> :P [17:20] <Compn> quite the bikeshed on that one [17:27] <durandal_1707> great? [17:28] Action: Compn trolls durandal_1707 [17:39] <iive> so, what was the final color? [17:43] <durandal_1707> it is not about color [17:45] <Compn> iive : it was about what endianness it should be [17:46] <Compn> and the shed would look better or worse if the wrong one was used [17:46] <Compn> the old codecs are crazy :\ [17:47] <{V}> sheds have endianness? no wonder I'm having trouble storing my bike [17:47] <Daemon404> you merely need an endian-agnostic bike [17:49] <ubitux> i guess sox has a different (default?) endianess [17:49] <ubitux> ? [17:49] <iive> ok, front wheel first or back wheel first? [17:49] <iive> ;) [17:50] <iive> maybe I should take a look of the thread. But generally the output should be in the native format of the codec. Meaning, not doing additional conversion. [17:51] <durandal_1707> some encoders just changed order [17:52] <durandal_1707> imagine someone do this with flac.... [17:52] <iive> well, there won't be a way to detect it, would it? [17:52] <Daemon404> you could use a heuristic pretty well [17:53] <Daemon404> but it would be slow [17:53] <Daemon404> and error prone [17:53] <iive> it is not the job of the decoder to make guesses. [17:54] <iive> is g726 looseless? [17:54] <Daemon404> if the format itself has no endianness, it's 'wrong' of us output it as "BE" or "LE" [17:55] <iive> you will have to output it, eventually. [18:14] <iive> ok, so g726 is mac containers is BE and in other containers is in LE format. carl proposes a patch that creates a new decoder that outputs in a different endianes. Or to be exact, same decoder just label the output data as different format. [18:15] <iive> it is always better to use a reliable method than making wild guesses. [18:15] <iive> as of should it be different fourcc, codec_id or flag in extra-data... it's just a matter of preference. [18:16] <Compn> the bikeshed, it continues! [18:25] <iive> of course... it must be in blue. [18:37] <michaelni> ubitux, the following 2 subtitle patches seem to have no reply, dunno if they are still relevant but if they are they should get some kind of agree/disagree/obsolete/... reply "0808 17:03 Nicolas George (5.4K) [18:50] <wm4> michaelni: that's nicolas' charset non-sense, I hope it won't ever make it [18:50] <wm4> not in this form [18:50] <durandal_1707> reply with: rejected [18:50] <wm4> good intentions, horrible outcome [18:51] <Compn> wm4 : did you look at #libav-devel ban list? :P [18:51] <wm4> aha [18:51] <wm4> fun [18:51] <durandal_1707> who is on that list? [18:51] <wm4> mini [18:52] <wm4> the ban hostmask doesn't even match anymore [18:52] <Compn> wm4 : do you know if libav is working on subs? i saw elenril ask about it [18:52] <wm4> no, libav gives a shit about subs and it shows [18:53] <Compn> missing a word in there [18:53] <wm4> what word [18:53] <Compn> they give a shit, or they dont give a shit ? [18:54] <michaelni> I think someone should talk with nicolas about the subtitle/charset stuff, i mean maybe not ML and maybe not email but not sure [18:54] <durandal_1707> in rl? [18:55] <Compn> what country is he in ? [18:55] <ubitux> fr [18:55] <wm4> oh right [18:55] <wm4> +don't [18:55] <ubitux> michaelni: i've not that much clue on what should be done for charset honestly [18:55] <Compn> michaelni : your idea to automatically guess charset ? [18:55] <wm4> it needs utf-16 support [18:56] <ubitux> yes right [18:56] <Compn> only seen one utf16 sample? [18:56] <wm4> the subtitle parsers can read any encoding just fine, and the application can detect and convert it [18:56] <wm4> except utf-16, because that is not an ascii superset [18:56] <Compn> so whats missing? just a way to select the charset? or the entire iconv conversion ? [18:56] <wm4> but nicolas wants to add some naive super broken and enforced charset detection [18:57] <wm4> (and conversion) [18:57] <Compn> wm4 : well thats because ffmpeg is both a program and a set of libraries. the 'controlling application' in this case would be ffmpeg [18:57] <Compn> so that feature would be useful for ffmpeg (the application) [18:58] <durandal_1707> who and why put stuff on our banlist? [18:58] <wm4> another case where ffmpeg (the application) causes sub-par stuff to be added to the libraries [18:58] <michaelni> Compn, ubitux i dunno subtitle stuff that well either, so iam not ideal for choosing a design [18:58] <Compn> durandal_1707 : one guy was spammer, dunno bout the other bans [18:58] <wm4> lol I'm probably the only one who uses the subtitle stuff (and not from ffmpeg.c/ffplay.c) [18:59] <Compn> durandal_1707 : can check irc logs, might bei n there [18:59] <Compn> yes i dont think vlc or mplayer use subtitle stuff yet [19:00] <Compn> should ask j-b what , if anything, he wants from lavf subtitle [19:00] <Compn> wm4 : probably you are the only user so far :D [19:00] <wm4> feels bad [19:00] <ubitux> :) [19:00] <ubitux> that's cool wm4! [19:00] <ubitux> :) [19:00] <j-b> well parsing subtitles is easy. Rendering is very very hard [19:00] <ubitux> you helped making vobsub demuxer better ;) [19:01] <Compn> j-b :want subtitle render in video filter ? [19:01] <wm4> j-b: libass ftw [19:01] <j-b> well, libass is a bit slow, and the fontcache, man! [19:01] <wm4> it also has a nice library name, providing for endless fun puns [19:01] <wm4> it's slow? [19:02] <wm4> but yeah... fontconfig [19:03] <durandal_1707> 75$ for specification [19:04] <Compn> j-b : or do you want some other way to render subs ? [19:04] <durandal_1707> yes via vlc [19:04] <Compn> wm4 : how do you render subs ? [19:04] <Compn> i mean [19:04] <wm4> Compn: libass [19:04] <Compn> do you render subs without libass ? [19:05] <wm4> no [19:05] <Compn> like mplayer has its old renderer... [19:05] <Compn> ok [19:05] <wm4> the old renderer was horrible [19:05] <wm4> like everything in mplayer, really [19:05] <Compn> it was a lot of hacks to deal with crappy 2001 hardware [19:05] <Compn> voodoo cards [19:05] <Compn> you werent around for the 3dfx cards wm4, you dont know how bad video was back then [19:05] <j-b> Compn: the issue is that it needs harfbuzz+fribidi+freetype+font-lookups to properly render subs [19:06] <j-b> Compn: and the only ones who do have that are gtk, qt or webbrowsers [19:06] <Compn> fribidi :\ [19:07] <durandal_1707> saste: write ripple filter? [19:08] <wm4> there was some talk (and even code) to use native APIs in libass (on windows/osx), but, it's probably well possible, but it went nowhere [19:10] <Compn> so we should just use libass until an ffmpeg wheel can be reinvented ? [19:10] <Compn> since it has fribidi and freetype and fontcache [19:13] <wm4> ffmpeg already has vf_ass, but of course it doesn't work the way you'd expect (you have to provide an external subtitle file to vf_ass directly) [19:14] <durandal_1707> and there is no subtitle filtering [19:15] <ubitux> and there is not really anyone doing relevant work on sub :( [19:16] <durandal_1707> or anything else [19:16] <Compn> wm4 : do you want it to automatically use embedded subs ? [19:16] <Compn> i mean, that seems like a simple addition [19:16] <Compn> we can shoot for that :P [19:17] <ubitux> durandal_1707: i'm in a learning stage, i'll be back to productive mode later [19:17] <wm4> bikeshedding, dude [19:17] <Compn> j-b : lavfi in vlc, when ? :) [19:17] <j-b> Compn: when there is a need? [19:17] <durandal_1707> vlc have own filters [19:18] <j-b> MOAR filters is always better [19:18] <Compn> j-b : yadif? :P [19:19] <j-b> Compn: already in. [19:19] <Compn> ok :) [19:19] <j-b> Compn: we have adjust, gradfun, yadif, some grain adder, hqdn3d, IVTC (and so many deinterlacer) [19:20] <j-b> I don't know what people need in addition to that [19:20] <ubitux> mandelbrot [19:21] <j-b> ? [19:21] <ubitux> j-b: ffplay -f lavfi mandelbrot [19:21] <wm4> maybe vf_pp7? [19:22] <j-b> ubitux: useless, it seems :) [19:22] <Compn> wm4 : someone ported that ? [19:22] <ubitux> j-b: you're always so boring :( [19:22] <wm4> Compn: it's already ported! it works with vf_mp! [19:22] <Compn> ehe [19:22] <j-b> ubitux: right... Did you ever launch VLC lately? [19:22] <j-b> wm4: what is pp7 ? [19:22] <ubitux> yes [19:23] <j-b> ubitux: 2.1.0 ? [19:23] <ubitux> but only through cvlc [19:23] <wm4> j-b: some crap nobody wants to port [19:23] <wm4> so the (very unclean) libmpcodecs wrapper stays in lavfi [19:23] <ubitux> j-b: yes that's my current version [19:23] <j-b> cvlc --video-filter puzzle [19:23] <j-b> cvlc --video-filter puzzle <input> [19:24] <ubitux> :D [19:24] <j-b> use your mouse know :) [19:24] <wm4> useful... should be ported to lavfi ASAP [19:24] <ubitux> oh shit it's interactive [19:24] <j-b> ubitux: you can use the rotation version too [19:24] <ubitux> it's extremely slow though [19:24] <j-b> so that a click or mouse rotates +/-90 [19:25] <j-b> ubitux: slow? no shit! [19:25] <ubitux> please optimize this [19:25] <j-b> riite [19:25] <ubitux> that will be the only reason i'll use vlc now [19:25] <j-b> well, VLC 2.2 has vhs effect [19:25] <Compn> j-b : wow thats a fun feature :) [19:25] <Compn> video puzzle [19:26] <j-b> Compn: the hard way is that one: [19:26] <j-b> cvlc --video-filter puzzle -V caca <input> [19:26] <Compn> caca :D [19:26] <j-b> it actually worsk [19:26] <j-b> but interactivity is limited [19:27] <durandal_1707> what deinterlacers? [19:27] <j-b> ubitux: 2.2 has vhs and oldmovies effects. You said boooring? [19:27] <wm4> why does cvlc not close when closing it with the window manager's close button? [19:27] <ubitux> j-b: :D [19:27] <j-b> wm4: because of a bug on VLC xv output, IIRc [19:28] <j-b> durandal_1707: that was a question for me? [19:28] <Compn> wm4 : so no downstream wants to use ffmpeg subtitle options. [19:28] <Compn> wm4 : so , we focus on ffmpeg subtitle for ffmpeg application only. [19:28] <wm4> the default video driver doesn't seem to implement the required X protocol (so that the application closes itself on request) while caca simply ignores it [19:28] <ubitux> oh i can reshuffle when i'm done [19:29] <Compn> caca is laggy in win32 :) [19:29] <Compn> vhs video effects? i love vhs... [19:30] <j-b> ubitux: note that there are 4 modes [19:30] <j-b> ubitux: 0 (jigsaw puzzle), 1 (sliding puzzle), 2 (swap puzzle), 3 (exchange puzzle) [19:30] <wm4> I certainly find it interesting that video filters can capture mouse input [19:31] <ubitux> j-b: how do you set it? [19:31] <wm4> what if several things want input? [19:31] <Compn> wm4 : isnt it like dvd menu ? [19:31] <j-b> --puzzle-mode <i> [19:31] <j-b> wm4: --video-filter zoom is interactive too [19:32] <j-b> maybe it is called magnify now, though [19:32] <wm4> it's called magnify [19:33] <Compn> thats neat [19:33] <Compn> people were asking how to make real pan and scan versions of films [19:34] <durandal_1707> you mean zoom & pan? [19:34] <j-b> swscale ftw [19:35] <Compn> durandal_1707 : i dont know what they call it in other countries, here its always p&s , make a 4:3 film from a 16:9 film by cropping and panning [19:35] <j-b> ubitux: and for more useless thing, vlc --video-filter psychedelic [19:35] <durandal_1707> what that does? [19:36] <j-b> I cannot explain [19:36] <durandal_1707> Compn: that can be done with lavfi [19:36] <j-b> It's a LSD simulator [19:36] <wm4> lavfi still has a long way to go [19:36] <ubitux> mmh [19:36] <durandal_1707> wm4: meh meh [19:36] <ubitux> i'm disappointed j-b [19:36] <j-b> ubitux: wut? [19:36] <wm4> durandal_1707: does lavfi have an lsd simulator? [19:36] <durandal_1707> geq [19:36] <cone-893> ffmpeg.git 03Carl Eugen Hoyos 07master:6e65e3483706: lavf/riff: Fix a comment. [19:36] <ubitux> j-b: it doesn't messed up the sound [19:36] <j-b> ubitux: lol [19:37] <ubitux> j-b: also, i think you can do the same with lavfi actually [19:37] <j-b> ubitux: tbh, the biggest mess with all this, is to decide what are the ones that are important [19:37] <j-b> ubitux: of course. [19:37] <ubitux> with hue and an eval [19:37] <ubitux> and for the moving one, should be doable as well [19:37] <j-b> is a grain filter more or less important than a denoiser? [19:38] <Compn> j-b : i get usermails to webmaster@mphq talking about getting viruses from downloading mplayer on trojan sites. what do you tell vlc users who get same problem ? :( [19:38] <j-b> what denoiser should be the most important? [19:38] <j-b> hqdn3d, gaussianblur, or temporal blur? [19:38] <ubitux> you're gonna make Daemon404 angry [19:39] <j-b> Compn: You don't imagine how long the answer to this question is going to be. [19:39] <Compn> j-b : dont you have automated form email reply yet? ::) [19:39] <Compn> i could just copy / replace vlc with 'mplayer' and send. :) [19:40] <Daemon404> gaussian blur is considered a denoiser? LOL [19:40] <j-b> no, because it is not there fault [19:40] <j-b> ubitux: it was done a bit on purpose, :) [19:40] <Compn> its the wild west of the internet fault [19:40] <Compn> but i like the wild west so , on it goes [20:12] <durandal_1707> phosphor deinterlacer? [20:16] <durandal_1707> there are bunch of nice (toy) filters to port from vlc, instead of libmpcodecs crap [20:16] <cbsrobot> durandal_1707: you could add a mda filter - if lsd is too boring [20:19] <durandal_1707> cbsrobot: mda? [20:19] <cbsrobot> durandal_1707: http://en.wikipedia.org/wiki/3,4-Methylenedioxyamphetamine [20:19] <durandal_1707> that is not cheap enough [20:20] Action: cbsrobot always reads durandal_1707's email as "one" "mda" at gmail [20:20] <cbsrobot> never thought it could mean something else ... [20:27] <durandal_1707> i have bunch of filters not applied because of bikeshed [21:27] <cone-893> ffmpeg.git 03Carl Eugen Hoyos 07master:9860fb445a90: Fix decoding of G.726 in AIFF. [21:54] <ubitux> http://xiphmont.livejournal.com/61334.html #daala, part4 [21:55] <durandal11707> and? anything? [23:15] <cone-893> ffmpeg.git 03Stefano Sabatini 07master:8efed2993733: MAINTAINERS: add my name to a few entries [23:41] <cone-893> ffmpeg.git 03Lukasz Marek 07master:1421ee263742: lavd/fbdev: pass proper memory map length [23:41] <cone-893> ffmpeg.git 03Lukasz Marek 07master:fb74c7e0446f: lavd/pulse_audio_dec: apply cosmetic changes [23:41] <cone-893> ffmpeg.git 03Stefano Sabatini 07master:f6b56b1f26a1: lavd/fbdev: use av_str2err() macro for printing error messages [00:00] --- Sat Oct 19 2013

1 0

ffmpeg.log.20131018
by burek 19 Oct '13

19 Oct '13

[01:57] <_8680_> Concatenating audio (.wav) files with `ffmpeg -f concat -i input-file-list -codec copy output-file` spews hundreds of warnings for me: <https://dpaste.de/CmV6>. Is ignoring these warnings safe? If not, what am I doing incorrectly? [03:22] <SarBear> hi [03:23] <SarBear> can ffmpeg be used to extract meta data from an mpeg-2 file? If not, do you know of any tools that can? [05:23] <kriskropd> so, I tried to get ffmpeg to concat videos again, this time I used some mp4 videos from youtube and noticed I was having the same problem as before: when I concat, only the first video is placed into the output video file [05:37] <kriskropd> oh, I got it :D for somereason, it won't work if I use the shell,b ut if I use a text file and list each file as "file '00.flv'" the ffmpeg concat works - this worked with the videos I was trying to concat earlier, if anyone here was here earlier for my conundrum [05:50] <kriskropd> ick, nwo i need to figure out why it's breaking between each video (I'm noticign some videos are 800x600 while others are 480x360 - what a pain [05:50] <kriskropd> now* [05:51] <relaxed> kriskropd: use MP4Box to concat mp4s, MP4Box -cat 1.mp4 -cat 2.mp4 -new combined.mp4 [06:39] <kriskropd> relaxed: thanks for the suggestion - appears to have the same problem though [06:44] <relaxed> kriskropd: well, if they have different resolutions it won't work [06:46] <kriskropd> relaxed: yeah i gotta learn how to re-encode or transcode in my script it seems - anyways, i'me tired of using my free time on this today, im off to bed [11:12] <guest35324165> hi [11:14] <pjetr> is there a way to keep the quality of an input video and just transcode it to another filetype? [11:15] <pjetr> I'm on a mac and just did: `ffmpeg -i in.mov out.avi` this encoded my 908M mov to an 1M avi. and the quality was horrible [11:16] <spaam> you want to change the container ? [11:16] <spaam> or change video codec? [11:16] <spaam> or what? [11:16] <spaam> maybe both? [11:16] <spaam> i think the default value is 200kbps [11:17] <spaam> and your .mov file is much more then that :) [11:17] <pjetr> maybe both, I'm not completely certain. [11:18] <pjetr> I exported the .mov from flash, when I want to convert a mov from flash to HTML-video, I get nothing. (transcoded on a Ubuntu server) [11:19] <pjetr> but if i convert it first using "Adobe Media Encoder" it convert's without a problem. But it's kinda difficult to automate that as a process [11:19] <spaam> html5-video need to be h264/aac in .mp4 :) [11:20] <pjetr> and ogg and webm [11:20] <spaam> yes :) [11:21] <pjetr> but now I'm just tooling around to see if I can convert my source file to something that the server can encode [11:21] <pjetr> So I wanted to convert it to a lossless AVI [11:27] <pjetr> and it's safe to say that I don't really know what I'm doing, or what most of the options mean [14:12] <luc4> Hi! Is there any guide on how to stream using http using the APIs? Not using ffserver. I had a quick look at ffserver and it seems pretty complex, maybe there is some tutorial or something else that can simplify the process? [14:15] <Mavrik> wel [14:15] <Mavrik> ffserver is about as simple as it gets [14:15] <Mavrik> what is your usecase exactly? [14:15] <Mavrik> and which formats would you like to stream? [14:16] <luc4> Mavrik: I'm not an expert of video streaming, I would like to stream mjpeg and h264 at the moment. [14:17] <Mavrik> in what container to what? [14:17] <Mavrik> what audio if any? [14:17] <luc4> Mavrik: no guide or similar to get an idea before digging into ffserver code? [14:17] <luc4> no audio, if possible, no container [14:18] <Mavrik> there's no generic "stream everything" guide [14:18] <luc4> simple mjpeg frames and h264 bytes [14:18] <Mavrik> you MIGHT find one if you know what do you want to achieve [14:18] <Mavrik> you can't really stream raw H.264 without a container around [14:18] <Mavrik> nothing's gonna play that [14:18] <Mavrik> so again, what are you streaming to_ [14:18] <Mavrik> ? [14:19] <luc4> myself... another application [14:19] <luc4> But I see your point, just didn't know. And do you also know for mjpeg? [14:19] <luc4> Do I need a container for that as well? [14:19] <Mavrik> mjpeg is just a set of JPEG images [14:20] <luc4> exactly [14:20] <Mavrik> mjpeg already is a container definition ;) [14:20] <luc4> oh what tha... you're right :-) [14:20] <luc4> sorry [14:20] <Mavrik> yep, mjpeg is just a sequence of jpegs [14:21] <Mavrik> there's no official standard for it [14:21] <luc4> Of course, I'm just stupid :-) assuming I want to send the jpegs I encode through http, can I do that with ffmpeg? [14:22] <Mavrik> luc4, yep [14:23] <Mavrik> luc4, of course you'll have to decide which side will be the server [14:23] <Mavrik> ffserver implements full HTTP server to serve data [14:23] <Mavrik> if you want to *SEND* data [14:23] <Mavrik> your target app will have to implement it [14:24] <again_ffm> Mavrik, do you know, ffmpeg can repeat input video? [14:24] <Mavrik> there was a command but it was changed [14:24] <Mavrik> i'm 100% sure it says so in the documentation if repeat is supported for a stream ;) [14:25] <again_ffm> Mavrik, if loop_input than, it just repeats image (img2video) [14:25] <luc4> Marik: ah yes, the code gave me that idea... so I have no other option than doing the entire work? Any tutorial/guide/whatever that you can suggest? Just want to know this before deciding which way to go. And thanks for your help ;-) [14:26] <Mavrik> luc4, well, I don't know how you imagined doing HTTP data transfer without having a HTTP server somewhere :P [14:26] <Mavrik> luc4, are you sure you wouldn't rather buy a Wowza license and use that? [14:26] <Mavrik> and then use ffmpeg to stream to Wowza? [14:27] <Mavrik> or one of the other streaming servers? [14:27] <luc4> Mavrik: I was not imaging a solution without an http server, but possibly that ffmpeg was implementing a minimal itself or maybe some other lib "wrapping" this... just wanted to ask before doing much useless work :-) [14:28] <Mavrik> luc4, mhm [14:29] <Mavrik> luc4, well& as I said, I suggest you find a good streaming server instead of rolling your own [14:29] <Mavrik> if you can't use ffmpeg/ffserver combo as they are now [14:29] <Mavrik> luc4, and then use your lib to create a MPEG2-TS video or something like that and UDP stream it to the streaming server [14:29] <Mavrik> if your target is streaming to multiple clients [14:29] <Mavrik> if you just want to transport video from one point to another [14:30] <again_ffm> Mavrik, and last q: ffmpeg can join repeated image and video? ffmpeg -i 1.mp4 -loop 1 -i Untitled.png -t 10 -filter_complex "[0] [1] concat=n=2:v=1:a=1 [v] [a]" -map "[v]" -map "[a]" 3.mp4 - say: Stream specifier '' in filtergraph description [0] [1] concat=n=2:v=1:a=1 [v] [a] matches no streams. [14:30] <Mavrik> forget about HTTP and just use UDP [14:31] <Mavrik> again_ffm, that looks like a syntax error in your filter spec [14:33] <luc4> Mavrik: maybe I could use a pipe or something like this to provide data to ffserver? [14:34] <again_ffm> Mavrik, may be but ffmpeg -i 1.mp4 -i 1.mp4 -filter_complex "concat=n=2:v=1:a=1" 3.mp4 works fine [14:34] <luc4> Mavrik: this is only for a local network, internet access is not supposed to be needed. [14:35] <Mavrik> luc4, look, you still haven't described your use case [14:35] <Mavrik> WHO IS THE TARGET? [14:35] <Mavrik> a single app? a browser? 15 people? [14:36] <Mavrik> are you transporting stream over internet or over a cable from one machine to another? [14:36] <Mavrik> how fast is your transport? do you need error correction? [14:36] <Mavrik> there's tens of ways of doing that [14:36] <Mavrik> and unless you know what you want to do we can't really help you :) [14:39] <luc4> Mavrik: stream coming from a camera will be transferred over a wifi link (Internet access is not necessary) to some clients (number is not known but limited to 1-5 devices). The client is already working. I need to implement the server. In my application I already use ffmpeg for other things, so it would be nice to use it again for this. Target platform is Android. http is preferrable to be able to see it on a browser, but other ways if simpler might be good as [14:39] <luc4> well. [14:40] <Mavrik> ah, cool [14:40] <Mavrik> :) [14:40] <Mavrik> luc4, if you want live streaming you'll have to use HLS, not pure HTTP for Androids [14:41] <luc4> "The client is already working": I mean I aready implemented it and wroks correctly. It is also using ffmpeg and it was really satisfying so re-using ffmpeg libs might be a good idea... [14:41] <luc4> Mavrik: yes, I was looking at HLS indeed. [14:41] <Mavrik> I think ffserver doesn't support HLS [14:43] <luc4> Mavrik: what is not clear to me... maybe you know it... is HLS the only way to transfer a video stream using http? [14:43] <luc4> Mavrik: I'm pretty ignorant, sorry. [14:43] <Mavrik> luc4, HLS is pretty much the only way to transfer _LIVE_ (e.g. not stored in a prebuilt file) video to be watched in browser on mobile devices [14:44] <Mavrik> also, HTTP is really shit for video transport [14:44] <Mavrik> so unless you have a requirement of using a browser [14:44] <Mavrik> there's tons of better transports for video than HTTP [14:44] <luc4> Mavrik: thanks for the confirmation. Then ffserver is out of the question :-) still can I use ffmpeg? Maybe rtp? [14:44] <Mavrik> if you DO want to use a browser, HLS is practically your only choice for mobile [14:45] <Mavrik> and flash for desktop [14:45] <luc4> preferrable... [14:46] <luc4> rtp is the other choice right? [14:46] <Mavrik> um [14:46] <Mavrik> not really [14:46] <Mavrik> or, rather I'll say: RTP is one of the worse things you can do to yourself [14:46] <Mavrik> besides jumping into a pool of razors [14:46] <Mavrik> UDP/MPEG2-TS, RTMP, RTSP are all preferrable than dealing with RTP [14:47] <luc4> I see... and I was also reading this: http://sirlagz.net/2012/08/04/how-to-stream-a-webcam-from-the-raspberry-pi/ [14:49] <luc4> This seems pretty interesting, but confusing... that is not hls... so what is it? [14:50] <Mavrik> Format mjpeg [14:51] <luc4> I've been doing some opensource with rasp myself, always using ffmpeg, so I could test on my own pi... but if it is not hls, what is it? [14:52] <Mavrik> it's just sending jpegs over an open socket [14:52] <luc4> Not http? [14:53] <Mavrik> test it [14:53] <Mavrik> does that even work in a browser? [14:55] <luc4> Mavrik: I'll try now :-) the guide says it should work on a browser... but I'll test it :-) [14:58] <Mavrik> luc4, yeah, maybe it even works [14:58] <Mavrik> probably browser keeps HTTP connection open and just redraws the frame [14:58] <luc4> Mavrik: I'm goind to tell you :-) [15:01] <luc4> Mavrik: nope... [http @ 0x8ac3f20] HTTP error 404 Not Found. Maybe my version of ffmpeg is too old... [15:01] <luc4> Mavrik: I'm testing on Ubuntu right now anyway... [15:13] <luc4> Mavrik: maybe something recently implemented? [15:13] <luc4> Mavrik: http://ffmpeg.gusari.org/viewtopic.php?f=12&t=914 [15:14] <pyBlob> I'm running this http://pastebin.com/aYVBMAFn command to capture from my webcam to a file and at the same time displaying it using ffplay [15:15] <pyBlob> that works ... only problem is that the video in the file is too fast+crippled when viewing it [15:15] <pyBlob> any ideas? [15:20] <pyBlob> oh ... I think I know the probelm [15:25] <pyBlob> the FILE<&1 didn't work as expected [15:29] <sacarasc> Try using tee, pyBlob. [15:29] <pyBlob> I had problems with tee ... so I wanted to try something new, which failed too xD [19:24] <smjd> how do I mute audio in ffplay? [19:33] <olspookishmagus> smjd: if you're not refering to when it's playing you can use the -an switch [19:33] <smjd> I managed to do it with pavucontrol [23:45] <plm> HI all [23:46] <plm> people, where I found samples of videos.. I would like 320x240 samples, any format, .ogg .mp4 [23:57] <saste> plm, why don't you create them youself? [00:00] --- Sat Oct 19 2013

1 0

ffmpeg-devel.log.20131017
by burek 18 Oct '13

18 Oct '13

[00:04] <j-b> smarter: if ever you need work, please let me know :) [00:06] <smarter> I may soon, we'll see ;) [00:15] <kierank> smarter: are you going to be in geneva in november? [00:16] <kierank> smarter: http://tech.ebu.ch/devcon13 [00:16] <kierank> might be interesting [00:16] <smarter> kierank: unless something unexpected happens, yes ;) [00:16] <kierank> you should go [00:16] Action: kierank is going [00:17] <smarter> 100 CHF to register? [00:19] <j-b> cheap [00:30] <saste> llogan: yes they're fast and reliable when they have to pay [00:33] <kierank> smarter: nominal fee i think [00:33] <kierank> for food etc [01:06] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:3ed65d98c616: avutil/log: fix race between setting and using the log callback [01:20] <wm4> michaelni: I doubt the code an optimizing compiler generates is even different before and after the call [01:20] <wm4> s/call/commit [01:21] <michaelni> wm4, i thought so too, but its more correct now [01:22] <wm4> you'd need at least a memory barrier I guess [01:22] <wm4> technically [01:24] <michaelni> yes [01:24] <wm4> also, while we're at it, ffmpeg shouldn't even have a global log callback [01:24] <wm4> what if two libraries loaded into the same process use libav* at the same time? [01:25] <wm4> that's not even unlikely, considering how important ffmpeg is [01:26] <michaelni> libs cant (shouldnt) set the callback currently for that reason [01:26] <Kovensky> "it hurts when I do that" "then don't do that, duh" :> [01:27] <wm4> "currently" [01:27] <wm4> so what about those things that actually require you to parse log output? (lol lavfi) [01:32] <michaelni> well, that either means we should add non global callback support somehow or use something else for these cases than av_log() + user app/lib parsing [02:48] <cone-159> ffmpeg.git 03Mickaël Raulet 07master:b4948943904a: hevc: optimize residual coding(cherry picked from commit 70692a44708157b4dfa50e402e446bfa2b27f55e) [02:48] <cone-159> ffmpeg.git 03Mickaël Raulet 07master:a7e300649a7a: hevc: fix pcm with different chroma luma bit widths(cherry picked from commit 6a444516f338424d062c0ef2806714036283603b) [02:57] <BBB> ubitux: yes, I'm using ssse3 as minimum also [02:57] <BBB> ubitux: too lazy to use sse2 only [02:58] <BBB> ubitux: if you have sse2, use vp8 ;) [08:43] <cone-569> ffmpeg.git 03Martin Storsjö 07master:d433e1aefabd: mem: Make av_strdup allocate using av_realloc [08:43] <cone-569> ffmpeg.git 03Michael Niedermayer 07master:21b3563dcbaa: Merge remote-tracking branch 'qatar/master' [09:53] <mraulet> michaelni: do you want me to crosscheck hevc in ffmpeg? [10:26] <cone-569> ffmpeg.git 03Dirk Farin 07master:2ac31773f3fc: avformat/hevcdec: cosmetics, whitespaces [10:26] <cone-569> ffmpeg.git 03Dirk Farin 07master:56cf6151ae9f: avformat/hevcdec: add more irap cases to hevc_probe() [10:26] <michaelni> mraulet, yes, please do! [10:27] <michaelni> theres 1 file left that doesnt decode correct from the stuff in fate [10:27] <michaelni> fate-hevc-conformance-PPS_A_qualcomm_7 that is [10:41] <durandal_1707> michaelni: there is floating exception for some values of cotrast in swscale [10:48] <durandal_1707> also x264opts is buggy and should be deprecated [10:57] <mraulet> PPS_A_qualcomm_7 still there with one thread [10:59] <cone-569> ffmpeg.git 03Carl Eugen Hoyos 07master:a0d13d84a928: Add some necessary casts in the wtv demuxer. [10:59] <cone-569> ffmpeg.git 03Michael Niedermayer 07master:f90eb8cfc8c4: Merge remote-tracking branch 'cehoyos/master' [11:49] <ubitux> i need a relatively high bitrate clip, any recommendation? [11:49] <ubitux> elephant dreams, sintel, ... ? [11:55] <durandal_1707> anything non-trivial with high motion [14:11] <cone-656> ffmpeg.git 03Michael Niedermayer 07master:2db65472376c: swscale/utils/sws_setColorspaceDetails(): fix indention [14:11] <cone-656> ffmpeg.git 03Michael Niedermayer 07master:d0a3bc13025b: swscale/yuv2rgb: clip cy, avoid division by 0 with 0 contrast [14:38] <SilverCode> hi [14:39] <SilverCode> what would be the easiest way to add a new boolean command line switch that I could access from libavformat/riff.c ? [14:41] <SilverCode> it doesn't need to be clean or extensible, I just need to be able to tell avconv to write a "hacked" size when writing an avi header [14:42] <durandal_1707> ffmpeg does not have avconv [14:42] <SilverCode> are avconv and ffmpeg not the same? [14:42] <durandal_1707> avconv is from libav [14:43] <SilverCode> ah [14:43] <SilverCode> off to libav-devel then [14:43] <SilverCode> thanks [14:48] Action: durandal_1707 away at lunch [15:05] <mraulet> michaelni: it works well for sequences I have [15:06] <mraulet> except for PPS_A_qualcomm_7 that has a weird behavior with one thread [15:09] <michaelni> mraulet, does PPS_A_qualcomm_7 work better with the openhevc code ? [16:01] <mraulet> it was working before [16:01] <mraulet> now we align both code [16:07] <saste> michaelni, are you fine with "[PATCH] lavu/opt: add AV_OPT_TYPE_CHANNEL_LAYOUT and handler functions" ? [16:09] <saste> also the patch doesn't make sense without the swresample patch which changes INT -> channel layout [16:15] <durandal_1707> isn't that breaking API/ABI? [16:17] <saste> durandal_1707, no [16:18] <saste> CHANNEL_LAYOUT options can be set by INT handlers [16:31] <saste> what's the supposed content of an av_opt_get() buffer? [16:39] <michaelni> saste, didnt i already say ok to that patch ? [16:40] <michaelni> or maybe just if wm4 is ok with it or somethig like that [16:42] <saste> michaelni, no you replied about the swscale filter patch [16:43] <saste> also you're maintainer of libswr so the patch may concern you [16:45] <saste> anyway i found a minor issue, will repost it soon [16:45] <michaelni> the only change affecting swr is the change of avopt type and obviously a more specific type is better (if it doesnt break ABi and all that which it doesnt ) [16:45] <saste> michaelni, ok [16:46] <saste> i'm almost done with it, will test more and will push if wm4 has no comments [16:46] <saste> i changed the av_get_channel_layout patch so that no manual tweaking is necessary at the next bump [16:47] <saste> all is handled by the preprocessor (would be otherwise extremely annoying when testing major bumping) [16:49] <saste> wm4? [16:58] <ubitux> wasn't ffv1 encode supposed to be threaded? [16:58] <ubitux> it doesn't seem to make much use of the cpu [17:13] <ubitux> it's too bad, because it's as fast as x264 with qp=0, but it's more than 2 times bigger [17:14] <ubitux> it's still smaller than huffyuv or utvideo, but ~3x slower [17:23] <durandal_1707> use slice threading [17:23] <ubitux> shouldn't it be set by default? [17:24] <durandal_1707> and you can change gop to make frame thread encoder faster [17:24] <durandal_1707> s/decoder [17:25] <durandal_1707> ubitux: compare with threading disabled [17:38] <saste> how can i show swscale or swresample options? [18:06] <durandal_1707> saste: aren't they displayed with ffmpeg? [18:06] <michaelni> ubitux, default ffv1 is using a older version for compatibility with most decoders [18:07] <saste> durandal_1707, yes, i was looking for something like ffmpeg -h scaler [18:24] <cone-186> ffmpeg.git 03Stefano Sabatini 07master:98e7c1eed559: lavu/opt-test: use automatic set and free handlers [18:24] <cone-186> ffmpeg.git 03Stefano Sabatini 07master:b236eb49e1a1: lavu/opt.h: fix grammar typo in av_opt_get* doxy [18:24] <cone-186> ffmpeg.git 03Stefano Sabatini 07master:d96e377c394c: lavu/channel_layout: change av_get_channel_layout() behavior at the next bump [18:24] <cone-186> ffmpeg.git 03Stefano Sabatini 07master:8696e51bafdc: lavu/opt: add AV_OPT_TYPE_CHANNEL_LAYOUT and handler functions [18:24] <cone-186> ffmpeg.git 03Stefano Sabatini 07master:904c89ac1bdc: lswr/swresample: convert ocl and icl options to AV_OPT_TYPE_CHANNEL_LAYOUT [19:02] <ubitux> durandal_1707, michaelni; ah, ok for ffv1 [19:03] <ubitux> i still don't get why threading would be affected by versions [19:03] <ubitux> but ok [19:09] <durandal_1707> ubitux: so how fast are other versions? [19:10] <ubitux> 21 fps 55 fps with -level 2 [19:10] <ubitux> -level 3* [19:10] <ubitux> sorry [19:11] <ubitux> which is pretty cool [19:14] <michaelni> ubitux, also try larger numer for -slices [19:19] <ubitux> 10 fps faster with -slices 16 [19:19] <ubitux> -rw-r--r-- 1 ubitux ubitux 628M Oct 17 19:17 out-ffv1-legacy.mkv [19:19] <ubitux> -rw-r--r-- 1 ubitux ubitux 631M Oct 17 19:12 out-ffv1.mkv [19:19] <ubitux> -rw-r--r-- 1 ubitux ubitux 489M Oct 17 19:15 out-x264.mkv [19:19] <ubitux> so legacy and x264 at ~21 fps [19:20] <ubitux> and ffv1 "3" at 65 fps [19:20] <ubitux> (x264 is lossless, qp=0) [19:20] <ubitux> so well ok, not as good, but definitely faster now, so that's a good trade off at least [20:15] <cone-186> ffmpeg.git 03Diego Biurrun 07master:f52fd3f3b26f: fate: aac: Add test for AAC-LD [20:15] <cone-186> ffmpeg.git 03Michael Niedermayer 07master:123c6ca80983: Merge commit 'f52fd3f3b26f0d80e4f0b374b7695383feca5b92' [20:30] <cone-186> ffmpeg.git 03Martin Storsjö 07master:a27f1116cdfe: fate: Increase the tolerance in the lavr tests [20:30] <cone-186> ffmpeg.git 03Michael Niedermayer 07master:57c018d54272: Merge remote-tracking branch 'qatar/master' [20:36] <humanvitr> Can anyone give me a direction... I'm implementing a new protocol, based in sockets for video transmission, I see they are listed in ./libavformat, as I want to implement one from scratch, where should I reference my files so ffmpeg can read from there?! [21:06] <michaelni> humanvitr, Makefile, allformats.c [21:14] <humanvitr> thanks michaelni [21:35] <humanvitr> michaelni, I compiled everything, but my ffmpeg says "Protocol not found", you know where else shall I put my protocol? [21:36] <cone-186> ffmpeg.git 03Clément BSsch 07master:0bf85803545d: avcodec/hevcpred: fix make checkheaders. [21:36] <michaelni> humanvitr, you need to rerun configure after you added the protocol [21:37] <humanvitr> I did... [21:38] <humanvitr> maybe REGISTER_PROTOCOL is not enough to ffmpeg gather the protocol [21:38] <durandal_1707> what protocol? [21:39] <humanvitr> I'm developing one, basically a raw socket [21:39] <llogan> name it "rawket" [21:39] <ubitux> kawaii [21:40] <ubitux> humanvitr: can you show the diff without the .c? [21:41] <ubitux> actually, make sure the name matches in the .c [21:44] <humanvitr> I think I found where I missed something [21:45] <humanvitr> you know what this line in the configure means? udp_protocol_select="network" [21:48] <humanvitr> I think is like some sort of heritage, because rtp uses udp, thank you all people [21:53] <durandal_1707> ubitux: you have flv spec? [21:53] <ubitux> they are public [21:54] <ubitux> https://www.adobe.com/devnet/f4v.html [21:55] <ubitux> annex E [21:55] <durandal_1707> you have it open? [21:57] <ubitux> yes [22:13] <ubitux> BBB: in vp8, there is something weird [22:15] <ubitux> afaiu, VP8_MULTIPLY_SUMSUB() is doing something like this: [22:15] <ubitux> a' = ((a+a)*y)>>16 - ((a*x)>>16 + a) [22:15] <ubitux> b' = ((b+b)*y)>>16 + ((b*x)>>16 + b) [22:15] <ubitux> i don't understand why it's not doing: [22:15] <ubitux> a' = ((a+a)*y)>>16 - ((a+a)*x)>>16) [22:15] <ubitux> b' = ((b+b)*y)>>16 + ((b+b)*x)>>16) [22:15] <ubitux> basically swapping the first 2 mul with the 2 following add [22:16] <ubitux> what am i missing? [22:16] <ubitux> (yeah i know i should be working on vp9 instead) [22:16] <ubitux> i'll likely won't have to care about this since i'm going to use pmaddwd from ssse3, but i'm curious [22:18] <ubitux> (x and y being the 2 constants) [22:24] <ubitux> or maybe it should be: [22:24] <ubitux> a' = (a*y)>>16 - ((a*x)>>16 + a) [22:24] <ubitux> b' = (b*y)>>16 + ((b*x)>>16 + b) [22:24] <ubitux> (looking at the c code) [22:24] <ubitux> but in that case i don't get why there is this extra x2 in the left side [22:51] <Compn> anyone seen samples with "aacf" isom ? [22:51] <Compn> some kind of aac obviously [00:00] --- Fri Oct 18 2013

1 0

ffmpeg.log.20131017
by burek 18 Oct '13

18 Oct '13

[01:24] <MorehouseJ09> hmmm ... just curious. I've been playing around with the libavformat / libavcodec libraries a lot and have figured out most issues [01:25] <MorehouseJ09> but in terms of pushing to a production system, do most people work with the libraries or just wrap ffmpeg? I saw a stack overflow stating you could call the compiler with -Dmain=ffmpeg [02:02] <BlkPoohba> I'm trying to rotate a video and getting an error. ffmpeg -i VID_20110108_085632.3gp -vf transpose=1 VID_20110108_085632-2.3gp Error: Encoder (codec none) not found for output stream #0:1 [02:02] <BlkPoohba> http://pastebin.com/iPrQkEAT [02:03] <sacarasc> Define an encoder. [02:05] <BlkPoohba> I have to rebuild it and define one? I installed this using yum. [03:40] <AisIceEyes> Hi! I'm trying to join together two videos on top of each other. I already did these two ffmpeg commands - http://pastebin.com/YgLtFC0C [03:41] <AisIceEyes> I'm currently stuck on making the "clear" version on top of the "blurred" version [03:41] <AisIceEyes> I'm not sure how to do that [03:41] <AisIceEyes> Can anybody help please? Been googling for around 2 days already [03:41] <AisIceEyes> Only achieved it by using OpenShot but yeah, would prefer if there is an ffmpeg command to merge the two videos on top of each other [04:47] <AisIceEyes> Do you guys think the blend video filter - http://ffmpeg.org/ffmpeg-filters.html#blend - can achieve what I have to do to put a video on top of another video (top video with resolution 1080x720, bottom video 1280x720)? [05:00] <relaxed> AisIceEyes: you need -filter_complex, pad filter, and the overlay filter [05:01] <relaxed> pad to create the blank canvas of the second video and overlay to place it there. [05:02] <AisIceEyes> relaxed: Thanks for the response. Will google what you mean and study. Thanks for the tip! [05:04] <relaxed> AisIceEyes: http://trac.ffmpeg.org/wiki/Create%20a%20mosaic%20out%20of%20several%20inpu… [05:04] <AisIceEyes> relaxed: Thanks for the link. Reading it now and will study and test. Thanks again! [05:05] <relaxed> in that example they use nullsrc instead of pad to create the "canvas", which is probably easier. good luck [06:21] <sim590> hi! I'm trying to screencast using ffmpeg. For now I have this `ffmpeg -f x11grab -y -r $FRAMERATE -s $SIZE -i :0.0 -f alsa -i hw:0,0 -acodec mp3 -vcodec ffvhuff -qscale 0 ./out.mkv`. But the audio recording just stops and goes on randomly. How can I improve the audio recording? I have an ATI atom CPU, so the problem may be about performence. [06:36] <relaxed> sim590: ffvhuff is lossless so you can omit -qscale 0 [06:36] <relaxed> use -acodec libmp3lame [06:38] <relaxed> how can it stop AND go on randomly? [06:57] <sim590> relaxed: I meant that the audio recording just has some holes [06:58] <sim590> when I watch the output, audio just stops sometimes and begins again [07:01] <sim590> it "cuts" [09:22] <Apic> Morn (UGT). [11:19] <olspookishmagus> hello, I've just processed a video with ffmpeg, as such: [11:20] <olspookishmagus> $ ffmpeg -i in.mp4 -vf "transpose=1" out.mp4 [11:20] <durandal_1707> and? [11:20] <olspookishmagus> and the resulting file is 5MB smaller than the original, why? [11:20] <durandal_1707> because you are transcoding [11:21] <olspookishmagus> transcoding is the same as: re-encoding? [11:21] <durandal_1707> if you just wanted to change some flag that says that file should be played rotated/flipped/whatever than filter is not for that [11:21] <Mavrik> yep [11:21] <olspookishmagus> enlighten me [11:21] <Mavrik> olspookishmagus, you're re-encoding the file and you didn't tell ffmpeg about what quality you want [11:22] <Mavrik> so it just chooses some default [11:28] <olspookishmagus> from a quick look, -sameq WON'T do what I want [11:29] <olspookishmagus> the same aricle mentions some tutorials, any tips as to where are those? [11:29] <durandal_1707> -sameq is long gone [11:29] <Mavrik> lucky. [11:29] <zap0> -sameq rarely worked rght [11:29] <zap0> right/ [11:31] <olspookishmagus> durandal_1707: this is what I've read [11:31] Action: olspookishmagus quotes: "You can check out the tutorials in this wiki to see if there are any articles describing the best practices in video encoding, that might help you with your specific case" [11:32] <olspookishmagus> I guess I should have a look at: https://trac.ffmpeg.org/wiki/How%20to%20encode%20Xvid%20/%20DivX%20video%20… [11:34] <olspookishmagus> http://pastie.org/private/bwnuajimr0uy7zr5nxxqsq [11:47] <olspookishmagus> any guidelines how to capture ffmpeg output? tee? [11:51] <olspookishmagus> well... durandal_1707, Mavrik, zap0 thanks! [11:51] <zap0> ;) [13:33] <ffm> Hello everyone, can somebody explain how to filter_complex "[0:0] [0:1] [1:0] [1:1] concat=n=2:v=1:a=1 [v] [a]" and "scale=320:200" ? [13:34] <durandal_1707> concat .... [v][a],[v]scale=...[v2],.... [13:40] <ffm> What is [v2] ??? and what command will be? This doesn't work =)))) ffmpeg -i d:\1.mp4 -i d:\1.mp4 -filter_complex "[0:0] [0:1] [1:0] [1:1] concat=n=2:v=1:a=1 [v] [a],scale=320:200" -map "[v]" -map "[a]" d:\3.mp4 Error: Cannot find a matching stream for unlabeled input pad 0 on filter Parsed_scale_1 [13:40] <ffm> Sorry: ffmpeg -i d:\1.mp4 -i d:\1.mp4 -filter_complex "[0:0] [0:1] [1:0] [1:1] concat=n=2:v=1:a=1 [v] [a],[v]scale=320:200" -map "[v]" -map "[a]" d:\3.mp4 Output with label 'v' does not exist in any defined filter graph, or was alread used elsewhere. [13:41] <ffm> This is not working too =))))))) [13:43] <durandal_1707> well i told you what to use, but it looks like you do not need my help [13:44] <durandal_1707> v2 is output of scale [13:44] <ffm> really thanks, i don't know that i should use concat and scale (plz see my very first question) =) [13:44] <durandal_1707> i don't have time [13:44] <ffm> go work then =) [13:54] <x-s4nd3r> I recently downloaded a .flv video, and it's audio is out of sync, how can I fix this? [13:56] <ffm> x-s4nd3r, this is a really good question. Do you see how many people are in this irc? They all have ffmpeg problems =) Thanks for asking =) [13:57] <x-s4nd3r> fuck you, ffm. [13:59] <file> O.o [14:04] <ffm> x-s4nd3r, If you used Windows, I can play video and sync a/v with just ctrl + '+' button with KMPlayer =), may be windows media player also can sync, let me google it for you.... and yes, it can. [14:04] <durandal_1707> x-s4nd3r: perhaps you could try asyncts filter [14:05] <x-s4nd3r> what is that, drundal? [14:05] <durandal_1707> an audio filter [14:05] <durandal_1707> if async is constant you could play with asetpts filter too [14:06] <x-s4nd3r> what the fuck are you saying, man [14:06] <x-s4nd3r> I didn't get a word you just threw outta your mouth. [14:06] <x-s4nd3r> I am a fuckin' newbie, alright? [14:06] <x-s4nd3r> don't go all dennis ritchie on me [14:06] <durandal_1707> you could also try aresample filter [14:06] <x-s4nd3r> how can I use these fuckin' things? [14:07] <x-s4nd3r> whats the command [14:07] <durandal_1707> but what you actually want to archieve? [14:07] <durandal_1707> also note that i'm not paid to listen your nice speak [14:08] <x-s4nd3r> drundal_1707: this is a fuckin' support channel, you gotta suck my cock when I ask for help. [14:08] <x-s4nd3r> now fucking suck it. [14:08] <saste> x-s4nd3r, keep calm or i'll kick you off [14:08] <x-s4nd3r> sorry saste [14:08] <ffm> deal with it is the best way. [14:09] <ffm> your video is corrupted, don't you understand it? [14:09] <durandal_1707> ffm: ffmpeg -i d:\1.mp4 -i d:\1.mp4 -filter_complex "[0:0] [0:1] [1:0] [1:1] concat=n=2:v=1:a=1 [v] [a],[v]scale=320:200[v2]" -map "[v2]" -map "[a]" d:\3.mp4 [14:10] <ffm> durandal_1707: thanks, man [14:11] <ffm> x-s4nd3r, is you want just play this video, you should say to us what video player you are using, may be it can just sync audio/video with hotkeys, for example kmplayer for windows - can [14:12] <x-s4nd3r> vlc. [14:12] <x-s4nd3r> how can I do this with vlc, ffm? [14:12] <ffm> x-s4nd3r, try K, J keys http://lifehacker.com/5910943/fix-out+of+sync-audio-in-vlc-with-a-keyboard-… [14:14] <x-s4nd3r> I don't know which audio delay I should use, ffm. [14:14] <x-s4nd3r> can you check out the videofile? [14:14] <ffm> x-s4nd3r,yes, i have vlc too, give video url [14:15] <x-s4nd3r> you can just view the video online [14:15] <x-s4nd3r> just check out the 3 minutes [14:15] <x-s4nd3r> http://sharesix.com/8b7mo27x3xc9 [14:15] <x-s4nd3r> ffm: http://sharesix.com/8b7mo27x3xc9 [14:15] <x-s4nd3r> ffm: just see from 02:00 [14:15] <EchoDev> Guys I'm trying to compile ffmpeg on windows but I seem to fail to make msys use mingw64 [14:16] <EchoDev> I copied the mingw64 build into the home folder and tried the export path command but I just can't make it pick up [14:17] <relaxed> EchoDev: there's http://ffmpeg.zeranoe.com/builds/ if you haven't seen it. [14:18] <EchoDev> relaxed yeah I have seen it but I want to compile it for LAV and there is a patched ffmpeg for that it seems [14:18] <ffm> x-s4nd3r, really sorry, but this is too long video, i will be download this for tomorrow, but try K or J key for few times, you should see that video and audio is in sync from some point (some video gets 200ms), you feel it [14:18] <x-s4nd3r> ffm, you don't need to download it. [14:18] <JEEB> EchoDev, the usual way I switch mingw toolchains on msys is by modifying the etc/fstab file in msys's root [14:18] <x-s4nd3r> ffm, click "free" [14:18] <x-s4nd3r> and then click the play button [14:18] <x-s4nd3r> just see 02:00 to 03:00 [14:18] <JEEB> and then relaunching the msys shell [14:19] <x-s4nd3r> thats it [14:19] <EchoDev> ok thanks JEEB let me try that out [14:19] <JEEB> because msys auto-adds whatever is "mounted" as /mingw's bin into PATH [14:20] <ffm> x-s4nd3r, audio delay is about 3 sec [14:20] <x-s4nd3r> how many times I should press the K or J button, ffm? [14:20] <x-s4nd3r> audio delay 3? [14:22] <ffm> x-s4nd3r, every press is + or - 50ms, so 2 press = 100ms, 20press (hold it, just hold) - 1s [14:23] <x-s4nd3r> should I stop when "audio delay 50ms" comes up? [14:23] <ffm> x-s4nd3r, no, it just says that now delay is 50ms, you should keep press or hold it [14:23] <x-s4nd3r> press what [14:24] <x-s4nd3r> I am very confused [14:24] <ffm> x-s4nd3r, press K or J key (K K K K K K K K K K K or J J J J J J J J J J J J J J) [14:24] <x-s4nd3r> how many times should I press K? [14:27] <ffm> x-s4nd3r, to say how many times you should press K I need to download video, what I said earlier is not possible, just press till you feel that a/v is in sync (if K doesn't not help you, press J) [14:27] <x-s4nd3r> K is +? [14:27] <x-s4nd3r> and J is -? [14:30] <ffm> x-s4nd3r, K and J for increase/decrease delay [14:33] <durandal_1707> for vlc help go to #videolan [14:34] <durandal_1707> but if you want to fix async with ffmpeg than this is right place [14:46] <ffm> durandal_1707, can you tell me, can ffmpeg concatenate image and video, or i should convert image to video and only then concatenate all? [14:47] <durandal_1707> ffm: you mean images as files? [14:48] <durandal_1707> you can with concat filter [14:48] <durandal_1707> not sure about concat demuxer [14:48] <ffm> durandal_1707, thank you [14:50] <ffm> durandal_1707, sorry for my last time [15:17] <smjd> -vsync and -af aresample=async=1 don't do any syncing while streaming [16:51] <buu> Is there any particularly clever way to split a 'm4b' file into multiple tracks based on chapter metadata? [16:56] <Mista_D> Anyway to see mediainfo's "Delay_Original_Settings" via FFprobe? it has "drop_frame_flag=0 / closed_gop=1 / broken_link=0" [17:44] <lakmuz> hi, i need to combine 2 video viles in one, it should play same time, first video should be at the left of video, second video at the right, any suggestions? [17:46] <durandal_1707> use overlay filter [17:48] <lakmuz> thanks [19:27] <starfox21> hey guys I have an mp4 with a major brand isom and I want to convert it to mp42, how do I go about that? [20:37] <Slackyman> hi [20:40] <humanvitr> Can anyone give me a direction... I'm implementing a new protocol, based in sockets for video transmission, I see they are listed in ./libavformat, as I want to implement one from scratch, where should I reference my files so ffmpeg can read from there?! [20:43] <ac_slater> hey guys, does anyone know a way (not through libav*) to discover the h.264 profile of a file? [20:43] <ac_slater> ie - baseline, main, etc [20:45] <ac_slater> My container is an mp4 bte [20:45] <ac_slater> btw* [20:45] <llogan> why not via libav*? [20:49] <ac_slater> llogan: Cause I'm not trying to do it programatically right now. Just looking for a utility. [20:49] <ac_slater> llogan: ah mp4info did the trick [20:51] <kriskropd> I'm having no luck concatenating files like this http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20concatenate%20(join%2C%20merg… [20:52] <kriskropd> I either get "mylist.txt: Invalid data found when processing input" when I try using a txt file [20:52] <kriskropd> or I get "Unrecognized option 'c'" [20:53] <kriskropd> I've tried this with ffmpeg version 0.8.6-6:0.8.6-1ubuntu2, ffmpeg version N-55937-g93cf7b0 and avconv version 0.8.6-6:0.8.6-1ubuntu2 [20:54] <kriskropd> im not going to paste bin the command because it is identical to the wiki example - the only thing is I'm merging flv files [20:54] <llogan> for avconv and libav's so-called ffmpeg go to #libav. obviously we only support ffmpeg from FFmpeg here [20:54] <llogan> the console output is still needed [21:04] <kriskropd> okay, I got it to finally parse, however it only merges the first video to output.flv and then spits out "[flv @ 0x33def00] Unsupported video codec (c)" and stops [21:06] <kriskropd> and here is a pastebin for output http://sprunge.us/dEQJ [21:06] <kriskropd> its identical to the example from the wikipage [21:09] <kriskropd> if I send the flv files to mp4, I don't get the codec error,b ut it still stops at merging only the first video file to the output [21:09] <kriskropd> mp4 versions of the videos did give me this error instead [mov,mp4,m4a,3gp,3g2,mj2 @ 0x2bdef00] Found duplicated MOOV Atom. Skipped it [21:17] <llogan> does ffmpeg have issues with one of the inputs? ffmpeg -i 00.flv -i 01.fl -i 02.flv ... [21:17] <kriskropd> llogan: do you mean, does ffmpeg complain when i "ffmpeg -i 01.flv 01.mp4", for instance? [21:18] <kriskropd> if so, no, it converts to mp4 just fine with no complaints [21:18] <kriskropd> but i still can't merge the mp4 fiels it outputs in that process using concat [21:19] <Slackyman> anyone here is an owner of FFMPEG copyright? [21:19] <kriskropd> Slackyman: ffmpeg is open source [21:19] <Slackyman> I'd like to know what's the matter with ANVSOFT [21:19] <Slackyman> yes, I know [21:20] <Slackyman> and Any DVD Converter is violating it since years [21:20] <kriskropd> llogan: also if I do 'ffmpeg -i 00.flv -i 01.flv output.mp4' (which might eb wrong, idk waht I'm doing, lol) it still only outputs the first video to the output file [21:20] <Slackyman> so do ANY VIDEO CONVERTER [21:20] <Slackyman> and any anvsoft software [21:20] <llogan> Slackyman: same old shit, different names, from the same Chinese guys. [21:21] <Slackyman> si Chinese are really the real stealers nowaday [21:21] <llogan> kriskropd: if you provide your inputs i can try to duplicate [21:21] <Slackyman> and we can't do nothing? [21:23] <llogan> you can add a "license violation" type ticket on the bug tracker [21:23] <llogan> like: https://trac.ffmpeg.org/query?status=closed&status=new&status=open&status=r… [21:23] <kriskropd> llogan: I pasted the output with flv here http://sprunge.us/dEQJ and my attempt with mp4 is found here http://sprunge.us/SdWQ [21:25] <Slackyman> the thing boring and getting me mad is that is difficult to say to someone that that software is someway "illegal" [21:25] <Slackyman> and I know of some italian universities adopting that software for transcoding [21:27] <llogan> Most violators don't care. Others just don't know or understand since legal stuff is complicated. [21:28] <durandal_1707> Slackyman: what software? [21:28] <Slackyman> durandal_1707: any dvd converter violates ffmpeg, libav and much more [21:28] <llogan> kriskropd: can you provide the input files? [21:28] <Slackyman> more! [21:28] <Slackyman> not now [21:28] <Slackyman> they're distribuiting a non-free ffmpeg.exe executable [21:29] <Slackyman> with enable-nonfree option [21:30] <durandal_1707> report them on trac and sue them! [21:30] <GoaLitiuM> where do you even need nonfree version of ffmpeg? [21:31] <Slackyman> I'm not an owner [21:31] <kriskropd> llogan: afraid I can't provide them, sorry :/ [21:31] <GoaLitiuM> or where people need it in general [21:31] <kriskropd> llogan: matter of privacy [21:31] <JEEB> <GoaLitiuM> where do you even need nonfree version of ffmpeg? <- right now the biggest reason to compile with --enable-nonfree is fdk-aac [21:31] <JEEB> which is the best aac encoder right now [21:31] <kriskropd> llogan: I didn't generate the original files either so I can't rereate the format [21:31] <kriskropd> recreate* [21:31] <GoaLitiuM> JEEB: that's what i thought, and isn't the ffmpeg's own AAC encoder good enough on most cases? [21:32] <JEEB> well, yes and no [21:32] <Slackyman> Any Video Converter is a freeware toolk using ffmpeg and libav to convert anything to anything [21:32] <llogan> GoaLitiuM: it's experimental, although it's being worked on now [21:32] <Slackyman> there's also a AAC+ [21:32] <Slackyman> it's non free [21:32] <Slackyman> I think [21:32] <JEEB> yes, but fdk does the job of both HE and LC AAC [21:32] <GoaLitiuM> yeah, HE-AAC [21:32] <JEEB> so you really don't need any other AAC encoder any more [21:33] <Slackyman> hey, guys [21:33] <Slackyman> my idea is to make a better ffmpeg gui [21:33] <JEEB> the AAC encoder is "OK" for quick jobs, but if you're trying to do the best job possible you just compile your own binary with fdk [21:33] <Slackyman> better than the one of anvsoft [21:33] <Slackyman> and release it gpl 3 [21:35] <Slackyman> Is there a real difference in using native rtmp support or librtmp? [21:35] <Slackyman> anyone see any difference? [21:35] <JEEB> the native rtmp support is good [21:35] <JEEB> you generally shouldn't need librtmp [21:36] <Slackyman> and so why to use it? [21:36] <Slackyman> :) [21:36] <Slackyman> does native rtmp accept ssl? [21:37] <JEEB> yes [21:38] <kriskropd> llogan: this might help http://sprunge.us/NNDY mediainfo on the flv file indicates it is sorensor spark video codec [21:39] <durandal_1707> kriskropd: it doesn't work with ffmpeg? [21:40] <Slackyman> sorenson is h.263 [21:41] <Slackyman> it should work with ffmpeg [21:42] <Slackyman> bye [21:42] <Slackyman> see ya [21:42] <llogan> bye [21:43] <kriskropd> durandal_1707: llogan: I pasted the output with flv here http://sprunge.us/dEQJ and my attempt with mp4 is found here http://sprunge.us/SdWQ [21:43] <kriskropd> durandal_1707: oh sorry, i didn't notice you just got here [21:44] <kriskropd> durandal_1707: I'm trying to concat some private flv files, but it seems the video codec is causign trouble [21:44] <kriskropd> causing* [21:46] <durandal_1707> kriskropd: error message says it all [21:46] <durandal_1707> it have unsuppoted codec 0xC [21:47] <durandal_1707> only 0x0 - 0x9 are supported in flv demuxer [21:47] <durandal_1707> without sample it is impossible to add support [21:49] <kriskropd> durandal_1707: what about when i copied the video to an mp4 container? This is the mediainfo for a file made with 'ffmpeg -i 01.flv 01.mp4' http://sprunge.us/WVYI [21:50] <kriskropd> durandal_1707: woudl the codec issue persist even after converting the file? [21:50] <kriskropd> would* [21:50] <durandal_1707> kriskropd: copy can not work as 0xC is unknown codec [21:50] <kriskropd> i see, so there really isn [21:51] <kriskropd> isn't much I can do withoutt digging up mroe info on this codec [21:51] <durandal_1707> perhaps there is flv spec that says what is 0xC... [21:58] <durandal_1707> hmm i just looked at spec and 0xC is nowhere there [22:01] <ubitux> isn't it some speex with particular profile or something? [22:01] <ubitux> maybe check in the swf specs [22:06] <ubitux> where is the flv? [22:06] <durandal_1707> codec is is UB[4] [22:06] <ubitux> is it audio or video codec? [22:07] <durandal_1707> video [22:07] <durandal_1707> and log says: Unsupported video codec (c) [22:07] <llogan> kriskropd: ...not even one sample? [22:08] <durandal_1707> i guess its just false positive [22:08] <durandal_1707> as codec is reported midstream [22:08] <durandal_1707> s/codec/stream [22:09] <durandal_1707> http://sprunge.us/dEQJ [22:10] <btree> Coelia Cezanne? [22:10] <btree> :) [22:10] <btree> nvm [22:10] <kriskropd> llogan: i'd be voiding copyrights if i shared it :x the video files are from an old CISSPA training dvd from Logical Security [22:11] <durandal_1707> perhaps it dvd got corrupted.... [22:11] <kriskropd> maybe :( but all the files playback just fine [00:00] --- Fri Oct 18 2013

1 0

ffmpeg-devel.log.20131016
by burek 17 Oct '13

17 Oct '13

[00:00] <Paranoialmaniac> michaelni: i'm not ok because hvcC format may be changed on a draft as the divx and my l-smash use configurationVersion==0. the point is parameter sets are placed at the end of hvcC. the change may change their offsets. it is hard to support all draft version hvcC [00:02] <Paranoialmaniac> there is no clue to distinguish them [00:03] <michaelni> Paranoialmaniac, there are no offsets in your patches [00:03] <Paranoialmaniac> ? [00:03] <Paranoialmaniac> ahhh [00:04] <Paranoialmaniac> sorry, i misunderstand :P [00:04] <Paranoialmaniac> and you pushed with my nalff support? [00:05] <Paranoialmaniac> with the hevc decoder [00:05] <michaelni> i pushed the hevc decoder patch, it wasnt split up based on authors [00:05] <michaelni> so i dont know what parts from what authors are in it [00:06] <Paranoialmaniac> hevc_decode_extradata is written by me [00:06] <j-b> funny how the hevc commit did not go through the ml [00:07] <Paranoialmaniac> sad [00:09] <michaelni> j-b, it got stuck on vlc side or ffmpeg side ? or maybe it will turn up later ... [00:09] <j-b> michaelni: that's a great question [00:09] <j-b> I guess it is a huge patch [00:09] <michaelni> yes [00:10] <j-b> so either git hook said "no way" or the ml said "fuck you" :) [00:10] <j-b> I don't know, and well, I do not care much :) [00:10] <Paranoialmaniac> michaelni: apply as you want... [00:11] <j-b> s/want/can [00:11] <michaelni> Paranoialmaniac, domo arigatou gozaimas [00:13] <wm4> I found a mail by mini containing a hevc patch in my spam filter [00:13] <wm4> google decided it was spam [00:13] <Paranoialmaniac> michaelni: https://github.com/VFR-maniac/libav/commits/hevc_nalff <-- feel free to pick patches if you need. some patches are already applied though [00:15] <wm4> (I wonder whether google's spam filter decision are the same for every account?) [00:15] <michaelni> Paranoialmaniac, thanks [00:23] <Compn> its in ?! [00:23] <Compn> ehe [00:23] <cone-770> ffmpeg.git 03gcocherel 07master:cf49d0156dfd: valgrind cleanup(cherry picked from commit 0d5efa40b94c0de92a1fe5091b21e0e2d36bae3d) [00:23] <cone-770> ffmpeg.git 03Yusuke Nakamura 07master:53f903b7c542: lavf/mov: Support HEVC demuxing. [00:23] <cone-770> ffmpeg.git 03Yusuke Nakamura 07master:16b6839de676: lavf/matroskadec: Support HEVC demuxing. [00:23] <Compn> did you check mailman ? [00:24] <michaelni> didnt check anything except my cvslog mailbox [00:41] Action: llogan increases cvslog message size limit [00:41] <j-b> now, I got it [00:41] <j-b> 550K [00:41] <j-b> no wonder it did not get through [01:28] Action: kierank wonders if he has the guts to add avx2 to swscale [01:29] <iive> you can do it! [01:42] <cone-770> ffmpeg.git 03Michael Niedermayer 07master:f58cb772d966: avformat/mov: fix "correctly" typo [01:42] <cone-770> ffmpeg.git 03Guillaume Martres 07master:fb3cea4be339: FATE: add HEVC tests [01:42] <cone-770> ffmpeg.git 03Michael Niedermayer 07master:48e1114abf77: fate: fix DBLK_A_MAIN10_VIXS_2 on big endian [01:44] <michaelni> smarter, hevc fate tests pass on big endian mips now as well as little endian (same 2 tests failing) [01:44] <smarter> I think right now there's one test failing on ppc [01:44] <michaelni> little endian x86_64 that is [01:45] <michaelni> smarter, i just fixed a big endian bug [01:45] <michaelni> maybe that was it ? [01:45] <smarter> oh [01:46] <smarter> ah I think elenril found that fix too [01:48] <smarter> mraulet fixed one test a few hours ago: https://github.com/OpenHEVC/libav/commit/19c5d9ed28606e7a166d180fc9f1517cb1… [01:51] <kierank> I need to find something small to port to avx2 [01:51] <kierank> in swscale [01:51] <kierank> then work on bigger stuff [01:52] <j-b> why need? [01:52] <michaelni> some of the simple unscaled converters might be an option [01:53] <kierank> j-b: porting some of the big yasm functions to avx2 seems hard [01:53] <kierank> lots of macros [01:54] <michaelni> stuff in: rgb2rgb_template.c should be easy to play, not sure how usefull optimizing them to avx2 is though [01:55] <michaelni> s/to play/to play with" [01:55] <kierank> i think output.asm might be easiest [01:56] <kierank> yuv2plane1 [01:57] <cone-770> ffmpeg.git 03Mickaël Raulet 07master:64b3aaf8f6c2: hevc: fixing TSCL_A_VIDYO_5 decoding output order(cherry picked from commit 19c5d9ed28606e7a166d180fc9f1517cb174146f) [09:32] <dol> hi all. Yesterday I asked a question about a sws_scale artefact here, which I have already posted in detail with pictures in http://forum.doom9.org/showthread.php?t=169036 [09:33] <dol> there is something wrong with my settings I guess because if I decode it with the cli, I don't see the issue anymore [09:33] <dol> so, could anyone help me to go through the code? it is just couple of lines [09:38] <saste> dol pastebin [09:40] <dol> @saste, ok I will. [09:45] <dol> @saste, here is the code [09:45] <dol> http://pastebin.com/dTGZbdJj [09:45] <dol> it's now more than couple of lines but it's pretty common code I believe [09:46] <dol> so, the problem occures in line 252 [09:47] <dol> you see that I call sws_getCachedContext with the current settings [09:48] <dol> if I set there, decodedpic->width + 1, everything works perfect without any issue [09:48] <dol> so I believe that the sws_scale works good if it does scaling [09:48] <dol> if I don't set it to +1, it means that srcW = dstW and it doesn't do any scaling [09:48] <dol> but I couldn't verify this with cli. [09:49] <dol> so I think there is a small trick that I forget [09:53] <dol> @saste ? [09:54] <saste> dol, no if dst pix_fmt != src_pix_fmt it will convert [09:55] <dol> yes, this is the problem. in my application I don't do any scaling. Therefore always the pix_fmt = src_pix_fmt [09:55] <dol> but I see the effect that you can see from the pictures in http://forum.doom9.org/showthread.php?t=169036 [09:55] <dol> however, if I set srcW+1, meaning that pix_fmt != src_pix_fmt, I don't see the problem anymore [09:56] <dol> but I believe that it scales the image which is not necessary. I am trying to find out why it works perfect with dstW = srcW and not with srcW = srcW+1 [09:57] <saste> i suspect the avpicture_fill() may be wrong [09:57] <saste> since it always assumes alignment of 1, which may not be the case [09:59] <dol> what can I try instead? [10:00] <saste> what is mvSize[0] and mvSize[1]? [10:01] <saste> look like linesizes, but you're using them as width/height [10:01] <dol> mvSizes are coming from the encoder, actualyy they are width and height of the source [10:02] <saste> are you sure the data is aligned to 1 [10:02] <saste> usually image data lines contain padding at the end [10:03] <saste> something like RGBRGB...RGBPPPP where PPPP are padding bytes [10:03] <dol> well, I don't know anything about alignment [10:03] <dol> can you tell me how to check? [10:03] <saste> usually you need both w/h and linesize to be able to render the image correctly [10:03] <dol> I can easily try out [10:03] <saste> i explained you, it's not complicate at all [10:05] <dol> I will debug the decoded image and see that it contains [10:07] <dol> @saste, decodedpic->data[0] has different values at the end of the array [10:07] <dol> they are not the same like PPPP [10:08] <saste> what is av_pic? [10:08] <saste> is it a global? [10:08] <dol> no it is AVFrame declared in the class [10:08] <dol> it contains the decoded frame [10:14] <dol> @saste, do you see any problem in the flow ? [10:14] <dol> or any weird thing? [10:20] <saste> no seems correct [10:20] <dol> this issue happens only in some weird resolutions [10:21] <dol> if it is for example, 800x500 resolution, no problem [10:21] <dol> but if it is 1382 x 864 or whatever vertical resolution, I see that a black bar comes on top of the image at the right hand side [10:24] <saste> dol again i suspect you're confusing linesize with width [10:24] <saste> this would explain the issue [10:25] <dol> you mean the destination linesize? [10:25] <saste> no in the source [10:25] <dol> in the avpicture_fill? [10:25] <dol> aha [10:26] <dol> but what should I use instead? [10:26] <dol> should I create the linesizes myself? [10:27] <dol> but then the constructed image will be distorted because linesize is coming from the decoding and if I choose smaller linesize, the next line will be shorter? [10:27] <dol> couldn't explain it well [10:29] <saste> what if the problem is really the source?, that is the encoding process? [10:30] <saste> anyway this is not really related to ffmpeg development, move to #ffmpeg [10:34] <JEEB> while you are technically correct, you know how many people related to ffmpeg are in there :P [10:35] <JEEB> for example, this guy had no response for about 6 or 8 hours or so [10:35] <JEEB> and thus I just decided to tell him to try here yesterday [10:35] <JEEB> #ffmpeg is generally OK for command line help and such, that's true though [10:35] <JEEB> just my five cents [10:48] <dol> @saste, encoding part is eliminated because I checked the encoded stream myself, which contains no black border at all [10:48] <dol> also, I can't reproduce it if I use ffmpeg cli with the follwoing command ffmpeg.exe -i 1.264 1.yuv -v debug [10:49] <dol> and yuv file doesn't show up any black border [10:49] <dol> so, I believe that I make a mistake during avpicture_fill or whatever call but I am unable to find it [10:53] <dol> @saste, when I have jsut realized that ffmpeg doesn't use swscale since the output is yuv file in the following command [10:53] <dol> ffmpeg.exe -i 1.264 1.yuv [10:53] <dol> can you guys tell me how I can export them to bmp so that it will use swscale? [10:55] <dol> ok, I tried ffmpeg.exe -i 1.264 -f image2 image-%3d.bmp [10:55] <dol> and the output bmp images doesn't show any border as well [10:56] <dol> I mean black border [11:00] <dol> still waiting for a push? [11:04] <cone-159> ffmpeg.git 03Derek Buitenhuis 07master:884fd4d2590c: tableprint: Fix use of a size_t print with MSVC [11:04] <cone-159> ffmpeg.git 03Derek Buitenhuis 07master:bc31a7a3b6bc: tablegen: Don't use cbrtf in host tools [11:04] <cone-159> ffmpeg.git 03Derek Buitenhuis 07master:479a52795526: cos_tablegen: Don't use lrint [11:04] <cone-159> ffmpeg.git 03Derek Buitenhuis 07master:c0085f94fea8: mpegaudio_tablegen: Don't use llrint [11:04] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:12e9821042b7: Merge commit 'c0085f94fea89b180e5727b193484a83586d3490' [11:18] <cone-159> ffmpeg.git 03Diego Biurrun 07master:3b4fa54866f5: cavs: more K&R formatting cosmetics [11:18] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:3a8b52c0113b: Merge commit '3b4fa54866f58bf3d8a8dcc460c73ef6564c0ad8' [11:22] <michaelni> dol, does that ffmpeg that works use the same flags and linesizes and w/h as you do ? [11:24] <dol> michaelni, I can't see the linesize information if I run the ffmpeg with -v debug option [11:24] <dol> what I see is the following [11:24] <michaelni> you can add a printf [11:24] <michaelni> or av_log [11:25] <dol> http://pastebin.com/Pm4Sq3Ee [11:26] <dol> yes but I need to compile it and I don't have my linux machine with me [11:27] <dol> Setting 'flags' to value '0x4', I think this is SWS_BICUBLIN [11:27] <dol> what is? w:iw h:ih flags:'0x4' interl:0 [11:28] <dol> width = iw? [11:28] <michaelni> bicublin is 0x40 [11:28] <dol> ah sorry SWS_BICUBIC [11:29] <michaelni> its using bgr24 you use rgb32 or something else IIRC [11:29] <michaelni> try -pix_fmt rgb32 [11:30] <dol> tried: http://pastebin.com/VPyWUyv4 [11:31] <dol> output show no artefact again [11:31] <dol> thanks a lot for your help btw. [11:39] <cone-159> ffmpeg.git 03Diego Biurrun 07master:29c455ce3daf: bitstream: Check the result of av_malloc() [11:39] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:8895b71d6fc1: Merge commit '29c455ce3daf7fb369ba20cf77c74bd8e3b43b55' [11:46] <michaelni> dol, if you think thats a bug in sws (which is possible), you could try to write a minimal testcase that is like a c file that creates a small yuv420 pic with some pattern and runs sws over it and dumps the result or so, and then open a bugreport on trak [11:47] <michaelni> the code snippet you posted doesnt look like it would built on its own [11:47] <dol> michaelni, it's written in Qt but I will try to write a minimal code in pure C [11:47] <dol> my workaround solution is just to set the width to width+1 [11:48] <michaelni> thanks [11:48] <michaelni> should make testig/debuging easier [11:48] <dol> michaelni, do you think that it can be a bug in swscale even if it is not reproducable with cli? [11:50] <michaelni> dunno, they might use different linesizes or something [12:12] <cone-159> ffmpeg.git 03Anton Khirnov 07master:dd33637c1862: tiny_psnr: switch f32 handling to floating point [12:12] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:6659eb711674: Merge commit 'dd33637c18629c3e554ebb146bbeb45c9745a5cf' [12:17] <cone-159> ffmpeg.git 03Anton Khirnov 07master:68edd5be0980: FATE: use proper comparison mode in the lavr tests [12:18] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:8778c49aceb9: Merge commit '68edd5be0980941924ae633d98589d56a8091bbd' [12:38] <cone-159> ffmpeg.git 03James Almer 07master:6b081eff4dfc: fate: add vorbiscomment cover art test [12:38] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:83bfcc3ce255: Merge commit '6b081eff4dfc3c899960f69f30cb567266b7dca3' [12:44] <cone-159> ffmpeg.git 03Yusuke Nakamura 07master:b81dbd6cb752: h264_parser: Fix POC parsing for the case where MMCO_RESET is absent. [12:44] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:91248f081f34: Merge commit 'b81dbd6cb7522bea96d78a52f8a4c25a47b820c9' [13:00] <cone-159> ffmpeg.git 03Yusuke Nakamura 07master:4baba6c813b7: h264_parser: Fix POC parsing for the case where MMCO_RESET is present. [13:00] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:f7f74a37b82d: Merge commit '4baba6c813b7a1f27370e20fb1a87b05fcb39208' [13:00] <Daemon404> michaelni, just fyi for trac #3048, you should refer to TheRyuu [13:01] <Daemon404> since he wrote the icl support [13:01] <Daemon404> he's working on inline asm too (supposedly) in ffmpeg [13:17] <cone-159> ffmpeg.git 03Ronald S. Bultje 07master:93f305473f88: lavc: Convert some remaining strides to ptrdiff_t [13:17] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:10c6d1b28cbd: Merge commit '93f305473f880729d18b5e42067f19d2106cb2e5' [14:01] <cone-159> ffmpeg.git 03Luca Barbato 07master:a84616b736fc: mpegvideo: K&R formatting cosmetics [14:01] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:8610751d07bc: Merge commit 'a84616b736fca5ebd6b87489dd41bc06ccdf7860' [14:17] <dol> michaelni, I have spotted the problem [14:17] <dol> well let say I have an idea [14:17] <dol> but still I need some help to figure out why [14:18] <ubitux> BBB: did you see any functional changes in the vp9 port? [14:18] <dol> so what I do is that, I increase the resolution of the frame at the encoder side and see what resolution the decoder provides me [14:18] <ubitux> BBB: or that's just random cosmetics, rename etc? [14:18] <dol> the strange thing is, I see no black border thing if the srcW = c1->coded_width [14:19] <dol> I see the effect when the srcW != c1->coded_width [14:21] <dol> or another saying is that, I see the effect when c1->width != c1->coded_width [14:21] <dol> I see no effect when c1->width == c1->coded_width [14:24] <dol> could anyone tell me in what cases coded_width is different from width? [14:39] <cone-159> ffmpeg.git 03Luca Barbato 07master:a90905db2e6a: ffv1: Assume bitdepth 0 means 8bit [14:39] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:ab16d5b7b56e: Merge commit 'a90905db2e6ab1840890f3a88bfd3bf008b9d886' [15:06] <cone-159> ffmpeg.git 03Luca Barbato 07master:95587859cc69: mpegvideo: Move obmc in a separate function [15:06] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:8210e3846129: Merge commit '95587859cc69e5abe37c9e3af48008032d98e262' into HEAD [15:14] <cone-159> ffmpeg.git 03Luca Barbato 07master:825c7c62bb66: mpegvideo: Move 8x8 in a separate function [15:14] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:4d41a25d3940: Merge remote-tracking branch 'qatar/master' [15:17] <dol> 2nd try: could anyone tell me how the coded_width is calculated from width information? [15:18] <dol> or where is the related code? [15:19] <michaelni> TheRyuu, Daemon404 says you wrote ICL support [15:20] <michaelni> first thanks for doing that :) [15:20] <michaelni> and maybe you want to comment on https://trac.ffmpeg.org/ticket/3048 ? [15:20] <Daemon404> he said the patch wasnt complete when i asked in here yesterday [15:20] <Daemon404> but didnt elaborate [15:21] <dol> michaelni, could you please comment on what I have said? [15:21] <dol> I see the effect when coded_width != width [15:22] <dol> and it will be a solution if I somehow how much pixels it has added at the end and substract that much from the srcW in the sws context [15:23] <dol> so apparently, in such resolutions, the width already contains the padding [15:23] <michaelni> dol, iam not sure what and where coded_width and width are related to the swscale issue [15:23] <michaelni> do you have a issue with swscale or a decoder ? [15:24] <dol> well, I am trying to figure out if it is an issue or not. but I have a problem with both decoder and swscale [15:24] <michaelni> Daemon404, hmm, ok, thx, i guess ill copy and paste a few lines from here to trak then so the others who look at trak are aware [15:25] <michaelni> dol, so maybe first find seperate the 2 and find out which of them is the problem [15:25] <iive> without looking at the code, I think that coded_width should be the width rounded up by macroblock size. [15:28] <dol> @iive, michaelni, look at this example. input resolution is 1124x702. When I debug the decoder, I see that the coded_width=1136 , therefore I see a black line at the right side. [15:29] <dol> however, if I use sws_getContext(c1->width - 1, c1->height, .... [15:29] <dol> I see the frame that I supposed to see without any black line [15:30] <dol> so somehow I need to create a logic between coded_width and width so that it will give me 1 [15:30] <dol> I know it is not an optimal solution but it is a solution [15:31] <dol> and the question is, why does it round up even if 1124 is perfectly dividable to 16? [15:32] <dol> so why does the decoder add 12 more? where does it come from :) [15:32] <michaelni> Daemon404, the patch in the ticket is incomplete and ugly yes (i assume that patch was what was meant) [15:32] <Daemon404> TheRyuu, has a WIP [15:32] <Daemon404> i think [15:32] <Daemon404> on HEAD [15:32] <Daemon404> and some ffmpeg-specific changes [15:33] <michaelni> ok, that would be great :) [15:33] <Daemon404> but he's lazy, so feel free to stab him [15:35] <iive> dol, video codecs work on (macro)blocks. this is why they always round up the resolution to a MB size. Some encoders put black bars, most recent just repeat the edge pixels. Usually to simplify calculation stride/line_size is used. It is the offset in bytes between two vertical pixels. [15:36] <iive> in other words, you add it to get the next line. it might be different than coded_width if you have higher sampling formats (9,10,12bit) [15:37] <iive> now, about swscale... it could be a bug in one of the code paths, but it could be a bug in calculation of image parameters. [15:37] <saste> dol: why do you read coded_width rather than width? [15:38] <iive> for example, if input and output widths are the same, then a faster color-format conversion only path is used. [15:38] <dol> @iive, I know some basics about encoding decoding and I know what the padding or linesize or stride does. but this problem is a bit different [15:38] <dol> @saste, I have realized that my issue happens when the coded_width is NOT equal to width [15:39] <dol> that's why I read it [15:39] <saste> dol: ok [15:39] <iive> just make sure that line_size is not smaller than any width [15:39] <saste> dol: what decoder? [15:39] <dol> @iive have you read my post in doom9? http://forum.doom9.org/showthread.php?t=169036 [15:39] <dol> @saste h264 [15:40] <saste> dol starts to make sense [15:40] <saste> that's new code and probably not very much tested [15:40] <saste> i mean when the coded_w/h changes [15:40] <saste> i bet it's a decoder issue [15:41] <dol> @saste, so you say that it is probably a bug that I need to report [15:41] <saste> but, it should be reproducible with ff* tools as well [15:41] <dol> yes that's the most strange part [15:41] <dol> i tried ffmpeg cli and didn't see the effect over there [15:41] <saste> do you have a sample? [15:42] <dol> what sample do you want? I can immediately produce [15:42] <saste> a sample triggering the issue, h264 content with changing h/w [15:42] <saste> (well it should not be hard to create using ts) [15:42] <dol> ok I will create encoded bitstream [15:43] <dol> yes indeed [15:43] <dol> but I don't know how to use it [15:43] <dol> so I will use my own tools :) [15:45] <saste> dol, ffmpeg -f lavfi testsrc=s=1124x702 -c:v libx264 -t 5 out1.ts [15:46] <saste> dol, ffmpeg -f lavfi testsrc=s=1124x600 -c:v libx264 -t 5 out2.ts [15:46] <saste> cat out1.ts out2.ts > out12.ts [15:46] <dol> @saste, this won't help since we won't be able to reproduce it with cli [15:47] <dol> I tried it with the same resolution and didn't pop up [15:47] <dol> so there is really a problem either with my code or the code path in the library [15:48] <saste> dol, try to write a selfcontained c program showing the problem [16:02] <saste> michaelni, willing to comment on the swscale filter patch? [16:42] <durandal_1707> michaelni: have you looked at gbra16 swscale patch?, now you can test it with fate samples [16:44] <michaelni> durandal_1707, not yet [16:56] <michaelni> saste, which patch ? [16:57] <michaelni> i see some sws channel layout patches but filter [16:57] <saste> michaelni: [PATCH] swscale: Support setting filters through AVOptions [16:58] <cone-159> ffmpeg.git 03Paul B Mahol 07master:4413dcc03524: avcodec/exr: piz decompression [17:02] <michaelni> saste, IIRC srcFilter is applied "before" scaling" and dst after [17:03] <michaelni> and it should be possible to apply them without scaling [17:04] <saste> michaelni, but in practice? [17:04] <durandal_1707> michaelni: why hevc/h265 is not in changelog? [17:05] <michaelni> saste, it should work unless theres a bug [17:05] <saste> also, what filter is used if noone is specified? [17:05] <michaelni> "1" [17:09] <michaelni> also the patch shuld be ok from a quick look, that is if wm4 is ok with it too [17:24] <cone-159> ffmpeg.git 03Ronald S. Bultje 07master:734ccf378588: avcodec/vp9: fix band_counts array size / padding [18:00] <cone-159> ffmpeg.git 03Carl Eugen Hoyos 07master:d0b7d24b80b3: Support HEVC in transport streams. [18:33] <cone-159> ffmpeg.git 03Timothy Gu 07master:b4da1fa550db: doc/utils: rewrite doc for time duration syntax [18:33] <cone-159> ffmpeg.git 03Timothy Gu 07master:155a5ff00d61: doc/utils: reformat doc for color syntax and add list of supported colors [18:33] <cone-159> ffmpeg.git 03Timothy Gu 07master:586b8ea24856: doc/filters: reference ffmpeg-utils manual for color and sizes options [19:25] <wm4> what's that talk about avcodec_decode_subtitle3? [19:27] <ubitux> multi frame, single packet [19:28] <wm4> I thought this idea fell into disuse [19:28] <wm4> not used with video anymore, and barely for audio [19:28] <wm4> and introducing avcodec_decode_subtitle3 would be retarded because AVSubtitle is still retarded [19:28] <ubitux> it's not [19:28] <wm4> so... just split your multi-frame packet into multiple packets? [19:29] <ubitux> check the commit, marton seems to need it [19:31] <TheRyuu> Daemon404: I haven't really looked at any ffmpeg specific changes that would be needed, my inline asm patch is against libav [19:31] <TheRyuu> if that's what you were referring to [19:41] <Daemon404> TheRyuu, there was something you were poking in ffmpeg [19:42] <TheRyuu> icl failing at preprocessing in one of the "make check" things? [19:42] <TheRyuu> that's really the only ffmpeg specific thing I have [20:17] <Daemon404> 300 get [20:23] <wm4> ubitux: someone wants to use AV_PKT_DATA_STRINGS_METADATA to update mid-stream ogg metadata, would this be ok? [20:23] <Daemon404> someone is using ogg? [20:23] <Daemon404> poor bastards. [20:23] <ubitux> good question, i wanted to have something like this a while ago [20:23] <ubitux> Daemon404: i do, especially for web radio [20:24] <ubitux> ogg/vorbis is stilly better than mp3/icy, and even saner [20:24] <Daemon404> i use aac [20:24] <wm4> yeah, web radio shit [20:24] <wm4> not sure what type of metadata is used with aac [20:24] <Daemon404> tho really, i use nothing [20:24] <wm4> maybe id3v2? [20:24] <Daemon404> because i dont do podcasts [20:24] <Daemon404> or radio [20:25] <ubitux> i stream my music to listen it from outside @home [20:25] <ubitux> wm4: anyway, dunno, send a patch? :p [20:26] <wm4> I know it's your standard reply, but I'm not quite following here, patch for what? [20:26] <ubitux> for what you were talking about? [20:26] <ubitux> did i misunderstand something, as usual? [20:26] <wm4> someone is interested in implementing this [20:27] <ubitux> why isn't he asking here / on ffmpeg-devel? [20:28] <ubitux> BBB: i'm going (and already doing actually) make extensive use of ssse3, is it ok? [20:29] <wm4> ubitux: mathstuf is the guy who wants to implement this [20:30] <wm4> and I guess he wants to know whether using AV_PKT_DATA_STRINGS_METADATA is the right approach [20:30] <ubitux> i've attempted several broken approach [20:30] <ubitux> like watching for changes in metadata in ffplay [20:31] <wm4> that wouldn't be too bad as long as there's some way to signal the application that metadata was updated [20:32] <ubitux> yes [20:32] <ubitux> and which meta as well [20:32] <ubitux> you could hack like i was doing the pointer values [20:32] <ubitux> but this is extremely awful ;) [20:33] <ubitux> the strings metadata actually sounds like a nice way to work around the problem [20:33] <ubitux> > side data [20:34] <mathstuf> ubitux: i have log messages stating that it sees the metadata and that its being put into the side_data [20:34] <mathstuf> but the app (mpv) doesnt get any packets with strings metadata side_data in it [20:34] <mathstuf> its probably being filtered out somewhere; havent traced it yet [20:35] <ubitux> ah erm, yeah the meta will be in the frame [20:35] <ubitux> (see add_metadata_from_side_data in lavc/utils) [20:35] <ubitux> meh i dunno [20:36] <mathstuf> yeah, but mpv only gets AVPacket structures, so AVFrame isn't extremely useful at that point [20:36] <ubitux> yes indeed [20:36] <ubitux> so those side data won't do [20:37] <mathstuf> i might get some time tonight to see where the packet gets dropped between the codec and the app tonight [20:39] <Daemon404> it amazes me that they still use trac [20:39] <Daemon404> rather than irc or eail [20:39] <Daemon404> email* [20:40] <mathstuf> well, bugs get lost easily on irc [20:40] <Daemon404> this isnt a bug thread [20:40] <Daemon404> it's a WIP aac improvement [20:40] <mathstuf> ah [20:40] <wm4> mathstuf: AVFrame is after-decoder, so indeed not really the right layer for metadata IMO [20:42] Action: Daemon404 loves how dimmu borgir are one of his standard tests [20:43] <Daemon404> although it makes perfect sense from an audio standpoint [20:44] <wm4> ubitux: the teletext problem is that each AVSubtitle is a separate teletext page, right? [20:45] <wm4> or how does this shit work [20:45] <ubitux> Daemon404: that's what he call white noise i presume? [20:45] <ubitux> wm4: i don't know © [20:46] <Daemon404> always ask ubitux about subttles [20:46] <Daemon404> he loves subtitles [20:47] <ubitux> :( [20:49] <wm4> he knows best [20:49] <wm4> and also doesn't care [20:50] <wm4> also avcodec_decode_subtitle3 is impending, and barely solving any problems [20:50] Action: wm4 sigh [20:50] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:9ec9d4636507: swscale/swscale_unscaled: fix alpha pointer & stride for planarRgb16ToRgb16Wrapper() [20:50] <cone-159> ffmpeg.git 03Paul B Mahol 07master:37d6b2b4e6f8: swsscale/swscale_unscaled: add GBRAP16 [20:50] <wm4> uh or am I getting this wrong, and it's just about modifying avcodec_decode_subtitle2 [20:50] <wm4> whatever [20:51] <ubitux> wm4: i suggested avcodec_decode_subtitle to avoid changing the behaviour :p [20:52] <Daemon404> wm4, solving? [20:52] <Daemon404> i read it as causing [20:52] <Daemon404> :> [20:52] <wm4> damn right [20:52] <wm4> because now you'd need a decode loop and save the current subtitle AVPacket to get all AVSubtitles [21:54] <durandal11707> ubitux: what should i use now to see FATE code coverage? [21:54] <Daemon404> gcov [21:55] <durandal11707> in other words i need it to do myself [21:55] <Daemon404> i think configure has support [21:55] <Daemon404> to do it automatically [21:56] <durandal11707> as i already stated i'm not interested in doing it myself [21:56] <ubitux> erm 404 mmh [21:56] <ubitux> lemme check what's wrong [21:57] <ubitux> ah [21:57] <ubitux> doc is broken [21:58] <ubitux> it prevents the build [21:58] <ubitux> i'm going to disable doc for that one as a workaround [21:59] <durandal11707> what is error? [21:59] <ubitux> see fate [21:59] <ubitux> they're all in red [21:59] <durandal11707> RED ALERT [22:00] <beastd> C&C [22:00] <ubitux> :) [22:01] Action: JEEB plays Hell's March [22:02] <ubitux> durandal11707: a new reports will be available soon; meanwhile you can lurk in http://lucy.pkh.me/ffmpeg-coverage-snapshots/ [22:04] <ubitux> durandal11707: coverage.ffmpeg.org fixed [22:07] <ubitux> it's fun, if i run the make again the make error goes away [22:09] <durandal11707> ubitux: since when coverage.ffmpeg.org appeared? [22:10] <ubitux> quite a while ago now [22:10] <beastd> JEEBsv: would sure be fun to play that song on a bass guitar now. unfortunately i don't have one at hand :( [22:11] <durandal11707> ubitux: i have never been informed [22:13] <ubitux> now you are [22:20] <durandal11707> too late, you have been fired [22:20] <ubitux> :( [22:27] <michaelni> is @samp{@@} correct texi syntax ? [22:28] <ubitux> probably not [22:28] <michaelni> what is the correct syntax ? [22:28] <ubitux> michaelni: do you have the issue as well or do you want me to test? [22:29] <michaelni> ill push a fix in a moment, though i cant reproduce it here [22:30] <michaelni> if you want to test @samp{@@ } [22:30] <michaelni> but iam pretty sure it will pass [22:31] <ubitux> the issue was indeed from here (replacing with just @@ fixed it) [22:32] <ubitux> michaelni: yeah that works [22:32] <ubitux> michaelni: but that's visible [22:32] <ubitux> possibly followed by @ [22:32] <ubitux> :ugly: [22:33] <ubitux> -@code{[0x|#]RRGGBB[AA]} sequence, possibly followed by @samp{@@} and a string [22:33] <ubitux> +@code{[0x|#]RRGGBB[AA]} sequence, possibly followed by '@@' and a string [22:34] <ubitux> i'd better do something along those lines [22:34] <cone-159> ffmpeg.git 03Michael Niedermayer 07master:9a63a45e48ac: doc/utils: fix docs build [22:34] <ubitux> good enough [22:34] <ubitux> :) [22:34] <ubitux> thanks [23:02] <llogan> why would @samp{@@} cause any issues? [23:03] <Compn> in code? url? xml ? [23:03] <llogan> 9a63a45 [23:05] <Compn> i dont know how to access that [23:05] <Compn> $ git log -p 9a63a45 [23:05] <Compn> fatal: ambiguous argument '9a63a45': unknown revision or path not in the working [23:07] <llogan> git pull first [23:07] <llogan> Move! [23:07] <Compn> oh there it is [23:08] <Compn> who knows [23:08] <Compn> maybe it broke on some old version [23:08] <Compn> of docs thing [23:10] <llogan> author: Compn, commit: "docs thing: fix @@ junk" [23:51] <llogan> saste: did i even mention that the SPI check arrived and it deposited fine? [23:51] <llogan> i thought i did, but i couldn't find any email that i did [00:00] --- Thu Oct 17 2013

1 0

ffmpeg.log.20131016
by burek 17 Oct '13

17 Oct '13

[00:19] <cbsrobot> Maverick|MSG: sorry i was afk [00:19] <Maverick|MSG> np [00:19] <Maverick|MSG> I'm probably going to wind up trying it either tomorrow or the next day [00:19] <Maverick|MSG> just wanted to try to code proactively [00:19] <Maverick|MSG> whatever *that* is [00:22] <cbsrobot> if I remember correctly the -to option will be convertet to the duration from the starting point, maybe if you set the -ss before the input (which should print a info iirc to put it after the -i, beacuase you could have multiple -i) that changes the starttime [00:49] <MorehouseJ09> what are your thoughts on the "Warning: data is not aligned! This can lead to a speedloss" warning when using sws_scale? [00:49] <MorehouseJ09> looked this up online and found that most people said to ignore it [00:50] <llogan> it's only a warning, not an error [00:50] <MorehouseJ09> should I worry about it? Is this a video by video thing or should I go back and peak through my code to see whats up? [00:52] <llogan> i'm not sure how much of a potential speedloss there could be [00:54] <MorehouseJ09> llogan: I'm only seeing it once, not for every frame as well so I don't think its anything to worry about for now ... [05:12] <buu> Why the heck does ffprobe print to stderr? [05:13] <buu> haha https://trac.ffmpeg.org/ticket/2226 [05:16] <buu> And should I ask why -print_format json; doesn't actually work? [05:58] <lvella> can I speedup a video just by remuxing it? [05:58] <lvella> I have an mkv of 2:05 duration, but I want it to be just 2:00, can I speed it up a little? [05:58] <sacarasc> I think you can change frame rate in the container, but I am not 100% sure. [05:59] <lvella> wouldn't that mess with the audio/video sync? [06:04] <theholyduck> lvella, easiest is just to cut 5 secs [06:04] <theholyduck> :P [06:04] <theholyduck> speeding up is theoreticcally possible, but a bit annoying [06:05] <theholyduck> lvella, and no, you cant speed up the audio by remuxing [06:05] <theholyduck> you can CUT the video [06:05] <theholyduck> by remuxing [06:05] <theholyduck> mkverge lets you do a lossless cut [06:12] <lvella> ok, the audio is flac, it supports (almost) any arbitrary sampling rate [06:13] <lvella> so, how can I increase the sampling rate a little bit (without actually resampling) so it will be a little faster? [09:16] <MorehouseJ09> hmm... I'm having trouble with a bus error. [09:16] <MorehouseJ09> I successfully build a packet and write it with av_interleaved_write_frame -- which returns 0 [09:16] <MorehouseJ09> but afterwards, when I call the write_trailer() method, I get a bus error. [09:17] <MorehouseJ09> when av_interleaved_write_frame returns 0, shouldn't the packet already have been written to the file? [10:26] <Apic> A wonderful splendid Prickle-Prickle Morning (UGT) to you all. [11:46] <burek> hi, is it possible to specify the source udp port for the rtp output, something like: ffmpeg -i ... -f rtp rtp://1.2.3.4:5678?localport=7777 [11:58] <allengreen> a naive question? can linux's signal nested? I mean if I am a signal handler, and in the handler, I write a access violation code, what would happen? [12:05] <Lanser> I'd like to use ffmpeg to convert a mp4/flv file into VOB. [12:06] <Lanser> How can I do it? [12:21] <relaxed> Lanser: ffmpeg -i input -target ntsc-dvd -q:v 3 -f vob output.vob [12:22] <Lanser> thank you, relaxed. may god bless your girlfriend with boobs. [12:22] <Lanser> big boobs, ofc. [12:22] <Lanser> but I am a lil confused. [12:22] <Lanser> what is this parameter? [12:22] <Lanser> -q:v 3 -f [12:23] <JEEB> -q:v sets the video quantizer [12:23] <JEEB> -f sets the output container format [12:23] <JEEB> (usually auto-set with the file name) [12:24] <JEEB> the target should already set bit rate and maxrate/bufsize so the -q:v is not 100% needed, just you overriding the rate control with another type :) [12:24] <JEEB> and because relaxed has no way of knowing which type of a DVD you want, he set the ntsc- prefix [12:24] <JEEB> instead of telling you to read the fine documentation and picking one ;) [12:25] <Lanser> what is a video quantizer, sir JEEB? [12:25] <Lanser> and -f? [12:25] <JEEB> already explained that :P Also written in the fine documentation [12:26] <Lanser> I want to start converting a video from 05:31 instead from the start. [12:26] <Lanser> is this possible? [12:26] <JEEB> quantizer basically sets the compression level (higher quantizer means the pictures are compressed more) [12:26] <relaxed> it's time to rtfm [12:26] <JEEB> indeed [12:26] <JEEB> :) [12:26] <Lanser> <Lanser> I want to start converting a video from 05:31 instead from the start. [12:26] <Lanser> <Lanser> is this possible? [12:26] <Lanser> JEEB? [12:27] <JEEB> http://ffmpeg.org/ffmpeg-all.html [12:27] <Lanser> but IS THIS POSSIBLE, jeeb? [12:27] <Lanser> that's my question. [12:27] <Lanser> is this possible? [12:27] <JEEB> I wouldn't be pushing you towards the documentation otherwise, would I :P [12:27] <Lanser> I don't wanna read all that. [12:27] <Lanser> it'd take me an hour. [12:28] <Lanser> just give me the page link [12:28] <Lanser> which gives me the parameter or something else, to start converting a video from 05:31 [12:28] <Lanser> instead of 00:00 [12:28] <Lanser> that'd be appreciated :-) [12:29] <JEEB> hint: 5.4, Main options [12:30] <Lanser> Is there any addon for Firefox which can put 2 downloads in a queue? when one download is complete, move to another instead of staying paused? [12:30] <Lanser> -t? [12:31] <JEEB> actual reading is recommended [12:31] <Lanser> is it the -t parameter, JEEB? [12:31] <Lanser> IS IT?! [12:31] <JEEB> lol [12:31] <Lanser> lol [12:32] <JEEB> you're an ass, unfortunately. And you seriously should know if that option is correct or not by reading the goddamn explanation of the option after the option :P [13:17] <burek> just to ask again in case someone knows: is it possible to specify the source udp port for the rtp output, something like: ffmpeg -i ... -f rtp rtp://1.2.3.4:5678?localport=7777 [13:25] <buu> So [13:27] <buu> Why does ffprobe print to stderr? How do you change that? Why does this guy think it prints to stdout: https://trac.ffmpeg.org/ticket/2226? And why doesn't -print_format work for ffprobe? [13:29] <buu> Oh and how do I get a text file output of the subtitles from a stream? [13:32] <saste> buu: you are confused [13:33] <saste> about text file you should write to a subtitles file (which is plain text) [13:41] <buu> saste: Confused about what? [13:41] <bouba> hi all, do you consider jpeg2000 or png as lossless codecs for thumb extracting plz ? (with -qscale:v 1 ) [13:42] <buu> saste: You mean I just need to say output.srt and I get text? [13:44] <buu> Oh. [13:44] <buu> How interesting. [13:44] <buu> I still wish it printed to stdout though. [13:45] <buu> Or json worked [13:57] <buu> saste: http://scsys.co.uk:8002/272029 [13:58] <saste> buu: yes, what do you want to print? [13:58] <saste> ffprobe -h [13:58] <buu> saste: I want the metadata and the chapter information? [13:59] <saste> buu: did you grep the manual? [13:59] <saste> you can do that [13:59] <buu> I've read the manual repeatedly [14:00] <buu> does it mention stdout someplace? [14:09] <matthias1> morning, i have one question: I'm compressing my avi file, but the fps is getting lower every time: frame= 8312 !fps= 39! q=-1.0 size= 6795533kB time=332.81 bitrate=167268.2kbits/s is this a problem? [14:10] <matthias1> at the beginning it was at about 45 and now it is at 15 [14:10] <buu> Also -pretty doesn't seem to change the output [14:11] <buu> matthias1: You're referring to fps during compression yes? That presumably just means its doing more work.. [14:11] <buu> per frame [14:17] <buu> Oh wtf [14:19] <buu> -show_chapters does something completely different HOW DOES THAT MAKE SENSE [14:24] <buu> Well, that gets bonus points for being massively confusing but it does work now [15:06] <matthias1> buu: Thanks! [15:21] <kode54> is it a bad idea in general to bundle a subset of FFMPEG with a project? I find it more convenient than packaging the entirety of the library and using the build scripts [15:21] <Apic> Does not sound bad to me. [15:21] <Apic> MPlayer did that too. ^>^ [15:21] <kode54> and since it's GPL, anyone can rebuild the whole project from scratch anyway [15:22] <kode54> I suppose I am missing out on the benefits of using the assembly bits, though [15:22] <kode54> I'll have to figure out a way to bundle a fat binary version of yasm with the repository and set the FFMPEG project to use that for .asm files [15:23] <kode54> or maybe just 64 bit binary, doesn't matter [15:23] <Apic> Still does not sound bad. [15:23] <kode54> depends on if the asm files are even designed to target 64 bit output [15:23] <kode54> ah [15:23] <kode54> are any of the .asm files designed for 64 bit output? [15:24] <kode54> I'll have to look, actually [15:26] <JEEBsv> < Apic> MPlayer did that too. ^>^ <- trying to copy the incestious relationship of ffmpeg and mplayer is _NOT_ a good idea [15:26] <Apic> ;-) [15:26] <kode54> heh [15:26] <Apic> True. [15:27] <JEEBsv> that said, no-one says that you may not have a git submodule of a certain known-to-work revision of ffmpeg in a directory of your git repository [15:27] <JEEBsv> that may be used if so wished [15:27] <kode54> kind of hard to have a git submodule in a mercurial repository [15:27] <JEEBsv> actually that's quite possible [15:28] <kode54> oh? hmm [15:28] <JEEBsv> just make sure you use the ffmpeg compilation system for ffmpeg, and to use --prefix to install the compiled binaries and headrs [15:28] <JEEBsv> *headers [15:28] <JEEBsv> and thus only use things you are supposed to use [15:29] <kode54> ah [15:30] <JEEBsv> or you could have a separate projectname-build repo [15:30] <JEEBsv> for simplified compilation [15:30] <JEEBsv> like mplayer2 and mpv have [15:30] <kode54> this is a Mac OS X audio player, and currently, I'm using a Framework project to compile the whole thing into a single dylib type object inside of a .framework bundle, which gets bundled inside a PlugIn's Frameworks directory. the Plugin .bundle gets bundled in the app's PlugIns folder [15:30] <matthias1> how can i extract only the video, without the audio from an avi file. my avi file is over 100gb big, but i do not want to lose quality [15:30] <kode54> use the -map switch to select just the video stream, and use -vcodec copy [15:31] <kode54> ie. if it's stream 0, you'd -map 0:0 -vcodec copy [15:31] <JEEBsv> you could just do -an or -vn depending on which you don't have [15:31] <JEEBsv> uhh [15:31] <kode54> at least that's what I remember [15:31] <JEEBsv> s/have/want/ [15:31] <JEEBsv> (audio/video none) [15:31] <kode54> oh, audio/video none switches, neat [15:32] <kode54> map is required for some of the stuff I end up doing, though [15:32] <JEEBsv> yes, if you have multiple streams of the type then yes [15:32] <kode54> namely, remuxing mp4 video files to mkv [15:32] <kode54> since ffmpeg doesn't seem to support iTunes subpicture streams [15:32] <JEEBsv> oh, and there's -sn as well for subtitles [15:32] <kode54> ah, sn [15:33] <kode54> also I was using -map_metadata -1 [15:33] <JEEBsv> yup [15:33] <kode54> because I was committing the grave sin of redistributing my iTunes store purchases to select friends [15:33] <kode54> you didn't hear that just now [15:34] <kode54> I love the Requiem author's reasoning for abandoning the project [15:34] <kode54> "my child is getting just old enough to become curious about what I do on the computer, but not old enough to understand the subtle nuances of civil disobedience" [15:35] <kode54> I mostly only unprotect my videos so I can freely use them with other players, like MPlayerX [15:36] <kode54> I have yet to be motivated to copy them over to my PS3 [15:36] <kode54> I don't think anyone else who uses the PS3 wants to watch my pony purchases [15:37] <kode54> nope, I'm not a brony, I just watch a cartoon series which is mildy amusing at times [15:38] <kode54> does the build system support producing Mac OS Frameworks out of the dylib/headers? [15:38] <kode54> probably little more than just assembling the dylib without an extension and the headers under a directory structure, along with an Info.plist file somewhere [15:39] <Benjojo> Good afternoon there, I am trying to compile a timelapse from images, I however made the images filenames unix timestamps. How can I get FFMpeg to read them with those names? [15:44] <Benjojo> Is it possible to do that? [15:48] <Apic> Nothing is true, everything is permissible. [15:48] <brontosaurusrex> Benjojo, not directly the last time i tryed [15:49] <Benjojo> brontosaurusrex: Oh dear. Looks like I'm going to have to try and change my image capture program! [15:49] <brontosaurusrex> Benjojo, what i did at the end is a script that makes temporary symlinks to my images with names that ffmpeg would support and then load this into ffmpeg ... [15:49] <Benjojo> brontosaurusrex: Do you happen to have that script on you? [15:49] <brontosaurusrex> I can find it, its bash [15:50] <Benjojo> brontosaurusrex: that would be great! [15:50] <brontosaurusrex> hold on ... [15:53] <brontosaurusrex> http://paste.debian.net/58351/ (probably interesting only from line 100 and on ) Benjojo [15:54] <brontosaurusrex> p.s. Only tested on OSX [16:14] <Benjojo> Thanks brontosaurusrex [16:18] <Apic> http://narf-archive.com/pix/f3366f6478d922273f64db94543488bd659802b4.jpeg [16:20] <durandal_1707> ;) [16:25] <Apic> =) [17:05] <Apic> http://bash.org/?4281 [17:05] <Apic> ww [17:14] <bouba> hello, some guru knows a better param than "-vf fps=fps=1/10" to extract thumbnails each 10sec ? it seems like ffmpeg drops frames to keep CFR [17:14] <bouba> plz [18:10] <brontosaurusrex> bouba, what is the input codec/stream type? [18:10] <bouba> brontosaurusrex>, h264 [18:11] <brontosaurusrex> ok, and this are huge/longer files? [18:11] <brontosaurusrex> also what is the needed thumbnail resolution? [18:12] <bouba> huge files give me a better precision than small ones [18:12] <bouba> resolution : 2048x1024 [18:13] <bouba> but the input is not 25fps [18:13] <bouba> more like 24,98... [18:13] <brontosaurusrex> ok, so you need thumbnails at 2048x1024 px resolution? [18:13] <bouba> yeah [18:13] <bouba> i tried various methods [18:14] <bouba> -r 1/10 [18:14] <bouba> -vf fps=fps=1/10 [18:14] <bouba> http://ffmpeg.org/trac/ffmpeg/wiki/Create%20a%20thumbnail%20image%20every%2… [18:15] <bouba> i also specified a lossless output with "-qscale:v 1" [18:15] <brontosaurusrex> ok, how about converting your input stream to something I frames only, like prores and then going for the fps=1/10 [18:16] <bouba> sorry, "something I frames only"...I-frame ? [18:16] <bouba> never used that [18:17] <bouba> brontosaurusrex, do you have a link/method for me please ? [18:18] <brontosaurusrex> http://en.wikipedia.org/wiki/Video_compression_picture_types#Intra_coded_fr… [18:19] <brontosaurusrex> Like i said i would try your method fps=1/10 method, but would do a temporary prores intermediate and then feed that [18:20] <bouba> thx, i'll try [20:01] <buu> So if I use -codec libmp3lame, why does the output say encoder : Lavf55.19.102 [20:01] <JEEBsv> because that's the muxer [20:03] <buu> oh [20:03] <buu> How about major_brand : mp42 ? [20:04] <JEEBsv> that's part of the "mp4" format itself [20:04] <JEEBsv> the brand as which the file is supposed to be read as [20:04] <JEEBsv> there can be multiple if the file fits under multiple specifications [22:46] <ac_slater> Hey guys, this isnt an ffmpeg issue but I think someone in here will be able to help me. I'm interested in taking RAW image data from a camera sensor and creating an RTSP stream. Does anyone know a library capable of this? (assume linux) [22:46] <ac_slater> (and I use RAW and RGB camera data.... kind of unimportant, but a detail) [22:49] <ac_slater> And I dont ever have a video container, I want to pipe the raw data into an RTSP pipeline of sorts. Anyway I'll stop flooding the channel. Any help/advice is appreciated. [22:58] <llogan> ac_slater: http://ffmpeg.org/ffmpeg-protocols.html#rtsp [22:59] <ac_slater> llogan: right. I'm looking for some development tools though. [23:00] <ac_slater> Maybe I should be on #ffmpeg-dev [23:00] <llogan> that's for development of FFmpeg and patch discussion [23:01] <llogan> (and it's #ffmpeg-devel) [23:03] <ac_slater> llogan: ah right.I must have misunderstood. It looks like the link you sent me was just a list of protocols ffmpeg can support. [23:03] <ac_slater> Are there APIs (and libraries) for these things (libavformat?) [23:04] <llogan> oh, the libraries. might find something useful in doc/examples [23:04] <llogan> or you ask on libav-user mailing list: http://ffmpeg.org/mailman/admindb/libav-user [23:05] <ac_slater> llogan: this is perfect. For some reason I didnt even think to actually look at libavformat. I'm new to RTSP so this should be interesting [23:05] <ac_slater> llogan: thanks for the advice. [23:06] <llogan> i have no experience with API usage, so I can't give you much more info [23:06] <ac_slater> it's ok, you'd helped a bunch. Thanks. [23:06] <ac_slater> you've * [23:16] <kode54> another of my peeves is that the switches for specifying start/end positions for selecting a range of an input are in timestamps [23:16] <kode54> there were times when I wanted to chop videos on exact frames [23:17] <kode54> I would be transcoding anyway, since the original would be in some weak compression that was fast enough for my capture machine to pull a full 60fps [23:18] <kode54> superdump: ahoy thar [23:19] <llogan> kode54: did you see if the select filter can do what you want? [23:19] <kode54> I hadn't [23:20] <llogan> select="gte(n,1234)" or something like that [23:20] <kode54> I should try that if I ever need to do some chopping [23:20] <llogan> http://ffmpeg.org/ffmpeg-filters.html#select_002c-aselect [23:25] <superdump> hello kode54 [23:25] <superdump> did you used to hang around foobar2000-related channels? [23:25] <kode54> still do [23:25] <superdump> and hydrogenaudio? [23:26] <superdump> :) [23:26] <kode54> and there [23:26] <superdump> good good, my memory works [23:26] <superdump> how goes it? [23:26] <kode54> just fine [23:26] <superdump> good good [23:26] <kode54> saw Gravity yesterday, man was that a ride [23:27] <superdump> cool :) [23:28] <kode54> also forked a Mac player called Cog a couple of weeks ago and have pushed over 150 commits to it [23:28] <superdump> why forked [23:28] <kode54> the original repository was dead in 2010 [23:28] <superdump> and i'd forgotten about cog :) [23:29] <kode54> then after pushing about 50 or more updates [23:29] <kode54> I noticed that mamburu@bitbucket already forked and effectively took it over [23:29] <kode54> the official update feeds redirect to his [23:29] <kode54> one of the plugins is based on FFMPEG [23:29] <kode54> I've done some silly things with that one [23:30] <superdump> are you going to merge with that bitbucket one? [23:30] <kode54> like modifying the libavformat MP3 parser to drop silly notions of seek tables [23:30] <kode54> I am on bitbucket [23:30] <kode54> but no idea if I can merge with either of the repositories in my current state [23:31] <kode54> since I manually imported updates from the other fork, since a number of them would have conflicted with changes I already made [23:31] <kode54> anyway, the FFMPEG thing [23:31] <kode54> I rewrote the seek function in the MP3 parser so it walks the entire stream [23:31] <kode54> and I added support for reading the delay/padding from iTunsMPB comments [23:32] <kode54> and also the seeking operation will try to rollback by up to 8 frames and discard that many frames worth of samples in addition to part of the last frame for a sample exact offset [23:32] <i336_> hi everyone. I have a rather basic graphics card in this PC atm, and while I can hear audio in the HD MP4 video I just downloaded off youtube, the a video window that opens is completely black. mplayer has options like '-vo x11' that cause it to render directly to X11 without optimization, and '-framedrop' to keep video playback 'fast', if choppy. What are ffmpeg's equivalents? I can't get my head around the plugin architecture :P [23:32] <i336_> s/a video/video/ * [23:33] <kode54> not sure I want to import the vorbis decoder and parser into my plugin though [23:33] <kode54> since some people on the foobar2000 forum have encountered problems with it [23:33] <i336_> (graphics optimization, that is - like opengl, which doesn't seem to want to work atm) [23:33] <superdump> hmm [23:33] <kode54> recent foobar adopted its own copy of ffmpeg with some codecs and parsers and demuxers enabled [23:34] <kode54> with the binaries renamed because random idiot codec pack updaters will find the common libav*.dll and replace them with their own copy [23:34] <superdump> orly? interesting [23:34] <kode54> and foobar's is not binary compatible with the official [23:34] <kode54> calling convention differences or something [23:35] <kode54> instructions to build it are posted on the site, along with the package of source which was used to build the latest bundled [23:35] <kode54> anyway [23:35] <superdump> still windows only i see [23:35] <kode54> the vorbis decoder fails on some really old vorbis files [23:35] <kode54> yeah [23:35] <kode54> portability is probably a 2.0 thing, if ever [23:36] <kode54> and even then, people will have to either open source their components, or provide binaries for every goddamn supported platform [23:36] <kode54> Peter already has a PFCv2 that works on Win/Mac/Linux [23:36] <kode54> but I doubt that foobar will appear there unless he decides to make a generic GUI framework as well [23:37] <kode54> then it will be total rewrites for all the components that have any Win32 code [23:37] <superdump> PFC? [23:37] Action: i336_ was going to say the same thing [23:37] <kode54> his foundation classes which are included with the SDK [23:38] <llogan> i336_: ffplay only supports sdl (if that is what you're asking). [23:38] <i336_> llogan: oh :< [23:38] <llogan> AFAIK [23:38] <kode54> the PFC v1 included with fb2k is only compatible with Windows [23:38] <i336_> and apparently doesn't provide any error feedback/detection if the video refuses to play [23:39] <i336_> ffplay <video I just downloaded off youtube.mp4> [23:39] <i336_> that's all [23:39] <llogan> and there is no console output at all? [23:39] <i336_> no, sorry, no video output - there is console output, but ffmpeg doesn't seem to realize the video isn't playing :P [23:39] <i336_> or at least it isn't telling me, lol [23:40] <llogan> i'm sorry, but i don't quite understand what the issue is [23:40] Action: i336_ runs "ffplay video.mp4" [23:40] Action: i336_ gets sound [23:40] Action: i336_ gets a video window that is completely black [23:40] Action: i336_ wonders what to do next [23:40] <i336_> :P [23:41] <llogan> you show the complete console output [23:41] <superdump> kode54: sounds like probably not then [23:41] <i336_> if you want... [23:41] <superdump> anyway, i was just swinging through [23:41] <superdump> have fun :) [23:41] <superdump> o/ [23:41] <i336_> http://sprunge.us/UFiD [23:42] <i336_> looks like normal output, but I can haz absolutely no video content :< [23:43] Action: i336_ notes that the video is 1080p but that his display is 1024x768 [23:43] <llogan> do you see anything with: ffplay -f lavfi -i testsrc [23:43] <i336_> oooooo, cute :> yes I do [23:44] Action: i336_ points out that maintaining that causes X to use 17% CPU and ffplay ~30% in one thread, 13% in another, 9% in another and 5% in yet another [23:45] <llogan> ffplay -f lavfi -i testsrc=s=hd1080 [23:45] <i336_> for a total of about 70% CPU (xmms chews cycles even when paused >.>) [23:45] <i336_> black && 100% CPU usage [23:46] <llogan> then i guess it's not the original video and I have no other ideas [23:46] <i336_> mmm :/ [23:46] <i336_> I also fiddled with that command and 'hd720' doesn't work either, which is depressing :P [23:47] <llogan> maybe someone on ffmpeg-user will have an idea. [23:47] <i336_> oh ok. [23:48] Action: i336_ discovers https://trac.ffmpeg.org/wiki/FancyFilteringExamples in the meantime [23:48] <i336_> coooool :> [23:48] Action: i336_ begins to understand why /usr/local/bin/ffplay is 11675564 bytes (ie 12MB) [00:00] --- Thu Oct 17 2013

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