December 2013 - Ffmpeg-devel-irc

ffmpeg-devel.log.20131225
by burek 26 Dec '13

26 Dec '13

[00:47] <cone-255> ffmpeg.git 03Michael Niedermayer 07master:f07ca542e371: avcodec/vmdav: return the amount of data that has been unpacked from lz_unpack() (as well as errors) [02:43] <cone-255> ffmpeg.git 03Michael Niedermayer 07master:3cfa310c5de5: avformat/ape: zero seektable&bittable and warn when they where only partially filled [02:43] <cone-255> ffmpeg.git 03Michael Niedermayer 07master:6aa50374bf27: avformat/nut: store version in the context [02:44] <cone-255> ffmpeg.git 03Michael Niedermayer 07master:804ea14b3564: avformat/nut: add minor_version field with version>=4 [02:44] <cone-255> ffmpeg.git 03Michael Niedermayer 07master:459db5127180: avformat/ape: free packet on avio_read() failure [12:47] <cone-705> ffmpeg.git 03Diego Biurrun 07master:d06ca2040c0f: error.h: Do not circularly depend on avutil.h [12:47] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:2c0b246b5bab: Merge remote-tracking branch 'qatar/master' [13:53] <cone-705> ffmpeg.git 03James Almer 07master:0d944ee34349: matroskadec: Fix bug when parsing realaudio codec parameters [15:03] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:2a5fb0b13e21: avformat/mov: check avio_read return in mov_read_dvc1() [17:48] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:ac480cb58dbe: avformat/isom: free extradata on failure to read it [17:48] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:ee4e80509312: avformat: add ff_get_extradata() [17:48] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:482b2042cdb0: avformat/adxdec: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:b59740eae537: avformat/aiffdec: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:9e5fa1e55d18: avformat/apc: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:f04fc386dc23: avformat/apetag: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:e93bc3efbf75: avformat/avidec: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:5ce04c54ddcf: avformat/bink: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:773f2a563f7a: avformat/cafdec: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:c633aeba45ab: avformat/ffmdec: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:d8d5258bb7e8: avformat/flvdec: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:215759a9252b: avformat/idcin: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:28cc7062c876: avformat/isom: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:3e5a7dffe803: avformat/mov: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:a1ce41c9a4ec: avformat/mpc: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:6f7daae04315: avformat/mpc8: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:0cedc6c066de: avformat/nutdec: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:bb6101d4de74: avformat/nuv: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:0b9ca70243c4: avformat/riffdec: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:61883a2e896a: avformat/rl2: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:fd335aa3260d: avformat/rmdec: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:2e2cab6e4f10: avformat/rsd: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:6fb29535fd54: avformat/vc1test: use ff_get_extradata() [17:49] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:239a0543a66b: avformat/westwood_vqa: use ff_get_extradata() [18:28] <cone-705> ffmpeg.git 03Stefano Sabatini 07master:7619a87cc8b9: lavfi/volume: support volume expression and per-frame expression evaluation [18:28] <cone-705> ffmpeg.git 03Stefano Sabatini 07master:97aa55463404: lavfi/volume: implement process_command() callback, with the volume command [18:28] <cone-705> ffmpeg.git 03Stefano Sabatini 07master:ca0d8e839e48: lavfi/volume: extend volume logging [18:53] <cone-705> ffmpeg.git 03Stefano Sabatini 07master:b80cf460903f: doc/protocols/udp,lavf/udp: apply minor documentation fixes [18:53] <cone-705> ffmpeg.git 03Stefano Sabatini 07master:702d9a23b3d3: lavf/tcp: honor listen option value in a tag [18:53] <cone-705> ffmpeg.git 03Stefano Sabatini 07master:1b4da43ce024: doc/protocols/tcp,lavf/tcp: apply minor fixes to TCP protocol documentation [18:54] <Compn> mega weird audio test file :D [18:54] <Compn> ehe [18:56] <saste> carl still using the mplayerhq.hu ffmpeg-devel address... [18:59] <Compn> does that bother you ? [18:59] <michaelni> saste, Test filter-volume failed. [18:59] <michaelni> Segmentation fault [18:59] <Daemon404> [12:56] <@saste> carl still using the mplayerhq.hu ffmpeg-devel address... <-- and still refusing to even use git p roperly [18:59] <Daemon404> and follow his own preachings [18:59] <Daemon404> being a hypocrite etc [19:01] <michaelni> saste, libavfilter/af_volume.c:390 [19:01] <michaelni> vol->var_values[VAR_NB_CONSUMED_SAMPLES] += buf->nb_samples; [19:01] <michaelni> print buf [19:01] <michaelni> $1 = (AVFrame *) 0x0 [19:01] <wm4> Daemon404: I think he did start using git format-patch [19:01] <Daemon404> wm4: wow [19:01] <Daemon404> mind = blown [19:01] <wm4> awesome huh [19:01] <wm4> his latest patch isn't though [19:01] <wm4> so he's still grumpy about it [19:08] <saste> michaelni, i'll fix is asap [19:14] <cone-705> ffmpeg.git 03Stefano Sabatini 07master:baeda2bf92a5: lavfi/volume: fix NULL reference in filter_frame() [21:02] <cone-705> ffmpeg.git 03Michael Niedermayer 07master:565f786d1da1: ffprobe: Dont clear AVFrame between uses. [21:34] <cone-705> ffmpeg.git 03Carl Eugen Hoyos 07master:e079661d23ff: ffmpeg_opt: Fix -target vcd [21:42] <JEEB> decided to try and report a bug with WMP / related Media Foundation filters in case of using ffmpeg + libx264 in mp4 :D http://social.technet.microsoft.com/Forums/windows/en-US/fed910ba-0b25-48f1… [21:46] <nevcairiel> i would try to rip out the pasp atom, just in case [21:52] <JEEB> this also led me to think about what would happen if the stuff in tkhd and stream/pasp + stream resolution would differ? do you first apply the pasp/stream SAR, and then apply to that the size in tkhd? :D [00:00] --- Thu Dec 26 2013

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ffmpeg.log.20131225
by burek 26 Dec '13

26 Dec '13

[00:00] <jnvsor> And vpx for that matter [00:00] <sim590> ogg is a codec too ? [00:00] <jnvsor> It's rediculous [00:00] <jnvsor> Yeah, ogg theora [00:00] <sim590> I thought about ogg as a container [00:01] <jnvsor> Yeah, but it's mainly used for theora video and vorbis audio [00:01] <jnvsor> But again - not even close to libx264 [00:01] <sim590> ;) [00:02] <sim590> I tried to add -s 1920x1080 but it seems to much for my little ASUS PC1015px. It screen is only 1024x600 but I have another 1920x1080 monitor plugged on it [00:03] <sim590> do you think I can tweak something in order to be able to record 1920x1080 ? [00:04] <jnvsor> Hmm maybe - on my system I downscale the video to save disk space [00:04] <jnvsor> Not sure if that would help a CPU bottleneck though [00:04] <sim590> does it have to do with the graphic card too ? [00:04] <jnvsor> No, it's almost all CPU [00:05] <sim590> is it possible to ask GPU to do some computing ? It seems pretty much like a graphical computing.. But, Idk [00:06] <jnvsor> In theory yes, in practice I've got no idea how to do that and it would probably be a pain :P [00:06] <jnvsor> Try this - it downscales the video before encoding it [00:06] <jnvsor> http://pastebin.com/FubazTyW [00:06] <jnvsor> change it to mkv before you run of course xD [00:11] <sim590> jnvsor: video outputs correctly.. it seems but audio is just really compressed and goes really fast [00:11] <sim590> it's not synced [00:20] <sim590> jnvsor: I'll just stick to recording my little 1024x600 screen [00:20] <sim590> jnvsor: thanks for all the help ! [00:33] <jnvsor> sim590: Compressed and goes really fast? [00:34] <jnvsor> Do you use pulseaudio? [00:40] <sim590> alsa [00:42] <jnvsor> So you have no pulseaudio on your system at all? [00:43] <sim590> noe [00:44] <sim590> I got libpulse package though [00:44] <jnvsor> Yeah if you don't have a pulse server there's not much point [00:44] <sim590> description: A featureful, general-purpose sound server (client library) [00:44] <sim590> oh [00:45] <sim590> on archlinux, pacman says: Required By : chromium ffmpeg fluidsynth gst-plugins-good mplayer mumble phonon-qt4 speech-dispatcher [00:46] <jnvsor> Well anyway, audio... There are options to change things about it like frequency `-ar` [00:46] <jnvsor> But I haven't used alsa in so long I don't know how to diagnose that issue :P [00:47] <sim590> jnvsor: thanks ! I'll try to play around with the -ar option [06:04] <i336_> Hi everyone. I'm attempting to capture my speaker audio output as it leaves the soundcard - as in, capture data going from software to the soundcard - so I can stream it over the network. This is not the fastest system so I don't have JACK or PulseAudio installed, and would rather not install it if I can avoid it. So far I've managed to get faint audio if I turn my microphone up all the way and record from ALSA hw:0,0, but that's pro [06:04] <i336_> (...probably just the mic lines getting interference from the speaker out :P) [06:04] Action: i336_ stabs irssi for the 23450982nd time [06:05] <i336_> Is it possible to "sniff" the audio data as it goes through ALSA...? [06:05] Action: i336_ gets an idea [06:05] Action: i336_ asks #alsa too :P [06:56] <i336_> I figured it out :P google ftw! and "capture device" settings ftw too :> [11:59] <dwirc> Hi folks. [11:59] <dwirc> Happy xmas for those who celebrate it. [12:00] <dwirc> Have a question regarding FFMPEG and h.264 streams (over RTSP)... tried default avconv from Ubuntu 13.10, ffmpeg from CentOS 6, and a fresh build on 12/24. Same results all 3 ways. [12:01] <dwirc> I've got a camera (hardware H.264 encoding with audio). What I want to ultimately do is pull the H.264 stream via RTSP (only method available from the camera), and restream that to an RTMP server. [12:04] <dwirc> Baby steps though... When I use FFMPEG, I set the input, tell it to copy the video codec, switch the audio codec to mp3, it gets about 30 seconds in, then hangs, every time. I've let it sit for hours there, and it never continues. [12:05] <dwirc> cmd line: ffmpeg -i rtsp://camip:554/12 -acodec libmp3lame -ar 22050 -f flv test.flv [12:06] <dwirc> Doesn't matter what I specify for the output paramters seemingly (-vcodec copy, or specifically x264...)... it always hangs about 30 seconds in. [12:06] <dwirc> ANyone seen this behavior with surveillance cams before? I searched, but didn't find anything similar in the past year or so. [13:39] <relaxed> dwirc: maybe -re before the input would help? [13:39] <relaxed> are you using a recent version? [20:32] <killer> hey , I m trying ffmpef -i file.mp4 | grep "Duration" but it is not filtering and giving me same result as ffmpeg -i file.mp4 [20:33] <sacarasc> killer: Try using ffprobe instead. [20:34] <DeadSix27> not much of a linux user [20:34] <DeadSix27> but didnt, "ffmpeg -i file.mp4 2>&1", work? [20:34] <DeadSix27> and then greping duration i mean. [20:37] <killer> I have been trying it for some time but it didn't worked but it seems to work now , I was forgetting the mp4 input after -i [20:37] <killer> oops [20:37] <kirbir> Marry Christmas to All! Help me please, is there a way to crop video as circle. Like here http://www.youtube.com/watch?v=S6C0aL8FpYo ? Google can't help me. Thank you. [21:03] <killer> ffmpeg -i file.mp4 -y file.mp3 2>&1 | grep "time=" (to get the time duration of the converted video continously) but it seems grep only shows after ffmpeg has completed converted [21:33] <DeadSix27> in ffplay, how do the filters work? (For example ass) [21:33] <DeadSix27> I tried to run ffplay by using: ffplay -i test.mkv -vf "ass=test2.ass" [21:34] <DeadSix27> but it doesnt open a video player/viewe thing. [21:35] <DeadSix27> in case you want the output: http://pastie.org/private/c8bjt686dpapo6ooamggw [21:41] <DeadSix27> Oh my fault.. it actually does play.. but i thought ffplay and ffmpeg share the same font ache [00:00] --- Thu Dec 26 2013

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ffmpeg-devel.log.20131224
by burek 25 Dec '13

25 Dec '13

[01:03] <michaelni> BBB, ubitux did you see the vp9 valgrind fate failures ? (http://fate.ffmpeg.org/report.cgi?time=20131223161213&slot=x86_64-archlinux…) [01:05] <kierank> lol http://support.apple.com/kb/HT5959?viewlocale=en_US [01:19] <BBB> michaelni: yes, ubitux told me, it looks like a bug in valgrind [01:21] <michaelni> hmm, ok [01:27] <BBB> I mean, the output is completely ok in fate so I find it hard to believe I'm doing something undefined there [02:04] <cone-252> ffmpeg.git 03Carl Eugen Hoyos 07master:05c3c568dccd: Read pictures in id3v2.2 [02:21] <BBB> ubitux: patch for parallelmode=1 bug on ML, please review (and michaelni: feel free to apply when OK'ed) [02:21] <BBB> should fix that vp9 issue on trac also [02:21] <BBB> haven't tested, but sounded identical in scope [02:22] <BBB> https://trac.ffmpeg.org/ticket/3228 [02:22] <BBB> that one [02:22] <BBB> and just tested yes it fixes it [03:42] <BBB> ubitux: and your fuzz crash also fixed, thanks for the sample [03:42] <BBB> now off to do speed measurements for 16x16 sub-IDCTs and then on to 32x32 IDCT [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:edca16f1af63: ffserver: strip odd chars from html error messages before sending them back [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:f684bbf224df: avcodec/jpeg2000dec: check transform equality in MCT [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:93f26b79926c: avcodec/jpeg2000dec: prevent out of array accesses in pixel addressing [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:47f8497837eb: avcodec/jpeg2000dec: fix context consistency with too large lowres [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:074ebfacf40f: avfilter/ff_insert_pad: fix order of operations [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:72f1907c96b5: avcodec/avpacket/av_packet_split_side_data: ensure that side data padding is initialized [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:fce2cfbdcfbd: avcodec/utils: add some saftey checks to add_metadata_from_side_data() [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:83e4aa3e7c57: avutil/opt: initialize ret [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:bdf6e6fff41b: avcodec/jpeglsdec: check err value for ls_get_code_runterm() [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:c88bdac4605c: avformat/thp: fix variable types to avoid overflows [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:4324d7badeb6: avformat/thp: force moving forward [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:b545d11d498c: avcodec/jpeg2000dec: non zero image offsets are not supported [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:70fcea3b7761: h264: Do not treat the initial frame special in handling of frame gaps [04:01] <cone-252> ffmpeg.git 03Kostya Shishkov 07release/2.0:3a3b5ae4c053: mpegvideo: Fix swapping of UV planes for VCR2 [04:01] <cone-252> ffmpeg.git 03Diego Biurrun 07release/2.0:6b683be641b6: mpeg12dec: Remove incomplete and wrong UV swapping code for VCR2 [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:28ac4e91dcfa: avutil: rename lls to lls2 [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:0cd61c7f7d10: rename new lls code to lls2 to avoid conflict with the old which has a different ABI [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:4f93400db1f2: avutil: reintroduce lls1 as the 52 ABI needs it [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:ae81a0e32de6: avcodec: move end zeroing code from av_packet_split_side_data() to avcodec_decode_subtitle2() [04:01] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:bd9dcb411d2e: avcodec/jpeg2000dec: Check precno before using it in JPEG2000_PGOD_CPRL [04:02] <cone-252> ffmpeg.git 03Paul B Mahol 07release/2.0:a9382fc15c15: avcodec/libopusenc: change default frame duration to 20 ms [04:02] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:66a9edfcf677: do O(1) instead of O(n) atomic operations in register functions [04:02] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:172f92976752: avutil/log: skip IO calls on empty strings [04:02] <cone-252> ffmpeg.git 03Michael Niedermayer 07release/2.0:b4552cc9b8c3: update for FFmpeg 2.0.3 [04:36] <cone-252> ffmpeg.git 03Carl Eugen Hoyos 07fatal: ambiguous argument 'refs/tags/n2.0.3': unknown revision or path not in the working tree. [04:36] <cone-252> Use '--' to separate paths from revisions [04:36] <cone-252> refs/tags/n2.0.3:HEAD: Read pictures in id3v2.2 [05:09] <cone-252> ffmpeg.git 03Jan Gerber 07master:b2597042e5a7: avcodec/libopusdec: Set codec->delay to pre_skip not fixed value [07:25] <ubitux> BBB: oh awesome [07:34] <ubitux> kierank: ahah nice [08:14] <cone-698> ffmpeg.git 03Michael Niedermayer 07release/1.1:848af79decd2: nutenc/write_index: warn if 2 consecutive keyframes have the same PTS and discard the 2nd [08:14] <cone-698> ffmpeg.git 03Peter Ross 07release/1.2:7ba102d00814: riffenc: add option to ff_put_bmp_header to ignore extradata [08:14] <cone-698> ffmpeg.git 03Peter Ross 07release/1.2:9f3135b30bfb: avformat/riffenc: indent [08:14] <cone-698> ffmpeg.git 03Peter Ross 07release/1.2:fa45feefad0e: wtvenc: populate VIDEOINFOHEADER2 [08:14] <cone-698> ffmpeg.git 03Michael Niedermayer 07release/1.2:5c502e5d4141: nutenc/write_index: warn if 2 consecutive keyframes have the same PTS and discard the 2nd [08:14] <cone-698> ffmpeg.git 03Peter Ross 07release/2.1:c3f962840793: riffenc: add option to ff_put_bmp_header to ignore extradata [08:14] <cone-698> ffmpeg.git 03Peter Ross 07release/2.1:f27895db0f10: avformat/riffenc: indent [08:14] <cone-698> ffmpeg.git 03Peter Ross 07release/2.1:94c3f8165c6a: wtvenc: populate VIDEOINFOHEADER2 [08:14] <cone-698> ffmpeg.git 03Michael Niedermayer 07release/2.1:b432043d556f: nutenc/write_index: warn if 2 consecutive keyframes have the same PTS and discard the 2nd [12:08] <cone-528> ffmpeg.git 03Ronald S. Bultje 07master:4a55bffad32e: vp9: fix bug in updating of coef probabilities with parallelmode=1. [12:08] <cone-528> ffmpeg.git 03Ronald S. Bultje 07master:acafbb4dd260: vp9: fix crash if segmentation=1, keyframe/intraonly=1 and updatemap=0. [12:49] <ubitux> BBB: sorry, got another one :) [12:49] <saste> ubitux, you gonna review volume patches? [12:50] <ubitux> saste: i don't think so, unless you insist [12:50] <saste> ubitux, i don't insist, i just ask because you was interested with the patch at some point [12:51] <ubitux> saste: give me a minute i'll scroll review it [12:51] <saste> as usual, i'm mostly interested in interface issues, functional bugs can be fixed later [12:56] <ubitux> saste: i think the reconfiguration of the dsp callbacks in filter_frame() should be in the EVAL_MODE_FRAME scope [12:57] <ubitux> i'm not motivated for a longer review now though [12:57] <ubitux> sorry :p [13:00] <saste> ubitux, isn't it? set_volume() -> volume_init() [13:04] <ubitux> in filter_frame(), aren't you doing the switch case outside the eval frame mode? [13:05] <ubitux> also, i think you're breaking the is_writable() case [13:06] <ubitux> in case of volume==1.0, and frame !writable, you need copy the data to the new buffer [13:06] <ubitux> i think you break that case with the goto end thing [13:14] <BBB> ah bah [13:14] <BBB> ok fine [13:15] <BBB> ubitux: wanna start reviewing the asm changes? [13:16] <BBB> I'll measure later but they did appear slightly faster in quick overall timings (not much, 0.02 sec over 9.2 seconds total, but still, that's 0.2% or so [13:35] <wm4> why does ffmpeg AVFrame have the number of channels (AVFrame.channels), while Libav doesn't need it? [13:38] <saste> wm4, libav computes channels from layout IIRC [13:38] <saste> in case layout is not known, that can be an issue [13:40] <wm4> well, I don't think Libav are that dumb to miss this kind of essential corner case [13:52] <saste> wm4, 23fc4dd6e7e150ea163a867dfaee2062ade90b74 [13:54] <wm4> well, whatever [13:54] <nevcairiel> libav does indeed not support unknown channel layout [13:55] <wm4> I noticed that my own code uses AVCodecContext.channels as fallback [13:56] <wm4> a concept of unknown channel layouts would be helpful indeed [13:56] <wm4> elenril wanted to add a new channel API that would include this [13:57] <wm4> merry christmess [14:20] <ubitux> BBB: sure [14:20] <ubitux> are you done? :) [14:23] <BBB> ? [14:23] <BBB> yes [14:25] <BBB> except for commit msg ofc [14:25] <ubitux> i'll look tonight probably [14:30] <saste> in what part of the code are the url options passed? [14:30] <saste> *parsed [14:41] <saste> uh... [14:41] <saste> so we have some protocols which accepts options only in the url (e.g. udp), and others which allow to set the options in the context as well (e.g. tcp) [14:43] <saste> i also like how there is no real validation, see all the uses of strtol [14:47] <saste> nice also how the list of options supported in the url can be different from the ones supported in the context [15:43] <ubitux> saste: do you think we could make showspectrum/showwaves output a picture of the full stream? [15:45] <saste> i was trying that and it was somehow problematic [15:46] <ubitux> :( [15:46] <saste> it depends how much you can "compress" the diagram [15:46] <saste> alternatively you can create several images and chain them [15:47] <ubitux> in the current state? [15:47] <saste> i'm not sure you can do it with showspectrum [15:47] <ubitux> or with a patch? [15:59] <saste> ubitux, in theory you can squeeze the output and tile them [15:59] <saste> in practice there was a problem [16:00] <saste> i don't remember exactly what was it [17:27] <wm4> michaelni: elenril says reusing an AVFrame should not leak (ffprove AVFrame issue) [17:40] <nevcairiel> the important extra would be clearing it before giving it back to the API [18:34] <BugMaster> http://blog.medialooks.com/0LyGa6/ <- "The filter uses LGPL builds of FFmpeg, is available for commercial licensing and can be used in commercial applications." while using postproc which can be build only with GPL license :-|. at least all other ffmpeg libs looks like to be really compiled with LGPL license [18:40] <Compn> BugMaster : you realize thats basically just ffdshow :P [18:41] <BugMaster> I don't use it. I simply found link to it today and read this blog in try to find how it better than LAV Filters [18:44] <BugMaster> and there argument is mostly is that source/demuxer/decoder filter all-in-one instead of separate entities. iirc ffdshow wasn't all-in-one in this sense (it was multiple filters in one library) but may be I remember wrongly [18:46] <Compn> youre right it wasnt all in one [18:46] <Compn> i mean, you had to call it 4 or 5 times to demux, decode, etc [18:46] <Compn> i think [18:47] <JEEBcz> ffdshow-tryouts had no demuxing capabilities [18:47] <Compn> hmmm [18:47] <Compn> you sure ? [18:47] <JEEBcz> yes [18:47] <Compn> i guess the mpcht had the demuxers [18:47] <Compn> havent used wmp or ffdshow in years anyway [18:48] <JEEBcz> yeah, mpc and mpc-hc had their own demuxers [18:48] <JEEBcz> ye olde gabest filters [18:53] <kierank> BugMaster: lol yet another "pro" thing violating the gpl [19:00] <Compn> BugMaster : wow they claim mpeg4, h264 and mpeg2 reverse playback [19:00] <Compn> thats a highly requested feature and rare [19:01] <Compn> why people want reverse playback who knows :D [19:01] <kierank> Compn: i-frame only [19:01] <Compn> Smooth reverse playback [19:01] <Compn> Another consequence of our design is the reverse playback feature, which is usually available for I-frame encoding formats only (without temporal compression), such as AVC-i and MJPEG. [19:01] <Compn> With the Multi-Format Source DirectShow filter it is available for most of the supported formats including MPEG-2, MPEG-4 and H.264.* [19:02] <Compn> i could be wrong [19:02] <Compn> i didnt test :P [19:02] <Compn> available for 1450 euro [19:03] <Compn> whoaaaaaaaaaaa [19:03] <cone-661> ffmpeg.git 03Alex Sukhanov 07master:251c96a70b0d: avformat/matroskadec: Fix start_time [19:13] <Compn> violating because it doesnt say how to build it kierank ? [19:13] <Compn> or providing any source ? [19:13] <kierank> provide source [19:17] <kierank> lots of companies troll OSS about violating patents [19:17] <kierank> I can fight back by saying they violate (L)GPL [19:21] <nevcairiel> such all in one things are usually rather terrible from a dshow perspective [19:21] <nevcairiel> of course it makes stuff easier, but why bother with dshow then. :) [19:22] <kierank> because you can get a team in india to get something working on the cheap [19:23] <kierank> without knowing what you are doing [19:32] <nevcairiel> i have been thinking about relicensing my stuff to lgpl since a few companies asked [19:32] <nevcairiel> was hoping yadif relicense goes through since that's the only gpl i use in ffmpeg [19:34] <Daemon404> nevcairiel: i dont really understand why you would have to be lgpl? [19:34] <Daemon404> i ean theyre just going to use it in a filter graph [19:35] <nevcairiel> they are afraid of GPL [19:35] <Daemon404> against all logic? [19:35] <Daemon404> "gpl doesnt affect this use" [19:35] <Daemon404> "BUT I A AFRAID!" [19:35] <Daemon404> s/A/AM/ [19:35] <Mavrik> uh [19:35] <nevcairiel> and honestly i have no clue if COM stuff can be used as gpl, its a bit grey [19:36] <Mavrik> if you link against other libraries you can quickly have tons of trouble with GPL [19:36] <Mavrik> since it's incompatible with pretty much anything [19:36] <Daemon404> Mavrik: a filter graph is not linking [19:36] <nevcairiel> well you dont link [19:36] <Daemon404> nevcairiel writes directshow filters. [19:36] <nevcairiel> they load at runtime [19:36] <Mavrik> ah yea :) [19:36] <nevcairiel> indirectly through windows even [19:36] <Mavrik> but you're right, GPL is like a bogey man for larger companies :/ [19:37] <Daemon404> depends on the comapny [19:37] <kierank> except when they're outsourcing to india [19:37] <kierank> indians love their --enable-gpl --enable-nonfree [19:37] <Daemon404> some companies are ok with gplv2 [19:37] <kierank> and the company asks no questions [19:37] <Mavrik> had to persuade one of the TV broadcasters to not do a rewrite of whole ffmpeg just because they were afraid of GPL -_- [19:37] <Daemon404> ive ever seen one ok with gplv3 [19:37] <Daemon404> never* [19:37] <Mavrik> even though they were their own customer :P [19:38] <Plorkyeran> sufficiently paranoid companies won't even use GPL tools Just In Case [19:38] <Plorkyeran> even if they're not distributing anything related to them [19:39] <Daemon404> if your google, you rewrite everything yourself and license it as apache2 [19:39] <Daemon404> youre* [19:39] <Daemon404> with c++-only APIs [19:39] <kierank> lol [19:40] <kierank> oh lord the medialooks guy is advertising it on linkedin [19:40] <kierank> it is taking a lot for me not to troll him [19:40] <Plorkyeran> bad C++-only APIs [19:40] <Plorkyeran> that don't actually take advantage of anything useful from c++ [19:42] <Daemon404> Plorkyeran: yes [19:43] <Daemon404> itll literally be exactly like a c api [19:43] <Daemon404> but with :: instead of _ [19:43] <Daemon404> all static funcs [19:43] <Daemon404> (is that the right term in c++?) [19:43] <kierank> public [19:43] <kierank> ? [19:43] <Plorkyeran> it's because they actually want to write C [19:43] <Plorkyeran> but C is not an approved language at google [19:44] <Daemon404> cant even write extern "C" for the API? [19:44] <Plorkyeran> they could provide a C api with a c++ implementation [19:44] <Daemon404> thats easy for them [19:44] <Daemon404> int thing_somefunc(int a) { return thing::somefunc(a); } [19:44] <Plorkyeran> but the issues with c++ apis don't matter for them internally [19:45] <Plorkyeran> so I guess they just don't bother [19:45] <Daemon404> yep.... [19:45] <Daemon404> they also do not really do releases [19:45] <Daemon404> just always use HEAD. [22:11] <jnvsor> Is BGR0 pixel format deprecated? If not I've found a bug [22:11] <nevcairiel> its not [22:11] <jnvsor> Righty then - bug report on it's way [22:11] <nevcairiel> but how does a bug depend on that [22:12] <jnvsor> Well I've tracked it down with gdb but the logic is a bit opaque so I wasn't sure if it was supposed to do that or not. I presumed not. [22:17] <Daemon404> clearly it was supposed to do this, instead of that. or maybe this, then that. [22:19] <jnvsor> sws_init_context gives a warning if the context->srcFormat changes since the start of the function. But before then it calls sws_setColorspaceDetails which calls handle_formats which calls handle_0alpha which changes the color space [22:19] <jnvsor> or no not color space, pixel format [22:20] <jnvsor> But since the warning says the pixel format is deprecated, and you said that bgr0 isn't, at the very least the warning is wrong [22:29] <cone-255> ffmpeg.git 03Carl Eugen Hoyos 07master:d63e9943615f: avformat/mov: Do not compute a grayscale palette for cinepak in mov. [22:29] <cone-255> ffmpeg.git 03Carl Eugen Hoyos 07master:691dec62011f: Allow stream-copying grayscale mov files. [22:55] <cone-255> ffmpeg.git 03Michael Niedermayer 07master:51fed95dde7a: swscale/utils: fix wrong deprecated message with rgb0 [23:29] <cone-255> ffmpeg.git 03Michael Niedermayer 07master:bb9f55163f17: avcodec/eatgv: use av_mallocz() for frame_buffer [00:00] --- Wed Dec 25 2013

1 0

ffmpeg.log.20131224
by burek 25 Dec '13

25 Dec '13

[11:05] <viric> can ffmpeg create a picture from a audio wave? [11:06] <viric> maybe I should use sox [11:06] <viric> not a sprectrogram [11:06] <ubitux> yes it can [11:06] <ubitux> see showwaves and showspectrum multimedia filters [11:06] <viric> I mean a picture like what audacity generaets [11:07] <viric> aha, I'll see [11:07] <viric> hm but it creates *video*. Not a picture. [11:08] <ubitux> -frames:v 1 [11:08] <ubitux> and it will create one picture [11:08] <ubitux> of course it might be tricky to have the whole spectrogram [11:08] <viric> right. [11:09] <viric> In fact I'd like one png per every 2 seconds of audio [11:09] <viric> in waveform [11:10] <viric> so that concatted pngs show the proper concatted waveform [12:05] <jnvsor> Hi, swscaler is giving me deprecated pixel format errors, how do I remove them? http://pastebin.com/uAsBiCRK [15:46] <zafu> hi, can I losslessly cut a mpeg4 video with ffmpeg? (along group of pictures boundaries) [15:54] <DeadSix27> JEEB: took me 1,5 days.. but the build succeeded. [15:57] <DeadSix27> (the build didnt take 1,5days though [16:01] <saste> zafu, segment [16:02] <zafu> saste: where can I find it? [16:03] <saste> zafu, segment muxer, ffmpeg-formats(1) [16:04] <zafu> thanks [16:05] <zafu> is it better than "-acodec copy -vcodec copy" ? [16:44] <DeadSix27> how exactly can i output raw video? [16:44] <DeadSix27> i mean, probably via -c:v rawvideo [16:44] <DeadSix27> but do i have to use a specific container? [16:44] <DeadSix27> (or should i just use .yuv as extension?) [18:02] <DeadSix27> now i see why you guys say libfdk produces way better AAC [18:27] <odie5533> http://ffmpeg.org/ffmpeg-filters.html#concat says the default value for a (number of audio streams) is 0, but when I use -filter_complex "concat" I do get concatenated audio output. Am I missing something here? [18:29] <Hejkki> hello, how do i concat 2 videofiles? I tried this already: avconv -i concat:MVI_2774.MOV\|MVI_2775.MOV -c copy kakka.MOV [18:29] <Hejkki> but the result gives me the first video only [18:29] <odie5533> Hejkki: https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20%28join,%20merge%29%2… [18:31] <Hejkki> Page How to concatenate (join, merge) med not found [18:31] <odie5533> Hejkki: You need a better IRC client. Try Googling "How to concatenate (join, merge) media files" and click the first link [18:32] <Hejkki> better than irssi? [18:32] <odie5533> Apparently. [18:33] <odie5533> It chopped off the last characters of the url [18:34] <jnvsor> My irssi works fine - could be the terminal emulator? [18:38] <Hejkki> odie5533: i tried that also, same result [18:38] <odie5533> Hejkki: what did you try? [18:39] <Hejkki> ffmpeg -i "concat:file1.mov|file2.mov" -c copy output.mov [18:40] <odie5533> That only works for "the concat protocol works at the file level. Certain files (mpg and mpeg transport streams, possibly others) can be concatenated. [...] In general, the demuxer is the better option." [18:40] <odie5533> So you should probably try the demuxer instead. [18:41] <Hejkki> ok [18:42] <Hejkki> but isn't demuxing same as separating audio and video? [18:43] <odie5533> It is. I think it separates the streams, concatenates them, then outputs them [18:44] <bencoh> (remux them before outputting) [18:44] <odie5533> Whereas the concat protocol (command you were trying to use) just concatenates the files at a file-level. I don't think mov supports that kind of operation [18:46] <Hejkki> ok [18:47] <viric> ls [18:47] <viric> oops [18:47] <odie5533> No files found. [18:48] <Hejkki> ok, somehow i managed to get the video concatinated to one .ts file [18:48] <Hejkki> now the same for audio [18:48] <Hejkki> the -ts has only video, no audio [18:48] <Hejkki> .ts even [18:48] <odie5533> Hejkki: Are you following the first part of that page where it says "Concat demuxer" "Instructions" [18:49] <odie5533> Because the only mention of ts files on that page is for using them as an intermediate with the Concat Protocol, not the Concat Demuxer. [18:51] <Hejkki> i got the command from somewhere else, and lost the command and page since my battery went off :( [18:51] <Hejkki> laptop [18:51] <Hejkki> but i do have the .ts file anyhow [18:54] <odie5533> Hejkki: https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20%28join,%20merge%29%2… [18:54] <odie5533> there's the page again [18:54] <odie5533> I'm not sure having a ts file is really going to help you.... but maybe. Who knows! [19:01] <Hejkki> odie5533: it helps because i got the video concatinated. now i also have concatinated audio, but separate files... and the laptop is now remuxing them [19:02] <Hejkki> ok, works, thanks !! [19:16] <DelphiWorld> Hi People [19:16] <DelphiWorld> ffmpeg -re -i rtmp://radio.gmo.ps/live/livestream -vn -acodec libfdk_aac -ar 44100 -ac 48k -f flv rtmp://172.16.10.100:1935/myapp/alray [19:16] <DelphiWorld> is suposed to supress the video stream, and push audio only stream to rtmp server [19:16] <DelphiWorld> but is not supressing the video [19:17] <DelphiWorld> Any CLUE? [19:20] <DelphiWorld> can someone tel me how to supress video? [19:21] <DelphiWorld> ffmpeg -re -i rtmp://radio.gmo.ps/live/livestream -acodec libfdk_aac -ar 44100 -ab 48k -ac 2 -vn -f mp3 rtmp://172.16.10.100:1935/myapp/alray [19:35] <LithosLaptop> maybe try the map option? [19:35] <LithosLaptop> https://trac.ffmpeg.org/wiki/How%20to%20use%20-map%20option [19:38] <LithosLaptop> btw for 48Kbit/s I would add -profile:a aac_he_v2 [19:51] <jnvsor> Anyone know how to get rid of the swscaler warning here: http://pastebin.com/uAsBiCRK ? [19:55] <DelphiWorld> LithosLaptop: explain for aac? [20:16] <DelphiWorld> hi saste the saster:) [20:31] <jnvsor> Do you think #ffmpeg-devel would know more about the swscaler warning? [20:41] <jnvsor> Can someone explain the src/dst_format/range options here in more detail? : http://www.ffmpeg.org/ffmpeg-scaler.html [20:47] <JEEBsv_> picture format for input/output and the value range (TV or PC) for input/output. Generally with YCbCr TV range isusual, and with RGB PC range is usual [20:47] <JEEBsv_> picture format is probably one of the PIX_FMT values defined in swscale's headers [20:48] <jnvsor> It says src_format should be an integer, I presume the enum in libavutil/pixfmt.h [20:48] <jnvsor> But when I add random integers nothing changes so I can't really tell if I have it working [20:48] <JEEBsv_> never used that interface so I have no idea :P [20:49] <jnvsor> Well it's giving me a warning when I try to scale something. This warning is all over the internet with no instructions on how to remove it. I hate warnings xD [20:51] <JEEBsv_> never seen that myself personally... I wonder if it's using the J PIX_FMT inside? [20:51] <JEEBsv_> no idea \o/ [20:52] <jnvsor> :( [20:52] <relaxed> the scaler filter has flags= and `man ffmpeg-scaler` mentions dst_format [20:53] <jnvsor> Yeah, that html page is just the man page [20:53] <relaxed> "Set destination pixel format (must be expressed as an integer)." <- wtf? [20:53] <jnvsor> Yep. Not helpful :) [20:53] <jnvsor> Wierd thing is if I set range [20:53] <jnvsor> for instance to "3": Value 3.000000 for parameter 'src_range' out of range [0 - 1] [20:53] <jnvsor> I thought the point was to set the range :O [20:54] <relaxed> they're really trying to kitchen-sink us with all these options. [20:55] <jnvsor> kitchen sink is good, but methinks the plumbing is broken [20:55] <relaxed> libav has nice plumbing, but not so many sinks [20:55] <relaxed> pick your posion [20:56] <relaxed> er, poison [20:57] <jnvsor> avconv is broken for my usage scenario, still I hate seeing error messages even if the command needs to be verbose [20:57] <relaxed> I'm curious about what's broken [20:57] <jnvsor> x11grab device [20:58] <jnvsor> `-framerate` option does nothing [20:58] <jnvsor> So avconv is useless for screencapture since video desyncs if there's a dropped frame [20:58] <relaxed> so it just does 25 fps? [20:58] <jnvsor> yep, no matter what I set it to [20:58] <relaxed> tried -r 25? [20:59] <jnvsor> Yes but that makes it worse - that says "The input will come in at 25fps" so if there's a freeze for instance it will just plop the next frame down without writing the intermediates [21:00] <jnvsor> ffmpeg works fine, I just want to get rid of all the warnings and this one is taunting me :( [21:01] <nycat> Is there any way outside of a while loop to have ffmpeg keep rebroadcasting an asf stream when it changes media? [21:26] <PowerCC> question, to burn-in subtitles -vf subtitles=path what would i have to do if I want to burn-in hebrew .srt? is it more of an X11 font issue? [21:26] <PowerCC> xQuartz (OS X) project suppose to support hebrew/latin [21:27] <jnvsor> I'd start by making sure the srt file is utf8 [21:28] <PowerCC> iconv [21:28] <PowerCC> done [21:28] <jnvsor> And then I wouldn't know what to do and I'd come here and ask for help :) [21:29] <PowerCC> ha [21:29] <PowerCC> thx [21:30] <PowerCC> it's most likely an x11 issue, xQuartz with so little apple backing anymore. [21:32] <PowerCC> jnvsor: what do you use ffmpeg mostly for? [21:33] <jnvsor> screencasting, converting files, and of course I use it indirectly through vlc etc etc [21:33] <PowerCC> lib.x264 [21:33] <PowerCC> it's too bad OpenCL can't be used for more than lookahead [21:33] <PowerCC> my cpu gets so taxed. [21:33] <jnvsor> `-threads 4` :) [21:34] <PowerCC> always -threads 4 although last night i forgot to indicate and it ran as 6 threads woke up to a nice and warm room ;) [21:36] <PowerCC> pm'ed instead of pastebin [21:37] <relaxed> PowerCC: https://trac.ffmpeg.org/wiki/How%20to%20burn%20subtitles%20into%20the%20vid… [21:37] <jnvsor> I think it's an i##n issue relaxed [21:37] <PowerCC> thx relaxed it's where i started [21:37] <relaxed> oh, ok [21:38] <PowerCC> do you recall what -data_partitioning 1 does [21:38] <jnvsor> Perhaps you need to compile an ffmpeg with extra options for multilingual or something [21:38] <PowerCC> used to always use it. [21:38] <PowerCC> i don't believe it's an ffmpeg issue [21:38] <PowerCC> i strongly believe it's within font_cnfig [21:39] <jnvsor> Got any other programs using font_config that don't show up right? (Try the video + sub in vlc, it's based on ffmpeg) [21:39] <PowerCC> ok, good idea. [21:39] <PowerCC> sec [21:40] <PowerCC> i did have to do: export FONTCONFIG_PATH=/opt/X11/lib/X11/fontconfig [21:40] <PowerCC> yesterday to get it to even take. [21:41] <PowerCC> shall i add /library/font? so it will be force to substitute? [21:43] <PowerCC> shows up fine in VLC ;( [21:45] <jnvsor> try `ldd /usr/bin/vlc` [21:45] <jnvsor> probably won't have fontconfig in it [21:45] <PowerCC> just edited font.config [21:45] <PowerCC> added another library [21:46] <PowerCC> k [21:51] <PowerCC> [AVFilterGraph @ 0x101d044a0] Error initializing filter 'subtitles' with args 'hebrew_test.srt' [21:51] <PowerCC> [Parsed_subtitles_0 @ 0x101d04120] Error decoding: Invalid data found when processing input (ignored) [21:51] <PowerCC> -bash: syntax error near unexpected token `(' [21:52] <jnvsor> sounds like a quote problem somewhere [21:52] <jnvsor> msg me the line again [21:52] <PowerCC> maybe i can try -sub_charenc option [21:57] <PowerCC> pm [21:59] <jnvsor> Looks like irssi doesn't support UTF8 unless what you pmed at the end was supposed to be random chars :P [21:59] <relaxed> of course it does [22:00] <PowerCC> paste that by mistake [22:00] <PowerCC> part of the .srt [22:00] <PowerCC> IRC won't support that [22:02] <jnvsor> Yeah idk - I steer clear of anything more complicated than ascii - it's probably a library somewhere. Maybe fontconfig. Try recompiling ffmpeg without fontconfig and see if it works or not. [22:02] <jnvsor> If it doesn't, try compiling fontconfig manually to see what happens [22:03] <PowerCC> thx [22:07] <PowerCC> bingo [22:07] <PowerCC> correct source encoding for hebrew is Windows-1255 [22:11] <PowerCC> not sure why wouldn't work with ISO-8859-8 [22:11] <PowerCC> which is hebrew [22:12] <PowerCC> but Windows-1255 did the trick with iconv [22:19] <PowerCC> thx jnvsor for all your efforts. [22:19] <PowerCC> happy xmas! [23:21] <sim590> anything wrong with that command http://ix.io/9ve ? [23:22] <sim590> considering that `arecord -l ` outputs that: http://ix.io/9vf [23:22] <sim590> There's something wrong with the audio part. [23:23] <sim590> I get this http://ix.io/9vg [23:31] <sim590> I changed the container format and it works.. but now I have a buffer underflow error.. [23:32] <jnvsor> Well first, change `-r` into `-framerate` so x11grab handles dropped frames for you [23:33] <sim590> yeah.. I just dropped -r and -s and it works but video quality is bad. [23:35] <sim590> ok.. now that's better.. but now I want a better resolution [23:35] <sim590> jnvsor: here's what I got now : `ffmpeg -f x11grab -y -framerate $FRAMERATE -i :0.0 -f alsa -ac 1 -i hw:1 -sample_fmt s16 ./out.mpg` [23:35] <sim590> and what container should I use ? Do you have an idea ? [23:35] <sim590> and why ? [23:36] <jnvsor> Codec is more important than container [23:36] <jnvsor> `-c:v libx264 -preset "ultrafast" -qp 0` is the fastest lossless you'll ever see [23:36] <sim590> is that good -vcodec ffvhuff ? [23:36] <jnvsor> IIRC ffvhuff is just raw footage - HUGE filesize [23:37] <sim590> okok [23:37] <jnvsor> as for container, it's secondary to codec - libx264 should work with mkv, avi, wmv, mp4 [23:39] <sim590> okk, now I get that libx264 is unknown :\ [23:39] <jnvsor> hmm, what's your configure line? (ffmpeg -v 9 -loglevel 99) [23:40] <sim590> ffmpeg -f x11grab -c:v libx264 -preset "ultrafast" -qp 0 -y -framerate $FRAMERATE -i :0.0 -f alsa -ac 1 -i hw:1 -sample_fmt s16 ./out.mpg [23:40] <sim590> http://ix.io/9vi [23:41] <jnvsor> Ooh you have your inputs and outputs mixed up there, let me rewrite that for you [23:42] <sim590> ok I have juste rewrite it. I get what you say [23:42] <sim590> but it only records audio now [23:42] <sim590> ffmpeg -f x11grab -y -framerate $FRAMERATE -i :0.0 -f alsa -ac 1 -i hw:1 -sample_fmt s16 -c:v libx264 -preset "ultrafast" -qp 0 ./out.mpg [23:44] <jnvsor> http://pastebin.com/4RaryQ6p [23:46] <sim590> jnvsor: works, but when I do mplayer ./out.mpg, I get only audio [23:47] <sim590> Video: no video [23:47] <jnvsor> what is the output of `ffmpeg -i out.mpg` [23:47] <sim590> if I change mpg to mkv, it has video ! [23:48] <sim590> jnvsor: http://ix.io/9vj [23:54] <jnvsor> Yeah: http://en.wikipedia.org/wiki/Comparison_of_container_formats [23:54] <jnvsor> mpg only supports mpeg1 and mpeg2 video [23:55] <jnvsor> yeah divx, mpg, mpeg and webm are the only containers that don't support h264 :) [23:56] <sim590> so mkv seems like a pretty good container [23:56] <sim590> it says it supports "virtually anything" [23:57] <jnvsor> So does avi [23:57] <sim590> yeah [23:57] <jnvsor> container really doesn't matter - as long as it supports your codecs it's fine :) [23:59] <sim590> it seems like it ! ;) But I heard ogg is free.. so that's a good choice for free software. [23:59] <sim590> I don't know about mkv [23:59] <sim590> http://en.wikipedia.org/wiki/Matroska seems like it's free too ! :D [00:00] <jnvsor> libx264 is *so* much faster than ogg [00:00] --- Wed Dec 25 2013

1 0

ffmpeg-devel.log.20131223
by burek 24 Dec '13

24 Dec '13

[01:08] <cone-897> ffmpeg.git 03Michael Niedermayer 07master:866c44d4b0f9: avcodec/wavpack: clear remainder of data in case of error in wv_unpack_mono/stereo() [02:00] <cone-897> ffmpeg.git 03Michael Niedermayer 07master:8b285f03f70e: avcodec/utils: fix memleak on avcodec_open2() failure [02:40] <BBB> wth, that parallelmode bug is a threading bug [02:41] <BBB> it works fine with -threads 1 [12:53] <cone-252> ffmpeg.git 03Tim Walker 07master:6c0e83511705: doc: fix a coding style error in the developer documentation. [12:53] <cone-252> ffmpeg.git 03Michael Niedermayer 07master:5b4c7b7dcf93: Merge commit '6c0e83511705e7f8f5100e7852a1e7fd16aa8877' [12:58] <cone-252> ffmpeg.git 03Tim Walker 07master:57ead8449e44: doc: fix link to the ISC license text in the developer documentation. [12:58] <cone-252> ffmpeg.git 03Michael Niedermayer 07master:5a8b4ed580e7: Merge commit '57ead8449e44bd07b3d4a5bd42f1aab64566f92e' [13:12] <cone-252> ffmpeg.git 03Tim Walker 07master:10d982480f9a: lavu: fix typo in documentation. [13:12] <cone-252> ffmpeg.git 03Michael Niedermayer 07master:5a1bfa7ed72d: Merge remote-tracking branch 'qatar/master' [14:23] <cone-252> ffmpeg.git 03Peter Ross 07master:d08ea75a825a: wtvenc: test avio_size() validity [15:58] <cone-252> ffmpeg.git 03Peter Ross 07master:fcbb94712d98: riffenc: add option to ff_put_bmp_header to ignore extradata [15:58] <cone-252> ffmpeg.git 03Peter Ross 07master:f93b0abe4041: avformat/riffenc: indent [15:58] <cone-252> ffmpeg.git 03Peter Ross 07master:6da21c1f8190: wtvenc: populate VIDEOINFOHEADER2 [16:37] <cone-252> ffmpeg.git 03Michael Niedermayer 07master:5c3079aaa94b: avformat/iss: check sscanf() return code [17:18] <RioKuro> Hey guys, I've got a problem. I'm trying to build HD Broadcast complient mxf files from dpx sequences. The dpx files have BT709 primaries but use full range levels. I'd like to scale them to studio levels but it's just not working. I've tried using the scale filter with in_range and out_range and I've also tried to use the -color_range option but still nothing. hope you guys can help. [17:18] <RioKuro> oh i'm using ffmpeg 2.1.1 btw [18:30] <cone-252> ffmpeg.git 03Michael Niedermayer 07master:57c63c92c579: MAINTAINERS: drop 2.0 from the releases that i maintain [18:30] <cone-252> ffmpeg.git 03Michael Niedermayer 07master:d94633409bb1: avcodec/libopencore-amr: remove av_free coded_frame. [18:30] <cone-252> ffmpeg.git 03Michael Niedermayer 07master:d5128fce3864: avcodec/apedec: more checks for k [18:30] <cone-252> ffmpeg.git 03Michael Niedermayer 07master:99978320c0dc: avcodec/apedec: use av_fast_padded_malloc() [18:46] <cone-252> ffmpeg.git 03Stefano Sabatini 07master:dd00f9cac373: tools/zmqsend: use valid NULL log context [18:46] <cone-252> ffmpeg.git 03Stefano Sabatini 07master:129ce05b0228: tools/zmqsend: rename ctx to zmq_ctx [21:20] <cone-252> ffmpeg.git 03Peter Ross 07master:686b6dc33a36: wtv: make ff_format_videoinfo2 visible to wtv encoder [21:21] <cone-252> ffmpeg.git 03Peter Ross 07master:e4be07657049: wtvenc: use codec tag + MEDIASUBTYPE_BASE_GUID, if there is no suitable guid [21:21] <cone-252> ffmpeg.git 03Peter Ross 07master:e93a73cbef53: wtvenc: output ff_format_none if ff_put_wav_header fails [22:37] <wm4> is it acceptable to direct users of my own project to the ffmpeg samples ftp to upload samples? [22:38] <wm4> it's probably hard to make users differentiate between things that may or may not be potential libav* bugs, but the probability is generally high [22:43] <ubitux> if it can trigger bugs in a third party app, even if it doesn't affect directly ffmpeg, i think it's interesting to have the sample [22:43] <ubitux> but i don't own the samples server [22:43] <ubitux> ...nor i'm paying the storage & bw [22:44] <beastd> wm4: neither do I own/pay for the sample server but IMHO it is fine, if you follow up on it e.g. fix it in your code and clean up or report bug against ffmpeg . [22:45] <wm4> beastd: yes, that's the intention [22:48] <beastd> wm4: Then I guess it is fine, but maybe wait a little bit for others to voice their opinion [23:14] <cone-252> ffmpeg.git 03Michael Niedermayer 07master:95953c1cfd51: avcodec/wavpack fix *void pointer arithmetic [00:00] --- Tue Dec 24 2013

1 0

ffmpeg.log.20131223
by burek 24 Dec '13

24 Dec '13

[01:17] <klaxa> oh by the way, parsing failed because the filename had some special characters apparently [01:17] <klaxa> i extracted the subs and called the file subs.ass and that worked [01:17] <klaxa> rather undesirable [01:18] <sacarasc> Like spaces? [01:21] <klaxa> >Applying option vf (set video filters) with argument subtitles=filename=[WhyNot] SaintYoung Men The Movie [BD 720p AAC][12DEEE77].ass. [01:21] <klaxa> >[Parsed_subtitles_0 @ 0x33e2440] Setting 'filename' to value '' [01:21] <klaxa> i escaped it properly in the shell [11:18] <JEEB> hmm, it must be flying right over my head right now, but how did I set width to automatically make the picture 1:1 SAR when I have height set to 720? [11:20] <JEEB> sar*iw/(1080/720) [11:20] <JEEB> this does it but isn't exactly elegant :D [11:21] <viric> sar*iw/(width/height) ? [11:21] <viric> hm [11:21] <viric> nah, forget it. I'd have to think that. But a formula of that kind is the way, I think [11:21] <JEEB> scale=sar*iw/(ih/oh):min(ih\,720) [11:22] <JEEB> this seems to do it :) [11:22] <JEEB> this also works for things that are smaller than 720 [11:24] <JEEB> (you probably want to also add the chroma subsampling thing to this, too) :D [11:38] <viric> JEEB: and what about encoders requiring multiples of 8, 16, ... in width or height? [11:38] <viric> I've hit that annoyance sometimes [11:39] <JEEB> viric, you just use the same way as you do with the chroma subsampling example in the scale filter's docs [11:39] <JEEB> for chroma subsampling I just ended up with scale=trunc(sar*iw/(ih/oh)/hsub)*hsub:trunc(min(ih\,720)/vsub)*vsub [11:40] <JEEB> truncate with division by the limitation, and then multiply by that limitation again [11:45] <viric> o [11:45] <viric> k [11:48] <clever> JEEB: making some decent progress on the rpi decoding and rendering [12:06] <juke> hi [12:07] <juke> is there a way to launch a script when an image is extract from video ? [12:57] <DeadSix27> is it possible to create a slideshow of a SINGLE image, with background image in ffmpeg? [12:57] <DeadSix27> (isnt really a slideshow anymore) [12:58] <viric> I can't understand what you ask for [12:59] <DeadSix27> Is it possible to create 1fps video, with only an image and music as source? [13:00] <DeadSix27> e.g -i <path to image> -i <path to audio> -c:v libx264 -r 1 -c:a mp3 out.mp4 [13:00] <DeadSix27> would that work? [13:01] <viric> you may need to use -shortest [13:01] <relaxed> ffmpeg -i audio -loop 1 -i image -shortest -c:v libx264 -c:a copy out.mp4 [13:01] <viric> to stop encoding when the shortest (V or A) stops. The image will be in an endless loop, so... [13:01] <DeadSix27> ah right. [13:02] <DeadSix27> would it use the image dimensions? [13:02] <DeadSix27> or do i have to define them manually? [13:02] <viric> it will use those. [13:03] <DeadSix27> shortest makes it 2s long. [13:05] <relaxed> DeadSix27: -bf 0 [13:06] <relaxed> or grep the duration, omit -shortest and use -t $duration [13:06] <DeadSix27> yes i just did the -t solution [13:06] <viric> why 2s long [13:06] <viric> ? [13:07] <relaxed> b frames from libx264 [13:08] <viric> ok. why shortest doesn't make it as long as the audio? [13:08] <DeadSix27> dont ask me [13:09] <DeadSix27> hm [13:09] <DeadSix27> using -bf 0, plays the audio fine, and length is right as well, but no video is showing [13:11] <relaxed> try with -x264-params bframes=0 instead of -bf 0 [13:14] <DeadSix27> that makes the video size somehow work i mean prior to that, player just didnt "get the size" and it was like audio only [13:14] <DeadSix27> now its getting the video size, but its still just black. [13:14] <DeadSix27> sec. [13:16] <DeadSix27> relaxed: http://pastie.org/private/ws5h4yemqvg5l72ke6bg [13:17] <relaxed> you need -loop 1 before the input you wish to loop [13:17] <relaxed> otherwise you're asking for one frame [13:17] <DeadSix27> ah thought it does that automatically. [13:18] <relaxed> and you don't need 0 bframes if you're using -t $audio_duration [13:18] <DeadSix27> right. leftover [13:18] <DeadSix27> (tried with, without, together) [13:19] <DeadSix27> btw is it an issue of the experimental aac encoder, that the quality sounds always way worse compared to libvo_aacenc? [13:20] <DeadSix27> compared to using* [13:20] <relaxed> libfdk_aac is considered the best that ffmpeg supports. [13:20] <DeadSix27> think that isnt compiled in zeranoes builds (or even linux only, forgot) [13:20] <viric> isn't it incompatible with gpl? [13:21] <viric> DeadSix27: you need a special build taking out all gpl pieces [13:21] <relaxed> yes, but that doesn't stop people from building it locally [13:21] <DeadSix27> :p [13:21] <viric> right, if you mean to distribute it. [13:21] <DeadSix27> well i dont really want to go through the compiling ffmpeg [13:21] <relaxed> or just use neroaacenc and stream copy the audio with ffmpeg [13:22] <DeadSix27> would be to heavy on the end, the main reason i use ffmpeg, is because its all in one. [13:23] <viric> leave a message to the aac encoder people [13:23] <viric> "this is very uncomfortable" [13:23] <DeadSix27> on the other hand, i always wanted to try to compile ffmpeg my own [13:24] <DeadSix27> is it much work to compile it with the mentioned library support? [13:24] <DeadSix27> (compared to doing it without) [13:24] <relaxed> compiling anything on windows is a chore, in my opinion [13:24] <DeadSix27> yep, but got used to that much. [13:25] <relaxed> however, once the tools are in place it's not that difficult [13:25] <DeadSix27> but that ofc, is the main reason i usually just use zeranoes builds [13:26] <DeadSix27> anyway, for later searching, what do i have to look out for? [13:26] <DeadSix27> ffmpeg without gpl? [13:26] <viric> for me, compiling anything on windows is very hard [13:27] <relaxed> DeadSix27: if you're not distributing ffmpeg you don't have to worry about the gpl or any of that. [13:28] <relaxed> I believe zeranoe's website has links and maybe a guide on how to compile ffmpeg on windows. [13:28] <relaxed> also, https://trac.ffmpeg.org/wiki/CompilationGuide [13:29] <DeadSix27> relaxed: what i meant by that, is that i thought its some kind of "option" when compiling [13:29] <DeadSix27> "--with-gpl" = disables that speicifc aac encoder [13:29] <DeadSix27> like that. [13:30] <DeadSix27> theres ffmpeg on android o.O [13:30] <relaxed> --enable-libfdk-aac requires --enable-nonfree [13:30] <DeadSix27> ah ok [13:30] <DeadSix27> thats what i meant. [13:30] <viric> DeadSix27: why do you encode aac? [13:30] <DeadSix27> required of players. [13:30] <DeadSix27> they dont support ffmpeg/ac3 [13:30] <DeadSix27> errr [13:30] <DeadSix27> mp3* [13:30] <relaxed> most video players on android use ffmpeg's libs [13:31] <relaxed> including my favorite, mx player [13:31] <DeadSix27> but ye, where i can, i just copy the audio [13:31] <DeadSix27> to prevent Quality loss [13:31] <DeadSix27> relaxed: i usally use Moboplayer [13:32] <tholin> I have a flv file with broken pts. Is there some way to remux it to fix the timestamps? [13:34] <jnvsor> How exactly are the PTS broken? Both tracks? Scaled? [13:35] <tholin> it jumps back and forward in time [13:36] <jnvsor> Audio as well as video? [13:36] <tholin> I think so [13:41] <jnvsor> aresample looks like a good lead, but I'm not sure how to rebuild PTS from scratch or if it's even possible [13:42] <viric> you can use setpts and asetpts in filters [13:56] <tholin> what I'm really trying to do is extract an flv from a pcap dump using a buggy tool [13:57] <tholin> I'm trying to fix it but the incremental timestamps adds up too fast and gets larger than the next absolute timestamp [14:05] <DeadSix27> relaxed: Do you know where exactly this is located: https://trac.ffmpeg.org/wiki/MingwCompilationGuide#Git [14:05] <DeadSix27> relaxed: see: "fix the HOME variable, so that it will point to $MINGW_PATH/msys/1.0/home/$USER" [14:27] <DeadSix27> relaxed: nvm. [14:38] <DeadSix27> relaxed: btw im not rly thaat skilled in this, but can you compile from linux to windows? [14:38] <DeadSix27> that would make this whole thing way easier, since i have a couple linux servers [14:39] <JEEB> yes [14:39] <JEEB> use zeranoe's mingw-w64 toolchain script to build you a nice mingw-w64 toolchain for either 32bit or 64bit binaries, or both [14:40] <JEEB> http://ffmpeg.zeranoe.com/blog/?p=242 [14:40] <DeadSix27> interesting [14:41] <JEEB> basically see that you have the dependencies, and then let it roll for some time... then add the bin directories within PATH so you can use them freely [14:42] <JEEB> and then you can configure with cross-prefix and friends [14:42] <JEEB> for example with 32bit binaries, --cross-prefix=i686-w64-mingw32- --arch=i686 --target-os=mingw32 [14:43] <DeadSix27> well not so far yet. [14:43] <JEEB> (yes, the dash after mingw32 is deliberate, since you have i686-w64-mingw32-gcc for example) [14:43] <DeadSix27> i dont even met the requirements atm [14:43] <JEEB> :) [14:43] <JEEB> well, shouldn't take too long to get those installed [14:43] <DeadSix27> -> cvs yasm git flex bison [14:43] <DeadSix27> was needed. [14:43] <DeadSix27> (Didnt know git wasnt installed on this server) [14:43] <DeadSix27> (funny) [14:44] <JEEB> but yeah, just noted you an example of the three settings you have to set when cross-compiling :) [14:45] <DeadSix27> was similiar to what i had before: --enable-cross-compile --target-os=mingw32 --arch=x86 [14:46] <DeadSix27> i like my style of time-wasting.. it began with a 10s compiling of a music piece [14:46] <DeadSix27> creating* [14:46] <DeadSix27> and ended in compiling ffmpeg myself for windows [14:46] <DeadSix27> just how the heck did i get there. [15:05] <saste> DeadSix27, install the static build and you're done [15:10] <DeadSix27> ? [15:43] <jarr0dsz> hi everyone, can anyone tell me how to make an image from a .flv ? just 1 single image in size 400x400 for example [15:43] <jarr0dsz> i cannot find any info on how to create single image from movie only convert movie fully to images [15:43] <jarr0dsz> dsfadsa [15:43] <jarr0dsz> fasd [15:43] <jarr0dsz> fsad [15:43] <jarr0dsz> fasd [15:43] <jarr0dsz> see no text this comes true? [16:13] <DeadSix27> JEEB: when compiling how do i tell it, where libfdk_aac is? [16:13] <JEEB> LD_LIBRARY_PATH or --extra-cflags="-I/your/windows/prefix/include" --extra-ldflags="-L/your/windows/prefix/lib" [16:14] <JEEB> uhh, not LD_LIBRARY_PATH [16:14] <JEEB> PKG_CONFIG_PATH [16:14] <JEEB> I meant :P [16:14] <JEEB> one of those two alternatives [16:16] <DeadSix27> JEEB: thanks worked [17:06] <RioKuro> Hey guys, I've got a problem. I'm trying to build HD Broadcast complient mxf files from dpx sequences. The dpx files have BT709 primaries but use full range levels. I'd like to scale them to studio levels but it's just not working. I've tried using the scale filter with in_range and out_range and I've also tried to use the -color_range option but still nothing. hope you guys can help. [17:07] <RioKuro> oh i'm using ffmpeg 2.1.1 btw [17:59] <PowerCC> hello [18:02] <PowerCC> in order to hardcode (burn-in) subs if it's not srt it must be converted to .ass still? [18:03] <PowerCC> libass [18:03] <klaxa> i don't think so [18:03] <klaxa> wait... [18:03] <klaxa> if it's not srt t must be converted to ass? [18:04] <klaxa> hmm... the docs only mention .srt and .ass [18:08] <PowerCC> for example dvdsubs [18:08] <PowerCC> or picture based [18:09] <PowerCC> overlay video filter [18:10] <PowerCC> i am interested in doing stuff on the fly (transcode) [18:13] <PowerCC> something like [18:13] <PowerCC> ffmpeg.exe -i path-to-movie\movie.mkv -filter_complex "[0:0][0:4]overlay[v]" -vcodec mpeg2video -b 16384k -maxrate 30000k -bufsize 4096k -ab 448k -ar 48000 -ac 6 -acodec ac3 -copyts -map [v] -map 0:1 -f mpegts - [18:55] <pablocastellanos> Hi, how I can duplicate a stream and transcode it but putting it at a specific stream position? [19:19] <jnvsor> When I use a scale filter sqscaler complains about a deprecated pixel format - What can I do to remove this warning? [20:11] <jnvsor> Trying to get rid of the swscaler warnings about pixel format here, any ideas? http://pastebin.com/BNXehJ7K [20:19] <stieno> How can one establish inter stream synchronisation? [20:20] <stieno> Using rtp, rtcp video stream? [20:25] <brontosaurusrex> is there an explanation how 10bit > 8bit and vice versa works by default in ffmpeg? (dithering or what?) [21:10] <_fil_> Hi, I need to convert mp4 to mpeg in order to create a DVD with iDVD. How do I specify that I don't want to lose quality? [21:14] <sacarasc> _fil_: When encoding to a lossy format, you will always lose quality. [21:23] <Looney> quick question and I will be out of your hair [21:24] <Looney> what is the difference between neon and no neon support [21:24] <Looney> https://github.com/yixia/FFmpeg-Android/blob/master/FFmpeg-Android.sh [21:24] <viric> neon are simd instructions for ARM [21:24] <Looney> ok [21:25] <viric> neon code is likely to process things faster on an ARM that understands neon [21:25] <Looney> so, its like hardware acceleration [21:25] <Looney> ok, and what is vfp [21:26] <klaxa> something with floating point [21:26] <Looney> only asking as I need to build ffmpeg with ssl enabled for android [21:26] <klaxa> http://en.wikipedia.org/wiki/ARM_architecture#Floating-point_.28VFP.29 [21:26] <Looney> and I want to make sure my ssl build is compatible [21:26] <viric> vfp instructions are instructions for an ARM typical FPU [21:26] <Looney> ok [21:26] <viric> there are different versions even. [21:27] <Looney> so, if I build openssl with arm6 and arm7 I am good to go ? ? [21:27] <viric> without any vfp, floating point code may be written with soft fp, no hw fpu [21:27] <viric> ssl won't use vfp [21:27] <Looney> I also hope so [21:27] <viric> floating point encryption? :) [21:28] <Looney> because those computation won't be much fuss [21:28] <Looney> no, https stream :-) [21:30] <Looney> gotta get going for now [21:30] <Looney> will bug you mateys more morrow ;-) [00:00] --- Tue Dec 24 2013

1 0

ffmpeg-devel.log.20131222
by burek 23 Dec '13

23 Dec '13

[01:57] <cone-372> ffmpeg.git 03Michael Niedermayer 07master:2c635fabbf7e: avformat/isom: check avio_read() result [01:57] <cone-372> ffmpeg.git 03Michael Niedermayer 07master:9b195dd5793e: avformat/4xm: shrink packet if it was only partially initialized [04:30] <clever> Compn: and i have colors! [09:25] <ubitux> erm [09:25] <ubitux> valgrind is not happy at all& [09:28] <ubitux> BBB: i think valgrind doesn't like what you did with the stack in 8d4c616fc05f2f3c76d13594788129df72069f30 [09:29] <ubitux> see what happened in c65fe9e..cc0e2ba: http://fate.ffmpeg.org/history.cgi?slot=x86_64-archlinux-gcc-valgrindundef [09:30] <ubitux> 2089 1783 tests passing [09:32] <ubitux> ==16153== Uninitialised value was created by a stack allocation ==16153== at 0xA52F49: ??? (vp9itxfm.asm:600) [09:38] <ubitux> i don't see the problem, cglobal -512, loop1 +256*2, -512, loop2 +16*2, -32, RET (+512) [09:40] <nevcairiel> doesnt say it overreads stack, just that it handles uninitialized values [12:33] <cone-541> ffmpeg.git 03Justin Ruggles 07master:be7c323176e2: Add a libwebp encoder [12:33] <cone-541> ffmpeg.git 03Michael Niedermayer 07master:6fe5f770db3a: Merge commit 'be7c323176e2e5fcf30e3d2ff20975b2f936811b' [12:44] <BBB> ubitux: stack? valgrind doesn't check stack, only heap [12:44] <BBB> ubitux: stack check is done by the other tool [12:44] <BBB> I forgot its name [12:46] <BBB> asan [12:46] <BBB> anyway I can have a look [12:47] <BBB> it's possible that we read 4 bytes for the dc (instead of 2) and it doesn't like that [13:16] <cone-541> ffmpeg.git 03Guillaume Martres 07master:7398e0516f7f: hevc: move DSP declarations from hevc.h into hevcdsp.h [13:16] <cone-541> ffmpeg.git 03Michael Niedermayer 07master:5410a5dc6608: Merge remote-tracking branch 'qatar/master' [13:38] <BBB> ubitux: oh I have a hunch [13:38] <BBB> ubitux: let's say I make a register that holds a stack value [13:38] <BBB> ubitux: and the stack value is undefined [13:38] <BBB> ubitux: then, let's say that I pxor that register with itself [13:38] <BBB> ubitux: is the resulting value undefined? [13:40] <nevcairiel> who knows if such an analyser is smart enough to figure that out [14:44] <saste> priority=wish and type=enhancement are redundant [14:44] <saste> priority=wish IMHO doesn't make much sense [14:44] <saste> an enhancement request can have a priority level [15:06] <ubitux> BBB: no valgrind checks the stack as well for uninit [15:06] <ubitux> BBB: note that the uninit appears in a later function [15:07] <ubitux> (typically some C in DSP code) [15:07] <BBB> yes, it's because it thinks we write uninitialized data in blockq, which is then used in the next transform [15:07] <BBB> but the data is not uninitialized, it is a zero [15:10] <ubitux> so you think it's a valgrind bug? [15:10] <ubitux> it's strange that it doesn't spot such similar pattern elsewhere [15:12] <cone-541> ffmpeg.git 03Mason Carter 07master:ed5bed415220: VC1: Fix intensity compensation performance regression [15:12] <cone-541> ffmpeg.git 03Michael Niedermayer 07master:64b98df38910: vc1: simplify *_use_ic initialization [15:12] <ubitux> BBB: are you sure it's not because you write only half of this reg or something? [15:12] <ubitux> so some part is left uninit [16:10] <cone-541> ffmpeg.git 03Mason Carter 07release/2.0:aeac212fda1b: VC1: Fix intensity compensation performance regression [16:10] <cone-541> ffmpeg.git 03Mason Carter 07release/2.1:7eec11463fca: VC1: Fix intensity compensation performance regression [17:58] <BBB> ubitux: not sure - but regardless, the output is fine so I think valgrind is braindead [17:58] <ubitux> dunno& [18:04] <BBB> also first version (not working ofc) of 2x2 16x16 pushed to my github [18:05] <BBB> should have it working in a few hours or so, then will fix parallelmode bug, then will make a 32x32 [18:06] <ubitux> i'm still progressing on lpf [18:11] <ubitux> BBB: btw, no faster way than punpcklbw(psraw(paddw(x,8),4), psraw(paddw(y,8),4))? [18:11] <ubitux> (doing (A+8)>>4 word-wise twice and repack) [18:14] <BBB> packuswb you mean? [18:16] <BBB> why not paddsb(x, 4) and then pand(0xfc) and then psraw? [18:18] <ubitux> oh [18:18] <ubitux> you can actually do that byte wise? [18:19] <BBB> that's why you pand [18:24] <BBB> sorry had a phone call [18:24] <BBB> the idea is to add in bytes [18:24] <BBB> then clear the lowest 3 bits [18:24] <BBB> so pand(0xf8) [18:24] <BBB> and then shift in any unit (logical, not arithmetic) [18:24] <BBB> so e.g. psrlw or psrlq or so [18:25] <BBB> (if the lower bits are cleared, they all do the same thing) [18:29] <ubitux> that will work with unsigned data right? [18:31] <ubitux> i'm doing this part typically: dst[strideb * -7] = (p7 + p7 + p7 + p7 + p7 + p7 + p7 + p6 * 2 + p5 + p4 + p3 + p2 + p1 + p0 + q0 + 8) >> 4; [18:32] <ubitux> i was doing the add word-wise, with the add and shift [18:32] <ubitux> (and then repack because it doesn't stick in one reg) [18:32] <ubitux> you're suggesting to do all the add byte-wise (so with saturation, 0xff max) [18:32] <ubitux> wwww? [18:32] <ubitux> -wwww [18:33] <BBB> oh right, you can't do that in bytes [18:33] <BBB> yes you have to do that in words [18:34] <BBB> so like you did, just punpcklbw(data, zero) -> paddw(pixels) -> paddw(8) -> psraw(4) -> packuswb [18:34] <ubitux> (yeah right packuswb for repacking might be better :p) [18:34] <BBB> ;) [18:36] <ubitux> por(packuswb(psraw(paddw(paddw(punpckhbw(x, 0), ...), 8), 4)), packuswb(psraw(paddw(paddw(punpcklbw(y, 0), ...), 8), 4))) [18:36] <ubitux> probably something like that then [18:40] <BBB> why por? [18:41] <BBB> well anyway, yes in general something like that [18:41] <ubitux> to combine the two repacked data [18:41] <ubitux> but i'm certainly thinking it wrong [18:41] <ubitux> i'm not familiar enough with those instr. [18:42] <BBB> doesn't packuswb do that? [18:42] <ubitux> ah so one packuswb call with the two? [18:42] <ubitux> mmh makes sense [18:42] <BBB> x = packuswb(x, y) is like the inverse of x = punpcklbw(data, 0); y = punpckhbw(data, 0) [18:42] <BBB> or data = packuswb(x, y) is the inverse of x = punpcklbw(data, 0); y = punpckhbw(data, 0) [18:43] <ubitux> great, then packuswb( psraw(paddw(paddw(punpcklbw(x, 0), ...), 8), 4)), psraw(paddw(paddw(punpckhbw(x, 0), ...), 8), 4)) ) [18:43] <ubitux> so no simpler way? :P [18:43] <BBB> no sorry :-p [18:44] <ubitux> :( [18:44] <ubitux> i would be grateful to have something like 64 or 128 reg [18:45] <BBB> you'll love pshufb [18:45] <ubitux> i'm surprised thre is no "round packing" instr [18:45] <ubitux> i used a lot pshufb in my previous attempt [18:45] <ubitux> but i don't see much usage here [18:46] <BBB> and palignr [18:46] <BBB> look into that [18:46] <BBB> yes rounded pack woule bs useful I've wondered about that too [18:46] <BBB> basically a combination of how we use mulhrsw and packuswb [18:46] <BBB> but maybe just with a shift amount instead of a full multiplier [18:48] <BBB> suggest that to intel ;) [18:49] <ubitux> :) [18:49] <ubitux> i'll suggest plpfvp9 instead [18:51] <BBB> uhm right [18:51] <BBB> yes [18:51] <BBB> that's highly likely to make it [18:51] <ubitux> :)) [18:52] <BBB> palignr is useful for what you're doing above though [18:52] <ubitux> i was reading that [18:52] <BBB> if you have a reg containing p7,6,5,4,3,2,1,0 and another containing q0, 1,2,3,4,5,6,7 [18:53] <BBB> you can shift them into halfcombinations using palignr [18:54] <ubitux> i think i'm doing it the other way around [18:55] <ubitux> like doing the line i pasted with all p7 + all p7 + all p7 + ... + all p4 + all p3 ... [18:55] <ubitux> reg contains only one p/q [18:55] <ubitux> (or a sum, which i can substract for the later line) [18:56] <ubitux> but i'm just probably highly confused at what you are trying to tell me :P [19:00] <ubitux> palignr/pshufb are certainly extremely useful the other way around, like i was doing previously [19:01] <BBB> hm I guess you're right, yes [19:02] <BBB> I forgot the adds are vertical [19:02] <BBB> that's actually useful [19:02] <BBB> palignr is slow [19:03] <cone-897> ffmpeg.git 03Michael Niedermayer 07master:de2a2caf4ded: nutenc/write_index: warn if 2 consecutive keyframes have the same PTS and discard the 2nd [19:31] <j-b> 'morning [19:45] <michaelni> evening [19:45] <jnvsor> evening [19:46] <jnvsor> Ran into a bug where on debian sid when compiled with --enable-libopencv and using a `-f pulse` input ffmpeg doesn't respond to SIGINT [19:47] <jnvsor> Affects avconv too. Dissapears when run under valgrind. Any ideas? [20:02] <michaelni> jnvsor, which version of ffmpeg is that ? and how to reproduce ? [20:08] <jnvsor> git HEAD - on debian sid, let me make a pastebin [20:09] <jnvsor> michaelni: http://pastebin.com/0PWKG3fG [20:10] <jnvsor> q still stops it, and running it in valgrind makes it work too [20:11] <jnvsor> I wanted to git-bisect it but I keep getting "No rule to make target" on older versions [20:15] <michaelni> jnvsor, does this also happen without --enable-shared ? [20:16] <jnvsor> I'll recompile and find out [20:22] <jnvsor> Yep, still happens [20:25] <michaelni> probably best if you open a ticket on https://trac.ffmpeg.org/ [20:26] <jnvsor> Righty then [20:30] <cone-897> ffmpeg.git 03Michael Niedermayer 07master:88956f73f29c: ffmpeg: use timestamp from before and after the decoder to detect backward moving dts for discontinuity detection [20:30] <cone-897> ffmpeg.git 03Michael Niedermayer 07master:102b794e0948: configure: support raising major version in soname [21:01] <jnvsor> How long does it usually take for a registration email from trac? [21:05] <michaelni> should be very quick [21:05] <ubitux> check your spam [21:06] <jnvsor> Hmm, I asked for a resend twice - it's been about 30 minutes since registration. I checked spam [21:07] <michaelni> hmm, beastd, are trac mails working (see above) [21:07] <ubitux> try to nag beastd [21:09] <Timothy_Gu> jnvsor: it shouldn't take very long. 5 minutes max I guess [21:11] <jnvsor> Well I've even tried the web interface *shudder* - I thought maybe you do mails in a cron job or something [21:21] <ubitux> any idea how to change an xmm reg value within gdb? [21:21] <ubitux> (gdb) set $xmm4.uint128 = 0x2a2929282a2b2c2c2c2b2c2f31333536 [21:21] <ubitux> Numeric constant too large. [21:21] <ubitux> something like that, that works [21:25] <michaelni> jnvsor, beastd: Dec 22 20:29:31 ... to=<jonathan-vola@...>, relay=mx2.hotmail.com[65.54.188.110]:25, delay=2.3, delays=1/0/0.53/0.71, dsn=2.0.0, status=sent (250 <BAY0-MC3-F49X1TMgPE00456129(a)BAY0-MC3-F49.Bay0.hotmail.com> Queued mail for delivery) [21:26] <jnvsor> Sent already or just queued? I've still got nothing here [21:26] <michaelni> iam no mail expert but status=sent to me sounds like it was sent [21:27] <jnvsor> Yeah, don't suppose you could nab the token so I could activate it? [21:28] <ubitux> > hotmail [21:28] <jnvsor> I know, I know :P [21:29] <Timothy_Gu> Once Hotmail/Outlook ate one of my library overdue notice, and I had to pay $10... [21:29] <jnvsor> Outlook? Give me a little credit :P [21:31] <Timothy_Gu> I used Hotmail since I was 5 (seriously) and now I completely dumped that. [21:32] <jnvsor> Can't think of a good alternative that would warrant the change. [21:52] <cone-897> ffmpeg.git 03Mason Carter 07master:61ae993957f8: vc1: Fix intensity compensation performance regression [21:52] <cone-897> ffmpeg.git 03Michael Niedermayer 07master:c19f193c6b23: Merge remote-tracking branch 'qatar/master' [22:21] <cone-897> ffmpeg.git 03Ivan Kalvachev 07master:1c63aed232fe: Convert XvMC to hwaccel v3 [22:32] <jnvsor> Allrighty, I've narrowed the sigint bug down to a combination of gnutls and libopencv [22:50] <beastd> jnvsor: Did it arrive in the meantime? [22:53] <cone-897> ffmpeg.git 03Michael Niedermayer 07master:911676c64374: avutil/avstring: add av_strnlen() [23:01] <jnvsor> micheal send me the token [23:01] <jnvsor> I've spent an hour bisecting the configure, now I have an exact list of options - remove any one of these and the bug dissapears... [23:01] <jnvsor> They seem totally unrelated [23:03] <beastd> jnvsor: sounds nasty [23:03] <beastd> jnvsor: still strange that the email didn't arrive... [23:04] <jnvsor> Yeah, I had it send 3, so 3 in a row didn't arrive [23:04] <jnvsor> If trac is working it must be hotmail [23:05] <beastd> worked for at least 3 people today (probably more I did not look that close) [23:11] <jnvsor> So basically libopencv + gnutls + shlibdir + pulse input = no sigint response [23:22] <BBB> ubitux: ok that works now, still need to do performance metrics [23:22] <BBB> ubitux: will try to finish that tonight, submit patches for merge and then work on parallelmode bug [23:23] <ubitux> OTOH packuswb doesn't work for me :p [23:23] <BBB> awh [23:23] <BBB> I'm sorry [23:23] <BBB> what did it do to you? [23:23] <ubitux> 13: /x $xmm4.uint128 = 0x0000fc2c0000fcf00000028400000688 [23:23] <ubitux> 12: /x $xmm5.uint128 = 0x0000fa390000f9390000fa1b0000fbdc [23:23] <ubitux> => 13: /x $xmm4.uint128 = 0x00000000000000000000000000ff00ff [23:23] <ubitux> not exactly what i expected [23:23] <ubitux> :D [23:24] <ubitux> (yes the input values look weird as well) [23:24] <ubitux> probably a shitload of bugs previously :p [23:27] <ubitux> (ah i used punpck*wd instead of the bw) [23:28] <ubitux> actually it works now :p [23:29] <BBB> lol [23:29] <BBB> well that's good [23:30] <ubitux> anyway, WIP still @ https://github.com/ubitux/FFmpeg/compare/vp9-lpf [23:31] <ubitux> i'm assuming it will be mostly straightforward now until 4w/2w code [23:31] <ubitux> it's fun [23:31] <ubitux> i really like that :D [23:35] <BBB> good :) [00:00] --- Mon Dec 23 2013

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ffmpeg.log.20131222
by burek 23 Dec '13

23 Dec '13

[02:12] <llogan> giby: "error oprning alias". is that a mov renamed as avi? [02:13] <giby> non, taken with my camera [02:13] <llogan> try "-use_absolute_path 1" as an input option [02:15] <giby> still the same bug :( [02:20] <llogan> giby: do you have a sample file? [02:21] <llogan> actually i should go...i recommend sending a message to the ffmpeg-user mailing list. [02:23] <giby> I have a file [02:23] <giby> I can open it in iphoto or iMovie [02:38] <SLAYBoz> how come when i use aac as audio encoder, on final video. i cannot see what audio-bitrate i used [04:22] <hendry> hi i havea a 1920x1080 12GB wedding video I want to make available via mp4 for Iphone users via HTML video http://s.natalian.org/2013-12-22/krfeature.txt [04:22] <hendry> I can't find a decent preset, can anyone help or recommend something? [05:56] <SLAYBoz> how come when i use aac as audio encoder, on final video. i cannot see what audio-bitrate i used [05:58] <relaxed> if you specify a bitrate why do you need to see it? [06:00] <SLAYBoz> relaxed i might want to see it later [06:01] <relaxed> mediainfo will give you the size- divide that by time [06:09] <SLAYBoz> it doesn't [06:09] <SLAYBoz> with mp3-audio it does [06:10] <SLAYBoz> with mp3-audio or ac3-audio it does [07:54] <Zeranoe> Can FFmpeg handle an flv file that has a different frame size mid way through? [09:04] <DeadSix27> can anyone explain what this means: Application provided invalid, non monotonically increasing dts to muxer in stream 0: 8030805 >= 8029920 [12:27] <Looney> trying to build ffmpeg with ssl for android using ndk [12:28] <Looney> following https://github.com/yixia/FFmpeg-Android/blob/master/FFmpeg-Android.sh [12:28] <Looney> but build failing with following message [12:28] <Looney> arm-linux-androideabi-gcc is unable to create an executable file. [12:28] <Looney> C compiler test failed [13:35] <vivienschilis> Hi, I was wondering if ffmpeg is deinterlacing automaticaly videos [13:35] <vivienschilis> I made a simple ffmpeg conversion without using yadif [13:35] <vivienschilis> and the result was progressive [13:36] <vivienschilis> (using the idet filter on the output file i could see that all frame were progressive) [13:37] <vivienschilis> michaelni: could you highlight me on this? [13:37] <JEEB> no, ffmpeg doesn't deint but depending on the encoder you might be encoding interlaced pictures as progressive :P [13:37] <JEEB> unless ffmpeg auto-inserts yadif in the middle, which I kind of don't believe to be what's happening [13:39] <vivienschilis> JEEB: thanks! it would lead to some async if one field is considered as one frame no? [13:41] <vivienschilis> is it safe to always apply yadif ? [13:44] <JEEB> vivienschilis, basically two fields interleaved and encoded as a frame would be my guess :P [13:45] <JEEB> you can check the encoded file and see if it has combing [13:45] <JEEB> (in places where the original interlaced content had them when interleaved) [13:45] <JEEB> and no, it's not always safe afaik, unless yadif nowadays only works with frames marked interlaced [13:53] <relaxed> that's an option [13:54] <relaxed> deint=1 "only deinterlace frames marked as interlaced" [13:57] <JEEB> ok then [14:08] <jnvsor> I don't really use IRC, have I got it working properly? [14:12] <jnvsor> When recording from a stream with lower framerate than the output stream ffmpeg skips frames and uses timestamps. Blender doesn't read timestamps. Is there any way to force frame duplication? [14:17] <Mavrik> jnvsor, use fps video filter [14:22] <jnvsor> Well it's doing something but while ffmpeg says frame=2999 when I stopped it, blender is looking at 1503 frames [14:31] <jnvsor> Blender probably just doesn't load files over 60fps properly. Thanks for the help! [14:32] <vivienschilis> JEEB: relaxed thanks for your help [14:53] <stockholm> hi [14:58] <stockholm> http://pastebin.com/wgNhtKHG [14:58] <stockholm> i used to have quality issues when reencoding the video [14:58] <stockholm> i want to play it on my dvd player [14:58] <stockholm> which does not speak mp4 [14:59] <stockholm> this current encoding however, is not accepted by my dvd player [14:59] <stockholm> furthermore, the sound seems to have been lost [14:59] <jnvsor> Well first off that's the libav ffmpeg (long story) not the ffmpeg ffmpeg [15:00] <stockholm> how do i fix both the sound and get my dvd player to accept the encoding of the video? [15:00] <stockholm> yes, i use debian [15:00] <jnvsor> As for audio, you have no audio codec in the command, add `-acodec copy` to the end [15:00] <stockholm> this is the one that comes with debian. [15:00] <stockholm> thanks [15:01] <stockholm> jnvsor: so why does my dvd player not acept the encoding? [15:01] <relaxed> you command is using the mpeg1 container [15:01] <relaxed> your* [15:01] <jnvsor> Not a clue - which formats does your dvd accept? [15:01] <relaxed> also, half the option you use don't pertain to mpeg2video [15:01] <stockholm> it does eat avi files i download from the net [15:02] <stockholm> i am not sure how to find out what my dvd accepts [15:02] <relaxed> ffmpeg -i /tmp/input.mp4 -vcodec mpeg4 -vtag XVID -qscale 3 -acodec libmp3lame -ab 160k output.avi [15:02] <jnvsor> A manual would probably have that (or google) [15:03] <stockholm> i went for a conservative encoding (mpeg-2, i hope, i dont really know a whole lot about video encoding) [15:04] <jnvsor> maybe libx264 would work too [15:04] <stockholm> jnvsor: i have been googling for some days, and encoded perhaps 20 test snipplets to get around the coars block videos i got before [15:04] <relaxed> stockholm: try the command I pasted ^^ [15:04] <stockholm> will do [15:04] <jnvsor> No I meant google your dvd player to see which formats it accepts :P [15:05] <stockholm> jnvsor: ah, right :-) [15:05] <stockholm> jnvsor: i really should [15:15] <stockholm> i keep running up and down the stairs here, since my dvd player is downstairs... an other roundtrip! [15:15] <stockholm> relaxed: it said: resolution not supported. [15:16] <stockholm> relaxed: can i just add a -s 720x... ? [15:17] <stockholm> trying that... [15:20] <tsjiller> 15:18:01 < rage\> fiska torsk [15:20] <tsjiller> 15:18:27 < rage\> NiteLite: hvordan får jeg hentet ut den infoen fra min crapple telefon? [15:20] <tsjiller> sorry [15:20] <tsjiller> uhm [15:24] <relaxed> ffmpeg -i /tmp/input.mp4 -vcodec mpeg4 -vf scale=720:-1 -vtag XVID -qscale 3 -acodec libmp3lame -ab 160k output.avi [15:24] <stockholm> with -s 640x480 it produced a playable video!!! [15:25] <stockholm> but yours is actually better, because it preserves the aspect ratio [15:26] <jnvsor> you can use something like `-filter:v 'scale=w=-1:h=480'` to scale it down if that's what you want [15:26] <stockholm> so when i re-encode a real big video (1.3 gig in mp4) down to that above, it becomes time consuming... what codec is the fastest to use? [15:26] <relaxed> is ffmpeg is too old for that command, and I beat you to it. [15:27] <stockholm> lol [15:27] <jnvsor> ah nuts, didn't read xD [15:27] <jnvsor> well he's not using ffmpeg [15:28] <jnvsor> he's using the Libav version [15:29] <relaxed> semantics [15:29] <jnvsor> I doubt they've updated the "Deprecated" thing since the fork however many years ago [15:30] <JEEB> yes, the cli interface in libav's ffmpeg binary is kind of withered, since the rewritten parts didn't get there. ffmpeg's ffmpeg has those changes merged in :) [15:30] <JEEB> so if using libav, avconv is preferred, if using ffmpeg ffmpeg is preferred [15:31] <stockholm> right [15:33] <JEEB> anyways, what relaxed has been saying will work on both :) [15:33] <JEEB> just that the newer binary in both projects will then accept the new settings etc. [15:34] <stockholm> http://pastebin.com/Smy6HTug [15:34] <stockholm> it didnt work, here is the error [15:35] <JEEB> [mpeg4 @ 0x11d0120] Invalid pixel aspect ratio 901/900, limit is 255/255 [15:35] <stockholm> yeah [15:36] <JEEB> try with avconv and see if it supports a setting called -setdar (if I recall correctly) [15:36] <JEEB> ah, it's a video filter [15:37] <JEEB> so after the scale you could add setdar or setsar [15:37] <JEEB> probably setsar is simpler since you want 1:1 SAR? [15:38] <stockholm> yes, preferably [15:39] <JEEB> setsar=1/1 [15:39] <JEEB> not sure how you added it after the scale [15:40] <jnvsor> if width/height isn't even the encoder will squeal, I use -filter:v `scale=w=trunc(in_w/2)*2:h=trunc(in_h/2)*2` to round it off [15:41] <stockholm> that looks advanced... [15:46] <stockholm> jnvsor: like this? http://pastebin.com/XXSkPSDW [15:46] <stockholm> breaks... [15:48] <jnvsor> you should put the filter after the vcodec [15:50] <stockholm> jnvsor: directly? or after the mpeg4? [15:51] <stockholm> neither works. [16:03] <stockholm> jnvsor: can you help me with the syntax? this fails: -vcodec mpeg4 -filter:v `scale=w=trunc(in_w/2)*2:h=trunc(in_h/2)*2` [16:03] <stockholm> jnvsor: with "Unable to find a suitable output format for 'XVID' [16:03] <stockholm> full paste? [16:04] <jnvsor> did you give it the extension xvid? [16:09] <stockholm> the full command line is [16:09] <stockholm> ....paste [16:10] <stockholm> http://pastebin.com/tSSLvkC4 [16:10] <stockholm> jnvsor: ^ [16:13] <jnvsor> Remove the xvid vtag part? [16:13] <jnvsor> I'm just guessing and semi-afk atm :/ [16:14] <stockholm> isnt that important? [16:14] <stockholm> its cascading... [16:14] <stockholm> now it complains about the next option/parameter pair: [16:14] <stockholm> Unable to find a suitable output format for '3' [16:15] <stockholm> which is from the qscale option [16:16] <JEEB> uhh [16:16] <JEEB> you have no output file name >_> [16:16] <stockholm> something is fundamentally broken with the format... [16:16] <JEEB> for xvid stuff I will guess you want out.avi at the end >_> [16:16] <stockholm> no, hte output file name is coming last [16:16] <JEEB> oh [16:17] <stockholm> look at the paste :-) [16:17] <JEEB> yes, I didn't notice it getting cut [16:17] <JEEB> the first line that is [16:17] <JEEB> (it spans multiple lines here when viewing it) [16:17] <stockholm> i am going through all the options and remove one by one and it all comes back with errors [16:17] <stockholm> yes, here too [16:21] <JEEB> vcodec is correct, although now that you're using avconv you could switch to the newer way of setting it - -c:v , not sure about the filtering part's correctness, but I will guess it's correct? -qscale should work, but you could use -q:v , -acodec could be switched to -c:a and -ab to -b:a . -threads auto shouldn't be needed, -t and -ss should be correct? [16:21] <stockholm> lets try that [16:24] <jnvsor> stockholm: try this? avconv -i input.mp4 -vcodec mpeg4 -acodec libmp3lame -t 4106 -ss 112 /tmp/output.avi [16:24] <jnvsor> if that doesn't work something wierd is going on [16:24] <stockholm> http://pastebin.com/EywgwRPT [16:25] <JEEB> yeah, and he can use the -c:v -c:a instead of -vcodec and -acodec [16:25] <stockholm> ok, trying your simplified version [16:25] <JEEB> uhh [16:25] <JEEB> -c:v `scale=w=trunc(in_w/2)*2:h=trunc(in_h/2)*2` <-wut [16:25] <JEEB> trying to set video encoder to something funky :P [16:25] <jnvsor> that's a scale filter not a codec ^^ [16:26] <JEEB> now try either with -vf or without the filtering at all [16:26] <jnvsor> try the simple version first, we can scale it once we know it's working :P [16:26] <stockholm> avconv -i /tmp/input.mp4 -c:v mpeg4 -c:a libmp3lame -t 4106 -ss 112 /tmp/output.avi <=works!!! [16:27] <jnvsor> gotcha [16:27] <jnvsor> do you want to scale it down? [16:27] <stockholm> i guess i need to. the resolution is humongours in the original :-) [16:28] <jnvsor> from 720p to 480p? [16:28] <stockholm> input is 1500xsomething [16:29] <jnvsor> avconv -i /tmp/input.mp4 -c:v mpeg4 -filter:v "scale=w=-1:h=480,scale=w=trunc(in_w/2)*2:h=trunc(in_h/2)*2" -c:a libmp3lame -t 4106 -ss 112 /tmp/output.avi [16:29] <stockholm> my dvd player does accept 720xsomething as input [16:29] <jnvsor> that will scale to 480 high then round it off so there are no uneven dimensions [16:29] <stockholm> nice [16:29] <stockholm> thanks [16:30] <stockholm> http://pastebin.com/xRPaBQLw [16:30] <stockholm> nix! [16:31] <stockholm> ah, because in_h? [16:31] <stockholm> no... [16:31] <jnvsor> the error is in the initial downscale [16:31] <jnvsor> very wierd, maybe you have a really old version, (Oh yeah you said you were on debian right? :P) [16:32] <jnvsor> avconv -i /tmp/input.mp4 -c:v mpeg4 -filter:v "scale=-1:480,scale=trunc(in_w/2)*2:trunc(in_h/2)*2" -c:a libmp3lame -t 4106 -ss 112 /tmp/output.avi [16:32] <jnvsor> try that [16:32] <jnvsor> should work on older versions [16:32] <stockholm> i really like debian... [16:32] <jnvsor> So do I, but I use sid :P [16:33] <stockholm> i used to when i was an active developer. [16:33] <stockholm> that works! [16:33] <stockholm> but it uses only 2 (of 4) cores [16:33] <stockholm> now i will try threads [16:33] <jnvsor> 2? Last I checked the default was 1 :O [16:34] <stockholm> no [16:34] <stockholm> top says 190% cpu [16:34] <stockholm> but i saw it use 390% already (aka 4) [16:35] <stockholm> but not now [16:35] <jnvsor> Oh right, default is "auto" [16:35] <stockholm> even if i say -threads 4 [16:35] <JEEB> there are official PPAs for libav for certain versions [16:35] <stockholm> PPAs? [16:35] <stockholm> can i eat those? [16:35] <JEEB> repositories [16:35] <stockholm> ah [16:35] <JEEB> https://launchpad.net/~motumedia [16:35] <jnvsor> Or just add an override for libav-utils to use the unstable repo :) [16:35] <JEEB> oh, it's debian? [16:35] <stockholm> yes [16:35] <JEEB> then yeah [16:36] <JEEB> testing/unstable [16:36] <jnvsor> Or just compile the real ffmpeg by hand [16:36] <JEEB> for libav and friends [16:36] <ubitux> if you're going to upgrade, why not upgrade to ffmpeg? [16:36] <JEEB> and yes, that's of course an alternative :) [16:36] <JEEB> there are static builds available too [16:36] <stockholm> yes, is there a repository for debian for that? [16:36] <ubitux> deb-multimedia probably [16:36] <JEEB> isn't deb-multimedia kind of old? [16:36] <stockholm> nix, that does not exist anymore [16:37] <JEEB> because it would have to keep API compat. [16:37] <stockholm> it was sold and does different stuff nowadays [16:37] <JEEB> no repo, but there are static binaries of current git HEAD available [16:37] <ubitux> http://www.deb-multimedia.org/dists/unstable/main/binary-amd64/package/ffmp… [16:37] <ubitux> 2.1.1 [16:37] <ubitux> sounds enough up-to-date [16:37] <JEEB> hmm [16:37] <stockholm> oh yes [16:38] <JEEB> or I guess they could update after debian got its stuff updated to libav9 [16:38] <JEEB> because older APIs became unneeded [16:38] <JEEB> yeah, unstable/sid [16:39] <jnvsor> http://stackoverflow.com/questions/512906/debian-how-can-i-pull-a-single-pa… [16:39] <jnvsor> You can add the unstable or testing repo then pin it to low priority and install the newer libav from usntable manually [17:44] <jnvsor> compiling ffmpeg doesn't give libx264 presets, what would be the equivillant of 'lossless_ultrafast'? Is libvpx better? (All the new presets are for libvpx) [17:46] <viric> jnvsor: libx264 has its own system of presets [17:48] <jnvsor> How do you mean? If I move the ffpreset file I get: "Preset lossless_ultrafast specified for stream 0:0, but could not be opened." [17:55] <BeWilled> will this count as 200 seconds? "00:00:300:00 ? [17:56] <BeWilled> :*200 [17:57] <ubitux> "200" should be fine [17:57] <ubitux> last sep needs to be a '.' iirc [18:25] <BeWilled> ubitux:hey I'm getting Invalid duration specification for ss: 00:00:200.00 [18:25] <ubitux> remove "00:00:" [18:25] <ubitux> and ".00" [18:29] <BeWilled> It worked, now it's time to paste them all together! thanks [18:33] <nse_> hi peopeeeee! [18:34] <nse_> i was looking how to "rip" audio from a video file I have! [18:34] <nse_> it's WAY difficult! someone could help me???? [18:35] <jnvsor> ffmpeg -i video.avi -c:a libmp3lame out.mp3 [18:35] <LithosLaptop> that transacodes [18:36] <nse_> jnvsor: but i need to select the audio channel [18:36] <nse_> jnvsor: my file has 2 ac3 audio channels [18:36] <jnvsor> in `ffmpeg -i video.avi` it should show you track IDs [18:36] <jnvsor> once you know the track ID run: [18:37] <jnvsor> ffmpeg -i video.avi -map {id} -c:a libmp3lame out.mp3 [18:37] <LithosLaptop> ffmpeg -i video.avi -map {id} -c:a copy out.ac3 [18:37] <jnvsor> Right, didn't notice ac3 :) [18:40] <nse_> jeez, you're just too good at it [18:40] <nse_> i managed to get it to work. THANKS!!! [18:40] <nse_> my command line: ffmpeg -i \[a4e\]Sentou_Yousei_Yukikaze_05\[h.264-ac3\].mkv -map 0:2 -c:a libmp3lame out.ac3 [18:41] <jnvsor> I'm surprised that worked, you put an mp3 stream into an ac3 container :O [18:41] <nse_> that's what i'm reading in man ffmpeg right now [18:42] <nse_> i think the "libmp3lame" is not well suited [18:42] <jnvsor> Yes, just copy the stream like lithos said [18:42] <jnvsor> replace "libmp3lame" with "copy" and it will work fine [18:43] <nse_> jnvsor: i wanted to "dump" the whole audio in a raw format maybe [18:43] <jnvsor> Yes, copy doesn't transcode, it simply copies the stream verbatim [18:43] <jnvsor> since it's ac3 audio an ac3 container seems good [18:43] <nse_> jnvsor: so i could encode it to mp3 using "lame"... is that a good way? [18:43] <jnvsor> You could, but litho's method copies the stream verbatim - no transcoding so no lost data [18:44] <jnvsor> it's as raw a copy of the original audio as you can get [18:44] <nse_> jnvsor: that's what i'm looking for =D [18:46] <nse_> aaah i get it. the format is still ac3. i just "copied" the original ac3 data from my video input strem. [18:46] <nse_> thanks for the tips. [19:20] <jnvsor> Hmm, does anyone else get no response from SIGINT when using a `-f pulse` input? [20:50] <jnvsor> How long does a trac verification email usually take to send? [20:53] <funyun> hi. can anyone help me with this error? http://pastebin.com/7F1Gimp2 [20:54] <jnvsor> get rid of the parentheses [20:54] <jnvsor> Oh right, escape the parentheses :P [20:54] <jnvsor> http://pastebin.com/hkEyr5PS [21:01] <funyun> jnvsor: that just puts me into something with this ">" [21:02] <funyun> i don't really know how to explain it [21:02] <funyun> but i have to do ctrl+c to get out of it [21:04] <brontosaurusrex> try -vf "pad=1280:720:(ow-iw)/2:(oh-ih)/2" [21:05] <brontosaurusrex> funyun, ^ [21:06] <funyun> brontosaurusrex: same result. i get > [21:09] <jnvsor> That's wierd, just pasted it into my terminal and I'm seeing it too [21:10] <jnvsor> Ah yes, moving the quote to the start of pad fixes it here [21:10] <jnvsor> http://pastebin.com/bzF1ZQdc [21:10] <funyun> jnvsor: brontosaurusrex's suggestion works for you? [21:12] <funyun> ah okay now it's working [21:12] <funyun> jnvsor, brontosaurusrex: thanks for your help. :) [21:19] <Sling> trying to convert some folders containing .VOB files into single mediafiles, preferably with the same size/quality as the 'original' [21:19] <Sling> ive used some different avconv options but I'm not really getting the proper settings I think, is there a standard way of doing this? [21:19] <Sling> ideally I would also want to automatically concatenate the .vob files [21:22] <jnvsor> Well I'm not sure on the specifics of the vob container/codecs but there's an article on the wiki which should help [21:22] <jnvsor> http://bit.ly/J7uV9B [21:23] <Sling> thanks, Ill search on from there [21:23] <Sling> can't imagine I'm the first to try this :) [21:23] <jnvsor> The article looks like it has a copy-paste solution :) [21:24] <Sling> yea the ffmpeg / concat=XXX one [21:48] <Sling> I'm using this now: ffmpeg -i concat:"VTS_01_1.VOB|VTS_01_2.VOB|VTS_01_3.VOB|VTS_01_4.VOB" -vcodec copy -acodec copy -map 0:0 -map 0:1 out.mkv [21:48] <Sling> but getting: [matroska @ 0x1f7f580] Can't write packet with unknown timestamp [21:49] <Sling> setting -fflags genpts had no effect [21:49] <jnvsor> That doesn't look like what was in the guide - did you try using a file with them in a list? [21:49] <Sling> it works without the copy stuff [21:50] <Sling> but I suppose I could try the list version [21:51] <Sling> that just goves me 'Unknown input format: 'concat'' [21:52] <jnvsor> what version of ffmpeg are you using? [21:52] <Sling> 0.8.9-6:0.8.9-0ubuntu0.13.10.1 [21:53] <viric> 2.1.1 is the latest released [21:55] <Sling> its telling me to use avconv though, which seemingly has the same options and same issues for me [21:56] <jnvsor> Then you're not using ffmpeg, avconv has an old copy of ffmpeg for compatability [21:57] <jnvsor> http://bit.ly/1kx0MPb [22:00] <Sling> yeah thats not working, same issue with the timestamp [22:02] <Sling> meh and with other settings it just mangles the whole thing [22:02] <Sling> artifacts and interlacing all over the place [22:02] <Sling> why is video so hard :) [22:02] <Sling> spend a whole evening just trying to fix one movie [22:04] <viric> Sling: -vf setpts=PTS-STARTPTS ? [22:04] <viric> I don't know if it's about pts [22:04] <Sling> I doubt it is, it doesnt fix the timestamp error [22:05] <Sling> the original is mpeg2video and ac3 audio [22:05] <Sling> apparently it cant just copy that to a mkv container [22:05] <viric> maybe you need the same for audio. asetpts [22:05] <Sling> hm nop [22:07] <Sling> mp4 output seems to work [22:07] <Sling> its complaining about pts not having a value but at least its producing something [22:08] <viric> -vf setpts=PTS-STARTPTS -af asetpts=PTS-STARTPTS together? [22:11] <Sling> ah this intermediate mp4 worked [22:11] <Sling> no encoding done so video/audio is all well [22:12] <Sling> step 1: avconv -i concat:"VTS_01_1.VOB|VTS_01_2.VOB|VTS_01_3.VOB|VTS_01_4.VOB" -c:v copy -c:a copy -map 0:0 -map 0:1 temp.mp4 [22:12] <Sling> step 2: avconv -i temp.mp4 -c copy output.mkv [22:18] <saste> we only support ffmpeg here [22:18] <Sling> yeah sorry, in this case I blame Ubuntu [22:21] <saste> Sling, why the double step? [22:21] <saste> mp4 -> mkv? [22:21] <saste> probably you should try with the concat demuxer, it will automatically fix timestamps (no transcoding involved) [22:34] <Sling> saste: I think this version of ffmpeg doesn't have that [22:34] <Sling> and yes the double step is to avoid the timestamp errors I was getting [22:34] <saste> Sling, i don't think it will fix anything [22:34] <saste> indeed it will probably make things worse [22:35] <Sling> well, the resulting file does what I want [22:35] <Sling> which is have the same quality as the original and play in my player :) [22:35] <saste> unless you're overusing muxer glitches [23:30] <brontosaurusrex> Sling, you could use mkvtoolnix (I don't know if ac3 or dts will work fine with mp4) [23:30] <brontosaurusrex> if mkv is good enough for you that is [23:31] <JEEBsv> ac3 is officially spec'd in MP4 methinks [23:32] <JEEBsv> DTS has an unofficial spec by the DTS corp [23:32] <brontosaurusrex> ok then [23:40] <klaxa> either i'm stupid now or the latest ffmpeg doesn't parse arguments correctly: https://gist.github.com/klaxa/8089277 [23:40] <klaxa> >Applying option vf (set video filters) with argument subtitles=filename=[WhyNot] SaintYoung Men The Movie [BD 720p AAC][12DEEE77].ass. [23:40] <klaxa> >... [23:40] <klaxa> >[Parsed_subtitles_0 @ 0x33e2440] Setting 'filename' to value '' [23:43] <JEEBsv> klaxa: if you need hardsubbing I'd poke you towards mpv's encoding thingy. #mpv-player [23:44] <JEEBsv> uses libavformat/libavcodec behind the scenes and supports segment linking as well [23:44] <klaxa> alright, thanks [23:44] <klaxa> oh does this mean i have to compile mpv on my headless machine? :S [23:45] <JEEBsv> should be doable since you don't need any of the fancy video renderers and such [23:45] <klaxa> hmm true [23:45] <klaxa> i'll do that first, i guess i can use libx264 with it? [23:45] <JEEBsv> naturally [23:46] <klaxa> everything else would be stuipd [00:00] --- Mon Dec 23 2013

1 0

ffmpeg-devel.log.20131221
by burek 22 Dec '13

22 Dec '13

[00:05] <cone-64> ffmpeg.git 03Paul B Mahol 07master:e518cb863edc: jvdec: Fix memory leak of jv->frames [00:05] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:b9f33ad32077: Merge commit 'e518cb863edc931888ccca6cad86f73ca7430cef' [00:13] <cone-64> ffmpeg.git 03Luca Barbato 07master:15739a9bd19a: jvdec: Do not feed the decoder with known wrong data [00:13] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:0c16f6449bb5: Merge commit '15739a9bd19a7d47ad8afb25348c684a3bdd6ef2' [00:20] <cone-64> ffmpeg.git 03Luca Barbato 07master:f412b2c9f3a7: flv: Use the correct type to hold the file offset [00:20] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:6fc9f5e0ff77: Merge commit 'f412b2c9f3a7add0ab8021262ec4bad249347e30' [00:31] <cone-64> ffmpeg.git 03Gildas Cocherel 07master:2a41826bea38: lavc: add HEVC profiles names [00:31] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:d6b7ea06f194: Merge commit '2a41826bea3833895dc06939831b7f35ca1f597e' [00:33] <cone-64> ffmpeg.git 03Gildas Cocherel 07master:321cb8b048c7: hevc: store profile and level in AVCodecContext [00:33] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:2fb9e77ac2da: Merge commit '321cb8b048c7a8c0d2950f9278f36d912a886fa0' [00:51] <cone-64> ffmpeg.git 03Guillaume Martres 07master:67bbaed5c498: hevc: don't check for errors in PTL code [00:51] <cone-64> ffmpeg.git 03Gildas Cocherel 07master:2d18aaa14bc2: hevc: refactor Profile Tier Level [00:51] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:e00281b6f507: Merge commit '2d18aaa14bc2233b0647d2c5abb72e0bc828c2a8' [00:53] <cone-64> ffmpeg.git 03Gildas Cocherel 07master:259cddb13798: hevc_ps: fix indentation [00:54] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:3cd36b07dd67: Merge commit '259cddb137988a25607f091c2414158ff81ca7d6' [01:02] <cone-64> ffmpeg.git 03Guillaume Martres 07master:78d0b7f36e39: hevc: warn when an unknown profile is used [01:02] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:9567162237d2: Merge commit '78d0b7f36e39bb4745613e312024ddd508e2f0fb' [01:20] <cone-64> ffmpeg.git 03Mickaël Raulet 07master:5c3fa74b94fd: hevc: support luma bit depth != chroma bit depth for PCM coding units [01:20] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:88839718b58c: Merge commit '5c3fa74b94fdedfd63b606f36920d7fbbc069b72' [01:29] <cone-64> ffmpeg.git 03Martin Storsjö 07master:d307e408d4a9: arm: Don't clobber callee saved registers in scalarproduct [01:29] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:70d6ce783cc5: Merge remote-tracking branch 'qatar/master' [02:22] <cone-64> ffmpeg.git 03Derek Buitenhuis 07release/1.1:1a5a6ac01b0a: pthread: Fix deadlock during thread initialization [02:22] <cone-64> ffmpeg.git 03Michael Niedermayer 07release/1.1:5bd291e265b9: avfilter/ff_insert_pad: fix order of operations [02:22] <cone-64> ffmpeg.git 03Michael Niedermayer 07release/1.1:b589b8a7042f: Merge commit '1a5a6ac01b0ad2cf3d2128372ea41f3c1cfc2d3f' into release/1.1 [02:41] <cone-64> ffmpeg.git 03Ben Jackson 07release/1.1:311583e77982: pthread: Avoid spurious wakeups [02:41] <cone-64> ffmpeg.git 03Diego Biurrun 07release/1.1:a3f8c6a42759: x86: ac3dsp: Remove 3dnow version of ff_ac3_extract_exponents [02:41] <cone-64> ffmpeg.git 03Anton Khirnov 07release/1.1:56eded8bc7bc: mpeg4videodec: split initializing static tables into a separate function [02:41] <cone-64> ffmpeg.git 03Michael Niedermayer 07release/1.1:2d7f139155dd: Merge commit '56eded8bc7bccdf14245bae3a45b0fecf9d9d122' into release/1.1 [02:49] <cone-64> ffmpeg.git 03Anton Khirnov 07release/1.1:bd405475ceb3: mpeg4video_parser: init mpeg4 static tables. [02:49] <cone-64> ffmpeg.git 03Anton Khirnov 07release/1.1:718a2ddcb898: h264/mpegvideo: do not provide pixel formats for hwaccels that are not compiled in [02:49] <cone-64> ffmpeg.git 03Reinhard Tartler 07release/1.1:3f7d89034bfe: alsdec: check block length [02:49] <cone-64> ffmpeg.git 03Martin Storsjö 07release/1.1:bdb975ab699a: arm: Don't clobber callee saved registers in scalarproduct [02:49] <cone-64> ffmpeg.git 03Michael Niedermayer 07release/1.1:50ff83e3afa7: Merge remote-tracking branch 'qatar/release/9' into release/1.1 [07:44] <clever> Too many buffered pts [07:44] <clever> what typicaly causes an error like this? [10:01] <cone-45> ffmpeg.git 03Martin Storsjö 07release/0.10:594b8436084c: arm: Don't clobber callee saved registers in scalarproduct [10:01] <cone-45> ffmpeg.git 03Martin Storsjö 07release/0.11:69f724a538ef: arm: Don't clobber callee saved registers in scalarproduct [10:01] <cone-45> ffmpeg.git 03Martin Storsjö 07release/0.6:a94d12b69027: arm: Don't clobber callee saved registers in scalarproduct [10:01] <cone-45> ffmpeg.git 03Martin Storsjö 07release/0.7:cce6bdc75839: arm: Don't clobber callee saved registers in scalarproduct [10:01] <cone-45> ffmpeg.git 03Martin Storsjö 07release/0.8:8925c44db1a2: arm: Don't clobber callee saved registers in scalarproduct [10:01] <cone-45> ffmpeg.git 03Martin Storsjö 07release/0.9:320df1c7c0fe: arm: Don't clobber callee saved registers in scalarproduct [10:01] <cone-45> ffmpeg.git 03Martin Storsjö 07release/1.0:a4c804d0a2ca: arm: Don't clobber callee saved registers in scalarproduct [10:01] <cone-45> ffmpeg.git 03Martin Storsjö 07release/1.2:3e65caf5bc3e: arm: Don't clobber callee saved registers in scalarproduct [10:02] <cone-45> ffmpeg.git 03Martin Storsjö 07release/2.0:62f05d6309b8: arm: Don't clobber callee saved registers in scalarproduct [10:02] <cone-45> ffmpeg.git 03Martin Storsjö 07release/2.1:0d32483a11b0: arm: Don't clobber callee saved registers in scalarproduct [11:41] <michaelni> clever, where is that error comming from ? [11:44] <michaelni> ubitux/saste, there are 4 patches for doc/examples on ffmpeg-dev, do you have time/want to review them or should i ? [11:44] <michaelni> "83663 N 1214 19:46 Ilya Basin (1.7K) [FFmpeg-devel] [PATCH] fix 2 memleaks in doc/examples/muxing.c (and replies to it)" [11:54] <wm4> clever: is that a mplayer error? [12:00] <cone-45> ffmpeg.git 03Peter Ross 07master:8baaa924bd42: wtvdec: when testing for truncated files, actually check the .wtv file size [12:12] <ubitux> michaelni: i almost never touched that example :p [12:13] <michaelni> ok, can you help "83792 1219 17:18 JULIAN GARDNER (3.5K) Re: [FFmpeg-devel] [FFmpeg-user] BUG in use of extradata and extradata_size with dvb subtitles and teletext" ? [12:13] <michaelni> iam not completely sure what its about [12:19] <michaelni> ubitux, ill look at the examples [12:19] <michaelni> example patches [12:21] <ubitux> yes i planed to look at this one [12:23] <michaelni> ok, thx [12:24] <wm4> btw. are the decoding example fine now wrt. to the frame_alloc shit? [12:24] <wm4> since elenril managed to unify these [12:25] <ubitux> the regression is fixed? [12:33] <cone-45> ffmpeg.git 03Ilya Basin 07master:b6714fa077d9: examples/muxing: fix av_frame_free() not called when got_packet is false [12:33] <cone-45> ffmpeg.git 03Ilya Basin 07master:7d1d0b3ecf98: examples/muxing: use S16 sample_fmt for resample src regardless of codec sample_fmt [12:33] <cone-45> ffmpeg.git 03Ilya Basin 07master:d1b8e01ef1fe: examples/muxing: fix memleaks in resampler [13:48] <cone-45> ffmpeg.git 03Michael Niedermayer 07master:263105deebbd: avcodec/vqavideo: check destination size on all exit pathes [13:48] <cone-45> ffmpeg.git 03Michael Niedermayer 07master:03acb035d129: avcodec/vqavideo: clear uninitialized parts of the decode buffer [14:10] <cone-45> ffmpeg.git 03Michael Niedermayer 07master:587e3c3426be: avformat/electronicarts: use 64bit variable for avio_tell() result [14:10] <cone-45> ffmpeg.git 03Michael Niedermayer 07master:a07d4bc80a7b: avformat/gxfenc: dont use a int intermediate to store in a unsigned int table [14:13] <saste> is there any use for softpulldown? [14:19] <saste> or in other words, isn't telecine already a valid substitute for it? [14:34] <saste> ubitux: ^^? [14:53] <ubitux> saste: dunno? :P [17:21] <cone-45> ffmpeg.git 03Michael Niedermayer 07master:55328d5f7246: avformat/electronicarts: detect endianness by comparing sizes instead of using an arbitrary threshold [19:25] <cone-45> ffmpeg.git 03Michael Niedermayer 07master:ea1e630c47e7: vcodec/rv34: use av_mallocz() to allocate tables [19:44] <clever> wm4: yeah, i found the error in mplayer source [19:44] <clever> wm4: i just cant figure out why its happening for this pix fmt [19:45] <clever> i'm trying to modify my AVHWAccel to output AV_PIX_FMT_RGB565BE [19:45] <clever> but it refuses to open or query the vo module [19:46] <wm4> it normally means that you're putting in more packets than you get frames out (for a certain codec delay) [19:47] <clever> and it not even trying to open a vo module, so no frames are going to the output [19:49] <clever> wm4: http://privatepaste.com/9cfb1bb465 it also ignores -frames 3 [19:50] <wm4> I wouldn't use mplayer to test this [19:50] <wm4> because mplayer is full of crap [19:52] <clever> i can pull the code back out and try it on ffplay, but do you know how well ffmpeg/ffplay handle rgb565 internally? [19:52] <clever> trying to avoid pixel conversion, so i just set the hw decoder to create 565 directly [19:52] <wm4> ffplay will just convert it to yuv420p I think [19:53] <nevcairiel> 565 is a rather terrible format, why do you want it anyway [19:53] <clever> there are no working vo modules that accept yuv420 [19:53] <clever> so if i pick yuv420, it has to convert in software [19:53] <clever> and most of the working modules are converting yuv420 into 565 in software [19:54] <nevcairiel> the decoder has to work in yuv420p internally, so there is a conversion somewhere, are you certain its not also done in software by the hw driver? :p [19:55] <clever> ive yet to get ffmpeg to play with rgb565, so i dont know [19:55] <clever> but acording to the docs, it should be all in hardware [19:56] <clever> i would stick to yuv420, but the compositor is only giving out green, it ignores the u/v channels completely [19:56] <nevcairiel> sounds like you might've send the information wrong [19:57] <clever> i was getting various ghosting artifacts when i gave ffmpeg bad yuv420, the colors where all over the place [19:57] <clever> but this time, i'm not even getting that, just green [19:58] <clever> http://gallery.earthtools.ca/index.py/paste/DSCF0558.JPG [19:59] <ubitux> sounds like zero-ed luma plane [19:59] <ubitux> oh well, zero-ed chroma [19:59] <clever> i tried doing a memset on the chrome planes with 128, but it does the same thing [20:00] <ubitux> can we see the code? [20:01] <clever> one sec [20:01] <clever> http://privatepaste.com/3c802f9886 is the vo module [20:01] <clever> createFrame allocates a buffer in the cpu and sets up the 3 plane pointers [20:02] <clever> updateFrame copies it to the gpu [20:02] <clever> dpFlip changes which resource is being displayed [20:02] <ubitux> ah, mplayer stuff [20:02] Action: ubitux goes away [20:03] <clever> ubitux: how easy is it to add a vo module to ffplay? [20:03] <ubitux> ffplay is sdl only so far [20:03] <clever> now you see why i went with mplayer, lol [20:04] <clever> the sdl libs are software only, -vo x11 works better [20:04] <ubitux> there is an accelerated video output driver in the rpi? [20:05] <clever> thats what i just made [20:05] <ubitux> just like vdpau, there is an api for hw decoding, and an api for hw rendering? [20:05] <clever> the gpu uses dispmanx to composite up to 20 layers into the final video output [20:05] <clever> for each scan line, it will fetch the relevant pixel data and convert it on the fly, from all layers that are in that line [20:06] <clever> with hw scalling, alpha, and pixel conversion [20:08] <clever> i wrote the hw decode and hw rendering as seperate modules, both working with yuv420, so it can easily work with other codecs [22:04] <clever> ubitux: ok, i finaly got SOME color, but its totaly scrambled [22:04] <clever> the source size is specified with width << 16, height << 16, and if i remove the shifting, i get various colored lines [22:15] <ubitux> BBB: WIP, probably not as side tracked as the first attempt: https://github.com/ubitux/FFmpeg/compare/vp9-lpf [22:16] <ubitux> only the fm/flat8in/flat8out check are done& now starts the "fun" :s [22:16] <ubitux> of course, it might not even actually work [22:17] <ubitux> i'm pretty sure it doesn't but well :) [22:26] <BBB> simd usually doesn't work on a first attempt [22:26] <BBB> ubitux: nice work! [22:27] <BBB> I have a working 8x8 sub-16x16 idct, and a non-working 4x4 sub-16x16 [22:27] <ubitux> :) [22:27] <BBB> or maybe we should call it 16x16 sub-8x8 and 16x16 sub-4x4 [22:27] <BBB> I'm debugging the sub4x4 but have had some busy work days lately [22:28] <ubitux> same, just started working on lpf only today [22:52] <BBB> well that was easy, 4x4 works also now [22:52] <BBB> silly typo [22:52] <BBB> (as is always the case) [22:52] <BBB> 2x2 still to be done, also no speed measurements yet [22:52] <BBB> then on to 32x32 again, I can feel it hating me [22:55] <ubitux> :) [23:37] <cone-372> ffmpeg.git 03Michael Niedermayer 07master:b797a00f6346: avformat/rmdec: zero string destination before use [00:00] --- Sun Dec 22 2013

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ffmpeg.log.20131221
by burek 22 Dec '13

22 Dec '13

[01:53] <msmithng> so I've got this high end encoder that I would like to supply mpeg2ts to as it's input... [01:54] <msmithng> I've been trying to use ffmpeg to push to localhost and pipe out via nc on a listening port, but I'm having no luck [01:57] <msmithng> the hardware encoder I'm testing can only use ip/port as it's input that it's scanning& further and I'm not 100% on this, but I think it's also looking specifically service_id, pmt_start_pid and start_pid& which obviously I'm not setting [01:58] <msmithng> crap, that might be it [01:58] <msmithng> anyone have any alternatives to dumping this to nc on localhost? [01:59] <msmithng> it looks like it's just flooding the socket and exiting [01:59] <msmithng> i.e. ignoring -re [02:00] <msmithng> this is the command I've cobbled together at the moment. [02:00] <msmithng> ffmpeg -re -i sintel-2048-surround.mp4 -c:v libx264 -profile:v main -level 3.1 -r 30 -g 30 -keyint_min 30 -minrate:v 1M -minrate:v 1M -bufsize:v 1.4M -c:a libfaac -ab 96000 -ar 48000 -ac 2 -vbsf h264_mp4toannexb -f mpegts - | nc -l 1234 [02:05] <relaxed> msmithng: look at ffmpeg -protocols, it supports outputting to tcp/udp [02:06] <relaxed> do you need -vbsf h264_mp4toannexb there? [03:33] <msmithng> dang, client dropped [03:34] <msmithng> anyone answer the above? :P [03:44] <lehrblogger> msmithng: relaxed said "look at ffmpeg -protocols, it supports outputting to tcp/udp [03:44] <lehrblogger> do you need -vbsf h264_mp4toannexb there? [03:45] <lehrblogger> oh, but maybe after you dropped [03:45] <lehrblogger> err before [05:23] <msmithng> yeah I must have dropped before any of that was said [05:25] <msmithng> lehrblogger: I guess the question is what is the appropriate way to output mpeg2 ts for another encoder that expects to be able to pull mpeg2ts over ip? [05:25] <msmithng> we're missing all the head end equipment that a normal broadcast bay would have so I'm making most of this up as I go along [05:26] <lehrblogger> msmithng: oh, i have no idea at all, i was just pasting what you might have missed from relaxed [05:26] <msmithng> :P [05:27] <msmithng> Not a problem what-so-ever [05:27] <lehrblogger> :) [05:28] <msmithng> I'll keep digging and if I come up with something clever or more likely, when I get stuck, I'll be back. [05:30] <lehrblogger> msmithng: rereading your earlier stuff, i only partially understand, but have you looked at http://www.scara.com/~schirmer/o/mplex13818/ ? [05:31] <msmithng> I have not& let me peek at that [05:32] <msmithng> "it does not even care for what data it shovels: just bytes" [05:32] <msmithng> lulz& sounds like I could maybe use this [05:32] <msmithng> thanks [05:33] <lehrblogger> yeah, no problem [05:33] <lehrblogger> just out of curiosity, are you using any particular devices for whatever it is you're doing? [05:34] <msmithng> the system I'm testing is an Envivio 4Caster [05:35] <msmithng> nothing else in particular at the moment [05:40] <lehrblogger> ok, thanks [07:25] <SirCmpwn> is there a way I can convince ffmpeg to completely ignore file extensions and to rely entirely on magic numbers and such [07:26] <sacarasc> -f [07:27] <SirCmpwn> I also don't know the input file format [07:27] <SirCmpwn> this is running out of a script [07:27] <SirCmpwn> basically, if I give it a gif with a .jpg extension, it pukes [07:33] <liam-kelly> Okay, so I am having some issues with ffmpeg and ffserver, when I run ffmpeg it works fine for a few seconds then freezes and I have to use ^c to get back to a prompt. http://pastebin.com/b63FCPKT Any ideas? [07:41] <maxiaojun> wonder why ffmpeg build requires yasm 1.2.0? [07:42] <sacarasc> To compile the assembly in the source. [07:43] <maxiaojun> i mean 1.1.0 doesn't work? [09:56] <maxiaojun> http://trac.ffmpeg.org/wiki/UbuntuCompilationGuide [09:56] <maxiaojun> libopus released a newer version 1.1 [14:36] <pron> is there any reason why ffmpeg when encoding should select second audio stream instead of first one? [14:36] <pron> its ts -> mp4 [14:36] <pron> ffmpeg 8.2 [14:37] <pron> i mean something liek Stream #0.2 -> #0.1 without any map options provided [14:39] <saste> pron, what is 8.2? or is it 0.8.2? [14:39] <saste> anyway it's way too old [14:39] <pron> yeh 0.8.2 [14:39] <saste> and ffmpeg select the "best" stream according to heuristics, usually the stream with the biggest bitrate or samplerate [14:40] <saste> *selects [14:40] <pron> humm they are kinda indentical at least from what ffmpeg outputs [14:41] <pron> ookay [14:48] <pron> saste http://pastebin.com/bdYZPzWU [14:49] <saste> pron, and where's the script output? [14:49] <pron> thats abit complicated , workin on it :P [14:50] <pron> there are tons of unrelated stuff [14:50] <pron> ^^ [14:50] <pron> in log file [14:54] <pron> saste http://pastebin.com/0F3LkmaD [14:57] <ubitux> ffmpeg 0.8? can't you pick something even older? [14:57] <pron> older might be a problem [14:57] <pron> :D [14:57] <ubitux> that was ironic ofc [14:57] <pron> ah [14:57] <pron> =) [15:07] <pron> soo [15:07] <pron> ubitux any ideas why ffmpeg took 2nd audio stream ?:} [15:07] <pron> or anyone else [15:09] <ubitux> use -map 0:a:0 to pick the first one [15:09] <ubitux> ffmpeg can randomly pick any stream, based on different heuristics [15:09] <ubitux> also, as said several times, 0.8 is way too old [15:15] <pron> ima bit afraid of upgrading it [15:15] <pron> dont want to break the stuff that uses it [15:16] <pron> well anyways [15:16] <pron> thx for infos and tips [15:16] <ubitux> you don't need to install it to use it [15:17] <ubitux> you git clone and build it, then run ./ffmpeg from the source dir [15:21] <pron> is there a way to tell ffmpeg to encode _all_ audio streams with same parameters [15:23] <ubitux> pron: -map 0:a maybe [15:41] <braincracker> hi [15:42] <braincracker> new ffmpeg is working? [15:42] <braincracker> or what is workign now? [15:42] <bencc2> in audio conference, is it bettero to mix several channels together or send separate channels? [15:42] <bencc2> what will give better quality? [15:42] <bencc2> I don't care about bandwidth [15:44] <braincracker> lossless monkjeyaudio is best [15:45] <braincracker> your $1000 studio mic in a studio will create some quality recording [15:50] <braincracker> how one simply warp audio speed? i'd like to slow down mp3's [15:54] <ubitux> braincracker: -af atempo [15:56] <braincracker> ty [15:56] <braincracker> will that simply warp? i do not want to convert pitch [18:03] <liam-k> Hello, I am having some issues with ffmpeg freezeing when using ffserver [18:04] <braincracker> that is nice [18:04] <braincracker> how cold is it? [18:06] <liam-k> cold enough that I have to use control c to regain control... [18:07] <liam-k> I detailed my problem here http://stackoverflow.com/questions/20716688/av-interleaved-write-frame-imme… and here http://pastebin.com/b63FCPKT [23:49] <dannyzb> How much of a performance hit do I get from encoding with "Main" profile? ( encoding time hit * ) [23:49] <dannyzb> compared to baseline [23:51] <giby> Hi, I need some assistance with a video conversion [23:52] <giby> here is a paste bin http://pastebin.com/gRSqniv7 [00:00] --- Sun Dec 22 2013

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