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January 2013

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ffmpeg-devel.log.20130125
by burek 26 Jan '13

26 Jan '13

[00:08] <cone-871> ffmpeg.git 03Stefano Sabatini 07master:40cb682ca0b9: doc/faq: mention concat protocol documentation in the protocol concatenation entry [00:08] <cone-871> ffmpeg.git 03Stefano Sabatini 07master:2756c3091a9d: doc/faq: fill missing word in the concat protocol entry [00:08] <cone-871> ffmpeg.git 03Stefano Sabatini 07master:069d15645413: lavf/img2dec: fix option help fields [00:08] <cone-871> ffmpeg.git 03Stefano Sabatini 07master:1ec3324f00e7: lavf/img2enc: extend current options documentation [00:49] <cone-871> ffmpeg.git 03Stefano Sabatini 07master:c4738c319610: ffmpeg: fix typo in open_files() message [01:27] <Compn> Cisco Exits the Consumer Market, Sells Linksys To Belkin [01:28] <Compn> not that it makes much difference in quality :P [01:38] <gnafu> I stopped caring about Linksys when they were bought by Cisco. I've always hated Belkin, for whatever reason, so this is almost worse to me. [01:39] <gnafu> "I was quit [when they were bought by Cisco]. I'm twice as quit now [that they were bought by Belkin]." [01:43] <llogan> i still use an old WRT54G*. [01:47] <Compn> those plastic blue network hubs/routers are terrible [01:47] <Compn> metal ones arent bad :) [03:06] <cone-871> ffmpeg.git 03Carl Eugen Hoyos 07master:a0d1440476ce: Fix compilation with --disable-everything on x86_32. [03:17] <ulatekh> Anyone here? I have a question & no one is responding on #ffmpeg. [03:28] <Darkarnium> Compn: the WRT54GLs were good though :( [03:29] <Darkarnium> As were the original WRT54GS, etc. [03:29] <Darkarnium> Doesn't matter as much now as single board PCs are so cheap, but they were great for hardware hacks previously [03:44] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:286930d302fd: gifdec: check that w,h is not zero [03:44] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:e9d443cf0850: eacmv: Free frames on resolution changes [05:43] <Mista_D> just a quick question, can ffmpeg route libavfilter(idet) output to csv/json format? [05:50] <Mista_D> Or can ffprobe use detection filters as `idet`? [06:24] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:ab6c9332bfa1: vqavideo: check chunk sizes before reading chunks [07:00] <funman> michaelni: do you keep LIBAVUTIL_VERSION_MICRO >= 100 as for libavcodec? [07:14] Action: funman assumed so [07:27] <michaelni> funman, yes but iam sleeping ATM [11:05] <funman> michaelni: alright, good night [11:09] <durandal_1707> michaelni: how 286930d302fd34cf can be true? [11:25] <durandal_1707> lol, luzero addding x264-params .... [11:36] <jeje> hello ;-) [11:40] <jeje> I have a question about avcodec_decodevideo2 function [11:40] <jeje> in fact, I use FFMPEG to decompress video streaming from IP camera in H264 [11:41] <av500> sounds reasonable [11:41] <jeje> the video source is in YUV420 and I use swscale to render directly in a DDRaw surface in RGB32 [11:42] <jeje> I have some problem with some video resolution [11:43] <jeje> when I can divide the width resolution with 32, it work well but when the video resolution can't be divide by 32, I have to add the modulo part [11:43] <jeje> in fact, i do [11:43] <jeje> av_image_fill_linesizes(m_lpFrame->linesize, PIX_FMT_YUV420P,dwWidth+(dwWidth%32)+32); [11:44] <jeje> it work if I do this, but I can't explain me why... [11:44] <durandal_1707> buffers must be padded it is in documentation [11:48] <jeje> you tell me I have to pad the input buffer or the output buffer? [11:49] <jeje> I'm sorry by I don't find this part in the documentation [11:50] <durandal_1707> jeje: i told you already, now please ask further questions in correct channel #ffmpeg and not here [11:51] <jeje> ok sorry [12:56] <burek> I've found this link by accident http://www.evermeet.cx/public/ffmpeg/snapshots/ [12:56] <burek> it looks like the guy has set up static FFmpeg binaries for Mac OS X Intel 64bit [12:57] <burek> if someone is using MacOS, we could check if it's ok and add it to the downloads page? [12:57] <burek> oh.. it's already there.. just ignore me :S [13:01] <durandal_1707> michaelni: looping over pointer in for is faster than using array[i] [13:29] <burek> am i doing something wrong or this version of ffmpeg does not exist: N-48289-g9d8db19-tessus [13:29] <burek> http://git.videolan.org/?p=ffmpeg.git;a=commit;h=9d8db19 [13:51] <cone-871> ffmpeg.git 03Janne Grunau 07master:8e148b874220: arm: h264qpel: use neon h264 qpel functions only if supported [13:51] <cone-871> ffmpeg.git 03Janne Grunau 07master:6bdb841b46d1: h264: copy h264qpel dsp context to slice thread copies [13:51] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:91ae9bc51e0f: Merge commit '6bdb841b46d170d58488deaed720729b79223b1d' [13:56] <michaelni> durandal_1707, the amd optimization manual says the opposit IIRC [13:56] <michaelni> also iam not sure to what code you refer to [13:57] <durandal_1707> the one in gifdec [13:57] <durandal_1707> reading pallete or any other for loop [13:59] <cone-871> ffmpeg.git 03Paul B Mahol 07master:f1412c799732: lavc/gifdec: use memcpy() [13:59] <cone-871> ffmpeg.git 03Paul B Mahol 07master:aaebdce3d907: lavc/gifdec: simplify "!= 0" checks [13:59] <cone-871> ffmpeg.git 03Paul B Mahol 07master:285128eedf9e: lavc/gifdec: do not return nonzero *got_frame if frame is not passed [13:59] <cone-871> ffmpeg.git 03Paul B Mahol 07master:51c7af9d3237: lavc/gifdec: move idx_line allocation out of gif_read_header1() [13:59] <cone-871> ffmpeg.git 03Paul B Mahol 07master:7003d650b075: lavc/gifdec: remove obsolete check [14:08] <cone-871> ffmpeg.git 03Paul B Mahol 07master:d07b0d992736: swscale: check flags instead of nb_components to find if pixel format have alpha [14:22] <cone-871> ffmpeg.git 03Janne Grunau 07master:c5c2060cf597: x86: h264qpel: add cpu flag checks for init function [14:22] <cone-871> ffmpeg.git 03Diego Biurrun 07master:2c10e2a2f624: build: Make the H.264 parser select h264qpel [14:22] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:e9125dd55634: Merge commit '2c10e2a2f62477efaef5b641974594f7df4ca339' [14:28] <cone-871> ffmpeg.git 03Luca Barbato 07master:4333df63551c: avstring: K&R formatting cosmetics [14:28] <cone-871> ffmpeg.git 03Diego Biurrun 07master:33552a5f7b6e: arm: Add mathops.h to ARCH_HEADERS list [14:28] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:b2d0c5bd13ab: Merge commit '33552a5f7b6ec7057516f487b1a902331f8c353e' [14:45] <cone-871> ffmpeg.git 03Vittorio Giovara 07master:a84fb6e06f7f: h264: Allow discarding the cropping information from SPS [14:45] <cone-871> ffmpeg.git 03Vladimir Pantelic 07master:b85a5e87af42: lavu: Add av_strnstr() [14:45] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:25be63005f0b: Merge commit 'b85a5e87af4254b80913fe33591d96361f30832b' [14:46] <durandal_1707> lol, now changelog will be full of libc cloned functions.... [14:47] <wm4> av_nih [14:49] <ubitux> hellow :) [14:50] <durandal_1707> wtf, what is wrong with people: http://linuxaudio.org/mailarchive/lau/2013/1/12/195808 [14:50] <ubitux> heh, luca adding API info into Changelog again? [14:50] <durandal_1707> no av500 on behalf of someone i guess [14:51] <av500> ...It seems madness to me that cdrecord would be considered obsolete. [14:52] <av500> durandal_1707: ??? [14:52] <av500> durandal_1707: do you see strnstr() in libc? [14:52] <av500> I dont [14:53] <durandal_1707> yup, it is here [14:53] <av500> BSD [14:53] <av500> yes [14:53] <durandal_1707> poor you :/ [14:53] <av500> not poor at all [14:55] <av500> and as usual, ffmpeg is free not to merge stuff :) [14:56] <wm4> not merging significant changes == suicide [14:57] <av500> so far av_strnstr() is unused :) [14:58] <durandal_1707> i only loled on changelog.. [14:59] <durandal_1707> wm4: what significant changes? move/rename/K&R formating/remove... [15:06] <michaelni> ubitux, hellow :) [15:08] <durandal_1707> and they are still struggling with "partial dirac hack" [15:09] <wm4> what hack? [15:09] <cone-871> ffmpeg.git 03Vladimir Pantelic 07master:0b55b16abc15: avfilter: allow setpts filter to use wallclock time for calculations [15:09] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:fc2922836bc2: Merge remote-tracking branch 'qatar/master' [15:10] <durandal_1707> that native dirac decoder output is not 100% same as reference implementation... [15:32] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:495cb44172d0: setpts: deprecate RTCTIME, we have time(0) which is more generic [17:02] Action: durandal_1707 cant understand how one can mistype ftp into tfp [17:04] Action: michaelni acn [17:22] <cone-871> ffmpeg.git 03Matthieu Bouron 07master:40b4468f6231: lavc/dnxhddata: fix bitrates for cid 1251 and 1252 in cid table [17:30] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:13aca070ab83: gifdec: resync support [17:48] <cone-63> ffmpeg.git 03Stefano Sabatini 07master:e4e36a4dd290: doc/filters: apply minor fixes [17:55] <durandal_1707> lol why gif muxer does not use gif encoder but encodes on its own? [17:56] <beastd> durandal_1707: dunno, maybe same reason gif demuxer didn't use gif decoder long times ago? [17:57] <durandal_1707> beastd: yes but that code was removed... some thougt that muxer can live... [17:58] <beastd> durandal_1707: muxer was not discussed at that time [17:58] <beastd> IIRC the demuxer got into focus because Reimar (or maybe someone else) detected vulns in the demuxer [17:59] <durandal_1707> lol muxer even have some kind of lut table [17:59] <beastd> durandal_1707: you plan to work on the gif muxer? i guess cleanup would be appreciated too [18:01] <durandal_1707> well, it is much more complicated ... [18:14] <durandal_1707> why is only way to use showwaves/showspectrum via amovie? [18:19] <durandal_1707> saste: i was wondering how to make showspectrum to render all channels and in more vivid colors [18:19] <saste> durandal_1707, with ffplay? [18:20] <saste> at the moment yes [18:20] <saste> unless you want to port filter_complex from ffmpeg [18:20] <durandal_1707> ffplay does not support even -af [18:20] <saste> and i have a patch for adding -af to ffplay (but which doesn't help in this case) [18:21] <durandal_1707> and why it is not on ml? [18:22] <saste> durandal_1707, it is [18:22] <saste> but in an old thread [18:22] <saste> i still have to address some issues [18:22] <saste> no one cares to comment about my image2 RFC? [18:24] <beastd> saste: is it on ML? [18:24] <saste> beastd, [RFC] image2 and filename patterns [18:24] <beastd> if yes i will read it later [18:25] <beastd> is it related to the recent tickets on trac? [18:25] Action: beastd works on sth else at the moment [18:25] <beastd> but will read it later and comment tomorrow if that is not too late [18:27] <saste> beastd, which tickets? [18:28] <beastd> saste: not sure and for some reason i can't check. but i had the impression there were some tickets related to image2 (input) patterns [18:29] <saste> no but there was a related discussion on ffmpeg-user [18:29] <beastd> ok, will check [19:08] <michaelni> did someone had some / worked on gbrp* output for sws ? [19:08] <michaelni> iam asking because if not ill look into it [19:09] <wm4> I think Daemon404 had a half-working patch? [19:10] <Daemon404> i have teh rgb path [19:10] <Daemon404> but it was not applied since swscale cannot handle a rgb only path [19:10] <Daemon404> it NEEDS a yuv path too [19:10] <Daemon404> otherwise it will crash, rather than chain conversions [19:11] <Daemon404> https://patches.libav.org/patch/27645/ [19:11] <Daemon404> make sure to read the replies [19:12] <michaelni> can i see the patch ? [19:12] <Daemon404> (i got teh title backwards!) [19:12] <Daemon404> i just linked [19:15] <bcoudurier> why gbrp instead of rgbp ? [19:16] <wm4> because why miss an opportunity to confuse everyone? [19:21] <Daemon404> bcoudurier, i just worked with whatever colorspace was already being used by things [19:21] <Daemon404> and existed [19:21] <Daemon404> ask whoever implemented it =p [19:27] <saste> what should I do with CREDITS? [19:27] <saste> michaelni, ok to mention the missing names in the log message? [19:29] <wm4> saste: btw. I attempted to write a patch for the cookie issue mentioned yesterday; fixing the error handling was simple enough, but I'm not really sure what the semantics are for domain matching + port names [19:30] <saste> wm4: what about CC-ing the author of the code (Micah Galizia)? [19:30] <saste> i'm not an HTTP expert, but if you post the patch i'll try to review it [19:40] <cone-63> ffmpeg.git 03Stefano Sabatini 07master:e80be5a0aa9f: lavfi/showwaves: fail in case of push_frame() error [20:14] <michaelni> saste, IMO removing names that are then mentioned nowhere expcet a log message of one "random" commit doenst feel correct [23:37] <michaelni> Daemon404, GBRP patches on ffmpeg-dev, your review is welcome if you want to review them [23:46] <wm4> michaelni: can anything be done about the remaining missing conversions? [23:48] <michaelni> wm4, which are missing ? [23:49] <michaelni> ill add gbrp16 output support too probably ... [23:50] <wm4> michaelni: some 64 bit RGB formats according to format_entries[], and some more weird formats [23:51] <cone-154> ffmpeg.git 03Michael Niedermayer 07master:c6ef7641fc12: ffv1enc: fix gbrp>8bit [23:53] <michaelni> i dont see a problem with adding the 64 bit formats [23:54] <michaelni> what uses them ? [23:54] <wm4> no idea [23:55] <wm4> oh right, PNG [23:57] <michaelni> ok, that should be useable for testing [00:00] --- Sat Jan 26 2013

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ffmpeg.log.20130125
by burek 26 Jan '13

26 Jan '13

[00:17] <KeshlWare> I has a PNG sequence. How do I loop it 3 times during encode? xwx [00:35] <notedible_> is there any way to mux dvdsubs into MP4 or will this never be supported since it is non-standard (although mp4box / nero recode / etc. support muxing)? [00:35] <notedible_> using ffmpeg 1.1.1 [01:09] <JEEB> notedible_, I'm pretty sure DVD subs in "mp4" is nero-specific, and mp4box sure found nothing else to do if they implemented it :V That said, nero chapters got implemented, too. [01:10] <JEEB> it is very much non-standard [01:10] <notedible_> JEEB: you're right, mp4box only transcodes from other softsubs to TTXT [01:11] <notedible_> is there a way to dump them to a file from ffmpeg? [01:12] <notedible_> i don't know what extension to put (.sub is a softsub format) [01:39] <frozenbone> hi folks - i've got a raw h264 video without a container - i've used ffmpeg to put it in an h264 container with -vcodec copy, but it takes up twice as much disk (one for the orig, one for the copy). Anyone know of a way to containerize a file inplace? Without the copy? [01:51] <stephanedev> frozenbone: cannot you delete the original file once you're done if you don't want it anymore? [01:52] <frozenbone> Thanks stephanedev - I can, but I'm up against a space crunch :-) Was holding out on buying another hard disk until I asked [01:54] <stephanedev> well, all i can say is that most of the time programs don't write in-place. if it looks like they are doing it, quite often they actually create a new file, write to it, delete the original one and rename the old one to the original name [01:55] <stephanedev> this helps with not losing data too in case of failure. imagine your computer crashes while overwriting the original file [01:56] <frozenbone> Thanks stephanedev for the reply :-) Totally understand [01:56] <stephanedev> you're welcome [03:12] <ulatekh> I'm trying to pack raw bottom-field-interlaced YUV into a container (e.g. "ffmpeg -i input.yuv -vcodec copy output.avi") but when I look at the result (i.e. "ffmpeg -i output.avi -vcodec rawvideo -f yuv4mpegpipe - | head -1") the header now says it's progessive! How do I preserve interlacing? Is this a bug? I'm using ffmpeg 0.10.6 on Fedora Core 17. [03:19] <ulatekh> Anyone awake here? [03:20] <michaelni> ulatekh, its probably a feature request [03:20] <michaelni> for .avi at least [03:20] <ulatekh> Should I be using a different container? [03:21] <ulatekh> I just want my raw YUV in a format that kdenlive will accept, since it won't accept a .yuv file. [03:23] <michaelni> maybe .y4m or .mov would work [03:23] <michaelni> you can also open a feature request on the bug traker for the avi case [03:24] <ulatekh> I've had bad luck with ffmpeg requests in the past...if I get any response at all, it's usually "send a patch", i.e. they expect me to fix it. [03:32] <ulatekh> Encoding to .mov worked, but decoding to raw video again (i.e. "ffmpeg -i output.mov -vcodec rawvideo -f yuv4mpegpipe -pix_fmt yuv444p - | head -1") freaks out with an endless stream of "Error while decoding stream #0:0" messages. [03:33] <ulatekh> y4m decode worked, but the header says the video is progressive now. [03:37] <michaelni> ulatekh, you will have to submit a feature request then ... or a patch ;) [03:38] <ulatekh> After seeing http://ffmpeg.org/doxygen/trunk/huffyuv_8c-source.html online, like 636 suggested adding "-flags +ilme" to the command line...result still progressive...argh [03:38] <ulatekh> like -> line [03:50] <ulatekh> I grabbed the source for my version of ffmpeg (i.e. "yumdownloader --source ffmpeg") and its libavcodec/huffyuv.c also sets the interlaced flag if "-flags +ilme" is set on the command line...but still no interlaced... [03:51] <ulatekh> Hey, the comment at the top of huffyuv.c says "michaelni"...dude, this is your code :-) [04:02] <ulatekh> Oh well, guess I'll have to work on this later...gotta sleep [04:48] <Demon_Fox> I think "Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16, 224 kb/s" Sounds good for a CD [05:16] <Mista_D> can I have ffprobe use `-vf idet` to check for interlacing, i need a json/csv output, instead of stderr?? [05:24] <Mista_D> anytime I run `ffprobe -filters` it shows none... compiled 5 versions already 0.6 - 1.1.1 [05:26] <Mista_D> is there a way to enable libavfilters in ffprobe? [05:51] <aboba> Does this channel also support avconv? [05:53] <aboba> I'm having some serious issues doing streaming with ffmpeg to twitch, and I think I might be screwing up my options somehow [05:53] <aboba> I've played with a whole bunch, but just can't seem to get it right [05:53] <klaxa> for avconv support visit #libav [05:55] <aboba> I set this all up using ffmpeg originally [05:55] <aboba> some options [05:55] <aboba> maybe thats the problem [05:55] <aboba> http://pastebin.com/PFD8FsGN [07:34] <fling> I have a fine script http://bpaste.net/show/72678/ [07:34] <fling> hom may I add a file length hh:mm:ss into a filename? [09:39] <pzich> is it possible to get ffmpeg fast seek to properly sync with audio? [09:40] <pzich> I'm currently running the same command but only changing the location of the -ss flag relative to the -i, the fast seek is much faster, but the sound is pretty horribly off [09:40] <pzich> if you have any other flag tips while you're at it: http://pastebin.com/qSkusKMK [09:53] <jeje> hi to all [09:53] <jeje> I use ffmpeg to decompress video streaming from IP cameras [09:54] <jeje> and I have to do some strange operation on the line size [09:54] <jeje> av_image_fill_linesizes(m_lpFrame->linesize, PIX_FMT_YUV420P,dwWidth+(dwWidth%32)+32); [09:55] <jeje> if I don't do this, all width resolution which are not divisible by 32 have a wrong stride in the renderer (make by swscale) [09:55] <jeje> If someone can explain me this part, regards [09:59] <jeje> for information, I use ffmpeg version I found on Latest Zeranoe FFmpeg Build Version 32 bits [10:04] <jeje> the video compression from camera is H264 of course [10:07] <jeje> the renderer is directly make on DirectDraw surface in RGB32 [10:09] <jeje> if needed I can make a pastebin code [11:05] <jeje> also with the last version of ffmpeg, I have a crash when using avcodec_default_free_buffers [11:05] <jeje> is this function deprecated? [11:28] <jeje> please someone can make a comment because It seems I just see the join and quit in my IRC soft [11:38] <Samus_Aran> can ffmpeg dump single frames for specific timestamps in a video? [11:40] <Samus_Aran> and if so, can it do it with a high level of time precision? mencoder is very inaccurate in this regard. [11:40] <durandal_1707> Samus_Aran: try select filter [11:41] <durandal_1707> it select decoded frames though... [11:42] <Samus_Aran> I'm trying to automate this from the command line [11:49] <Samus_Aran> the man page doesn't have any examples of the select filter, I'll try a web search, thanks. [11:50] <durandal_1707> Samus_Aran: man page is not documentation, documentation is on web page [11:51] <durandal_1707> http://ffmpeg.org/documentation.html [11:53] <jeje> durandal_1707>thanks. About the buffer padding, I only see the input buffer must be padded with FF_INPUT_BUFFER_PADDING_SIZE [11:53] <Samus_Aran> durandal_1707: man pages very often are documentation, i.e. man bash, man screen, etc. [11:54] <Mavrik> not here. [12:09] <Samus_Aran> what does this mean: "Buffering several frames is not supported. Please consume all available frames before adding a new one." [12:09] <Samus_Aran> my command was: ffmpeg -i Video.mkv -r 1 -vframes 120 -ss 01:30:14 Snapshot-%5d.jpg [12:11] <jeje> Can someone explain me how to get and set the padding value for an AVFrame according to the width and height of the resolution .... [12:14] <jeje> because I only use avcodec_alloc_frame to get my AVFrame, but I think I have to set a padding value [12:15] <jeje> I need to call av_image_fill_linesizes(m_lpFrame->linesize, PIX_FMT_YUV420P,dwWidth+(dwWidth%32)+32) to have a good result [12:16] <jeje> but I can't understand why and durandal_1707 tell me I have to pad and memset buffers [12:16] <jmbccoffee> av_image_alloc ? [12:16] <durandal_1707> buffers are allocated by other means ... [12:18] <durandal_1707> jeje: your buffer that you allocate must be padded [12:20] <jeje> but when I call avcodec_alloc_frame() to get my AVFrame, how can I set the padding buffer values? [12:21] <jeje> because I don't allocate buffers [12:21] <jeje> and it work [12:30] <durandal_1707> that function allocate AVFrame but not buffers..... [12:31] <jeje> I need to do a malloc or call an FFMPEG funvtion to alloc buffer so? [12:36] <jmbccoffee> i just do an alloc_frame, image_alloc, and i never touch a video buffer [12:37] <jeje> in the documentation of AVFrame, I just see: The buffers for the data must be managed through other means [12:38] <Mavrik> avpicture_alloc is the call that fills AVFrame buffers properly [12:39] <jmbccoffee> which is like obsoleted by image_alloc [12:40] <Mavrik> mhm, av_image_alloc actually [12:42] <jmbccoffee> he, of corse [12:58] <Samus_Aran> can someone tell me why this command eats up all my system RAM and swap before I can kill it: ffmpeg -i Video.mkv -ss 1:13:13.13 -vframes 1 Shot-%6d.jpg [13:00] <Samus_Aran> same error I mentioned a few minutes ago: [buffer @ 0xf43780] Buffering several frames is not supported. Please consume all available frames before adding a new one. [13:00] <Samus_Aran> I'm just trying to save one frame... [13:15] <Samus_Aran> this seems to work: ffmpeg -ss 1:13:13.13 -i Video.mkv -vframes 1 output.jpg [13:25] <durandal_1707> Samus_Aran: you can archive same with select filter [13:26] <burek> Samus_Aran, try: ffmpeg -ss 1:13:13.13 -i Video.mkv -vframes 1 Shot-%6d.jpg [13:27] <burek> (move -ss before -i) [13:27] <burek> it shouldn't crash your machine) [13:27] <burek> i really should read all the messages before replying :) [13:37] <Samus_Aran> burek: if it's after, it eats all the system RAM, then all the swap, then I can kill it [13:37] <burek> yes [13:37] <burek> you are actually parsing each frame that you decode [13:38] <burek> and i guess ffmpeg buffers it (you didn't show your full log, so we can't tell more than this) [13:38] <Samus_Aran> it repeated that message thousands of times [13:38] <Samus_Aran> presumably for every frame [14:12] <durandal_1707> Samus_Aran: select='gte(t\,10)*lte(t\,20)' <-- Select only frames contained in the 10-20 time interval [14:17] <Samus_Aran> durandal_1707: I am trying to grab an individual frame for a given timestamp [14:17] <Samus_Aran> which I have working now with the above command [14:18] <Samus_Aran> I just tested on a 29.970 FPS video, doing the match for the first 29 frames, and they were all dumped correctly (each a different frame), so it appears to be time-accurate [14:18] <Samus_Aran> *doing the math [14:19] <Samus_Aran> -ss $frame/29.970 [14:24] <durandal_1707> Samus_Aran: are you telling me you are calling ffmpeg multiple times to grab single frame? [14:47] <funyun> hi. can anyone tell me what these errors mean and what i can do to fix them? http://pastebin.com/nwc0qde4 [14:50] <JEEB> broken H.264 stream? [14:56] <funyun> JEEB: any idea how to fix that? [14:57] <JEEB> no idea :P [14:57] <JEEB> you could poke the issue tracker with a sample [14:57] <JEEB> and get a better explanation [15:35] <Samus_Aran> durandal_1707: what do you mean? I am calling ffmpeg once to grab one frame [15:36] <durandal_1707> Samus_Aran: so you need only single frame frome single file? [15:38] <durandal_1707> if you need multiple frames from single files select filter is better tool [15:38] <Samus_Aran> durandal_1707: if you want to tell me how to dump frames at specific intervals, feel free. but if it requires the entire video to be decoded, it isn't of any use [15:39] <durandal_1707> it should not need entire video to be decoded ... [15:39] <Samus_Aran> it seemed that is what the select filter would be doing [15:40] <durandal_1707> seemed? [15:40] <Samus_Aran> it's a filter, filters do things to frames [15:40] <durandal_1707> you actually tried it? [15:40] <Samus_Aran> at least in other video editing apps [15:40] <Samus_Aran> I have no clue how to use it [15:41] <Samus_Aran> the man page doesn't explain, and the documentation I was pointed to isn't indexed well enough for me to have any idea where to look [15:41] <durandal_1707> so is your solution working good for you? [15:41] <Samus_Aran> it seems to be accurate to the frame. [15:42] <Samus_Aran> takes about 0.5 to 1.0 seconds to dump a frame in PNG [15:46] <Mavrik> Samus_Aran: doing a full decoding pass and grabbing frames you need is almost always the most efficient way of doing what you're trying to do [15:46] <Mavrik> especially if number of frames is large [15:48] <durandal_1707> Samus_Aran: so you have any further questions? [16:00] <Samus_Aran> Mavrik: it takes me 1 second to grab a frame using my method, entire decoding would take over an hour. how is that more efficient? [16:01] <Mavrik> entire decoding would take over an hour? what? [16:01] <Mavrik> are you doing that on 300MHz ARM? [16:01] <Samus_Aran> Mavrik: decoding a 3 hour 720p video [16:01] <Mavrik> yes? [16:01] <Samus_Aran> takes hours [16:02] <Mavrik> well, that presumption is wrong. [16:02] <Samus_Aran> if it is decoding the entire video, it isn't [16:02] <Samus_Aran> it uses about 90% CPU time to play the video. [16:03] <Mavrik> 300MHz ARM? [16:03] <Samus_Aran> Mavrik: you're not being helpful in the slightest [16:04] <Mavrik> yes, because you're not reading what I write. [16:04] <retard> i've been using http://pastebin.com/1u96HKuz to extract clips from a file beginning at keyframes [16:04] <Mavrik> you managed to miss this too: < Mavrik> | especially if number of frames is large [16:04] <retard> which is ugly and slow but works [16:04] <Mavrik> mhm [16:05] <retard> without this convoluted bullshit the audio begins at the specified time while the video plays from the nearest keyframe [16:05] <Mavrik> if you give seek parameters before "-i" ffmpeg will skip to keyframe [16:05] <retard> leading to desynch in certain players [16:05] <Mavrik> and then do a decoding pass until your set time [16:05] <Mavrik> at least in new versions [16:05] <retard> it still desynchs the audio, i tried [16:05] <Mavrik> doesn't work on all videos though [16:05] <Mavrik> yeah [16:06] <retard> this script works with everything i throw at it, but it has to parse every god damn frame [16:06] <Samus_Aran> Mavrik: -ss skips to any frame, not just keyframes. I have tested it accurately on a 29.970 FPS video to capture individual frames. [16:06] <Mavrik> and again, you're not reading what I write. [16:06] <Samus_Aran> <Mavrik> if you give seek parameters before "-i" ffmpeg will skip to keyframe [16:06] <Mavrik> Samus_Aran: and next two lines. [16:07] <Samus_Aran> yes, it does that. it drops 0 to 90 frames when capturing one shot [16:07] <Samus_Aran> on my video sample [16:10] <jeje> thanks a lot at all for explication about the avcodec_alloc_frame and the needed of use av_image_alloc after to allocate the buffer of the AVFrame [16:10] <jeje> it works well now [16:11] <jeje> Just a final question. how to free the buffer of the AVFrame? [16:11] <jeje> Is this call enought: av_free( m_lpFrame->data[0]) and after av_free( m_lpFrame)? [16:11] <Mavrik> jeje: not always [16:12] <Mavrik> (depends on pixel format) [16:12] <jeje> so what is the "good" method to be sure to free all allocated? [16:12] <Mavrik> jeje: documentation says "The allocated image buffer has to be freed by using av_freep(&pointers[0]). [16:12] <Mavrik> " [16:12] <Mavrik> for av_image_alloc [16:13] <Samus_Aran> Mavrik: for my usage, there will be a possibly large gap between frame captures (e.g. 20-30 seconds) [16:13] <jeje> ok so I need to do av_freep(m_lpFrame->data[0]) and after av_free (m_lpFrame)? it looks like better? [16:13] <Mavrik> jeje: yea [16:14] <jeje> thanks a lot [16:45] <jeje> Hi, I have a crash in my application when I use av_freep( m_lpFrame->data[0]). [16:46] <jeje> i check before my AVFrame* m_lpFrame is not null (of course) and check if (m_lpFrame->data[0]!=NULL) but when I use av_freep(m_lpFrame->data[0]) I've got an exception on delete operator [16:50] <jeje> To allocate my AVFrame pointeur, I use m_lpFrame = avcodec_alloc_frame() and av_image_alloc(m_lpframe->data,...) [16:56] <jeje> sorry I just found my error : av_freep( &m_lpFrame->data[0]) [16:56] <yermandu> how i can convert to divx, my dvdplayer said that accepts only dvix 556 :| [19:40] <lodenrogue> hello [20:00] <Diogo> this is possible ffmpeg the output time in seconds?? [20:01] <Fjorgynn> what? [20:01] <llogan> i don't understand the question [20:01] <Fjorgynn> -t 60s or something? [20:02] <llogan> you don't need the "s" [20:06] <lodenrogue> i cant seem to install ffmpeg on Gnu/Linux mint [20:06] <Fjorgynn> why? [20:07] <lodenrogue> I tried installing it via the ./configure make make install but after it does all that there is no executable file to run. I also tried to install it via the package manager and it wont appear anywhere in the computer [20:10] <Fjorgynn> $ sudo apt-get install ffmpeg [20:10] <Fjorgynn> lodenrogue: appear? You just do $ ffmpeg in the terminal... [20:11] <lodenrogue> Fjorgynn: how do I get the program to run? I do ffmpeg in the terminal and all I get is release info [20:11] <Fjorgynn> lodenrogue: ffmpeg isn't gui [20:11] <lodenrogue> ??? [20:11] <lodenrogue> What? [20:11] <lodenrogue> lol omg [20:11] <lodenrogue> ok thanks [20:11] <Fjorgynn> JEEB: give me a pillow [20:12] <lodenrogue> is there a gui version of ffmpeg? [20:13] <lodenrogue> or something similar with a gui option? [20:13] <Fjorgynn> for what? [20:13] <lodenrogue> I want to do gameplay live stream on twitch.tv [20:14] <Fjorgynn> aha [20:15] <Fjorgynn> and why do you want ffmpeg? [20:15] <lodenrogue> I cant find any other GNU/Linux live stream recorder. And I read online that ffmpeg does that [20:16] <Fjorgynn> http://linuxgamecast.com/2012/08/l-g-c-how-to-broadcasting-to-twitch-tv-and… [20:17] <lodenrogue> thanks man! [20:17] <Fjorgynn> I googled twitch linux [20:17] <lodenrogue> I get too specific with my searches so I miss the simple solutions. Thanks [20:19] <hotwings> is ffmpeg git from videolan down right now? i get "Cloning into 'ffmpeg'..." but it just sits there forever [20:20] <JEEB> yes [20:20] <JEEB> and if you're on msys/mingw then it will clone like that [20:21] <JEEB> (it doesn't have the detailed view by default or whatever) [20:21] <hotwings> im using linux, which should give me more [20:22] <hotwings> and git://source.ffmpeg.org/ffmpeg.git is just a redirect to videolan? [20:22] <JEEB> yes [20:22] <JEEB> git://git.videolan.org/ffmpeg.git is the same [20:28] <hotwings> ok so it would be pointless to try cloning ffmpeg.org then. the way today has gone so far, im not surprised ... :\ [20:43] <llogan> hotwings: source.ffmpeg.org is the "official" one despite them currently being the same [20:44] <JEEB> I don't see any real reason to move off of videolan though [20:45] <beastd> JEEB: no reason ATM [20:45] <JEEB> well yeah, maybe in case videolan gets destroyed or if there is :drama: between videolan and ffmpeg [20:45] <JEEB> for whatever reason I don't really see that happening, though :) [20:45] <beastd> yeah [20:46] Action: beastd never saw any reason to split ffmpeg developer community and surely would have not seen it coming 5 years ago [20:46] <llogan> developering is hard. let's go shopping. [20:47] <JEEB> hats would be good [20:47] <JEEB> D_S would join us for that too, probably! [20:47] <llogan> handbag! [20:49] <hotwings> [11:43:49] <llogan> hotwings: source.ffmpeg.org is the "official" one despite them currently being the same <-- i was told the opposite a couple months ago :\ [20:49] <JEEB> hotwings, the configure script currently derps at you if your remote is not source.ffmpeg.org >_> [20:50] <llogan> hotwings: refer to line 4590 [20:50] <JEEB> so yeah, it's the "official" way of accessing it [20:51] <llogan> i never did end up with many hats in TF2... [20:51] <hotwings> im not saying youre wrong llogan, just noting whatever you were told before (even when its "official"), someone else will say the opposite :) [20:51] <hotwings> its a common theme [20:51] <llogan> then it sounds like to me then you're on the "official" internet [20:51] <hotwings> indeed [20:53] <beastd> hotwings: sorry for the inconvenience. but please just use/change to source.ffmpeg.org . we must put this systems at work now because the lazy way will sure be too late [20:54] <hotwings> i originally used source.ffmpeg.org, but was told its just a mirror or videolan and shouldnt use it but rather use videolan directly :\ [20:54] <JEEB> they shouldn't be mirrors, it's just redirection or whatever [20:54] <JEEB> so basically the same thing [20:55] <llogan> it doesn't really matter at this point. [20:55] <hotwings> why is there even two to begin with? seems pointless [20:55] <beastd> hotwings: tje [20:56] <hotwings> not following you beastd. tje a ffmpeg-scene thing or? [20:56] <beastd> hotwings: nope. typo [20:57] <beastd> hotwings: source.ffmpeg.org is just the DNS name you should use. the other one is just there because the repo is hosted by videolan and that is where the ffmpeg.git is at the moment [20:57] Action: llogan adopts tje as a ffmpeg-scene thing. [20:57] <beastd> tje :) [20:57] <hotwings> lol [20:57] <hotwings> tje! [20:58] <beastd> historically source.ffmpeg.org did not exist when ffmpeg.git repo was created. that is the reason those two names are in the wild [21:06] <Diogo> this is possible ffmpeg the output time in seconds?? [21:14] <Fjorgynn> Diogo: how do you mean? [21:14] <Fjorgynn> .... [21:23] <Diogo> change the formtar time to parse to a external program? [21:23] <Diogo> timeseconds not time for human reading? 00:00:20 [21:24] <Diogo> this is possible native ffmpeg outpout a json? [21:25] <Diogo> with a status of encoding? [21:27] <llogan> Diogo: ffprobe can output json: ffprobe -print_format json -show_streams input.foo [21:35] <Diogo> no and ffmpeg [21:36] <Diogo> during the encoding... [22:56] <mulicheng> Anyone using VDA hwaccel on OSX for h264 decode? [22:58] <anesbit> hi all. i need to determine exactly how ffmpeg converts multichannel (esp. 2-channel or stereo) audio input into single-channel (mono) output. in other words, i need to know how the "-ac 1" output option works when presented with multichannel inputs. [22:59] <anesbit> i'm pretty new to navigating the ffmpeg source code, so does anybody have any pointers, either as to where i should start looking, or what algorithm ffmpeg (or its libraries) uses? [23:00] <JEEB> ffmpeg uses libswresample, and I would guess you would have to look at ffmpeg.c to see which exact settings it uses for it [23:01] <anesbit> JEEB: thank you. [23:01] <anesbit> i'll have a look [23:05] <mulicheng> I stepped through the h264 code and found that the codec->id is always 0. Decoding works for me but the hwaccel is never found. I'm curious what initialization steps I may have missed to use hwaccel. [00:00] --- Sat Jan 26 2013

1 0

ffmpeg-devel.log.20130124
by burek 25 Jan '13

25 Jan '13

[00:44] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:7357ca900efc: sanm: Check decoded_size. [00:55] <llogan> anyone care or not if i get ffmpeg listed here? http://linuxaudio.org/members it would show that we support other (audio)/media related projects [00:55] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:9d6e0ac6737f: floatdsp: restrict->av_restrict [00:55] <llogan> http://linuxaudio.org/policy [00:57] <Compn> looks like another worthless thing , go for it :P [00:57] <Compn> ehe [00:58] <llogan> they has a lenghty (a lot of OT) thread on their user ML about ffmpeg vs libav. [00:58] <Compn> lol [00:58] <llogan> the OT involved monkeys. [00:58] <llogan> which i support. [01:02] <llogan> at most, if their foundation gets formed eventually, they could help sponsor trips to conferences, etc if we lack donation monies. [01:05] <llogan> not that any of us actually attend much social stuff (hissss!) AFAIK [01:19] <saste> https://ffmpeg.org/ffmpeg-filters.html#concat: There are nÃ(v+a) [01:19] <saste> i am the only one seeing that? [01:20] <Daemon404> no [01:21] <saste> ok, so no UTF-8 support in texi2html, shame [01:21] <saste> or we're doing it wrong [01:21] <saste> and same in the manual [01:26] <cone-16> ffmpeg.git 03Carl Eugen Hoyos 07master:f8b6d4818e1d: Only try to auto-detect LATM in mpegts if the LOAS demuxer was configured. [01:26] <cone-16> ffmpeg.git 03Carl Eugen Hoyos 07master:df39c3ce385c: matroskaenc: add codec_tag lists back. [01:34] <michaelni> saste, it displays correctly if you force utf8 [01:35] <saste> michaelni, man/html? [01:35] <michaelni> html [01:35] <michaelni> with firefox [01:35] <saste> we need to set the encoding in the generated page then [01:35] <saste> as for the man output i can't say... [01:35] <michaelni> i think so [01:47] <teratorn> anyone a gdb expert I could bother?? :) how can values in an expression be optimized out, and also at the same time cause a SIGSEGV? http://hastebin.com/maciwiyide.txt [01:48] <teratorn> (that's from a core file, if it matters) [01:49] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:ee9151b616fa: ff_mss12_decode_init: check dimensions [01:50] <michaelni> teratorn, i suggest that if possible you recompile without optimizations and get a better core file [01:51] <teratorn> michaelni: thank you, that's a good suggestion. I'm suspect of the data I'm passing in to the codec. [01:53] <saste> ^ i suspect this is a form of "silly" encryption [01:56] <saste> michaelni: ping on doc/eval: fix/extend documentation for taylor() function [01:57] <saste> no hurry since i'll work on it tomorrow evening, but i'd like to push the whole patchset [02:49] <cone-16> ffmpeg.git 03Carl Eugen Hoyos 07release/0.10:4e869e7a5f14: matroskaenc: add codec_tag lists back. [02:49] <cone-16> ffmpeg.git 03Carl Eugen Hoyos 07release/0.11:ebb3a5974fe3: matroskaenc: add codec_tag lists back. [02:49] <cone-16> ffmpeg.git 03Carl Eugen Hoyos 07release/0.9:9103d77ffc0e: matroskaenc: add codec_tag lists back. [02:49] <cone-16> ffmpeg.git 03Carl Eugen Hoyos 07release/1.0:d80702228320: matroskaenc: add codec_tag lists back. [02:49] <cone-16> ffmpeg.git 03Carl Eugen Hoyos 07release/1.1:33769e908dac: matroskaenc: add codec_tag lists back. [03:45] <burek> if anyone is skilled in ProRes, please tell me what to answer to this guy: http://ffmpeg.gusari.org/viewtopic.php?f=16&t=802 [03:48] <Compn> he just wants that other color profile [03:48] <Compn> not a matrix [03:49] <Compn> burek : http://ffmpeg.org/pipermail/ffmpeg-user/2013-January/012378.html [03:49] <Compn> ;) [03:50] <llogan> I do so hate mediainfo... [03:50] <burek> :) [03:50] <burek> yes, i see [03:50] <burek> so does he need some color conversion or something [03:51] <llogan> burek: talking about beer, someone is bringing over a 5 gallon keg in an hour... [03:51] <burek> horror... [03:51] <Compn> burek : no, his problem is that hes believing mediainfo is correct :) [03:51] <Compn> when in fact, its not at all :) [03:51] <burek> oh I see [03:51] <burek> so, his problem is not a problem at all? :D [03:51] <burek> I like those kinds of posts :D [03:52] <Compn> note that he never specifies what his true problem is [03:52] <Compn> his problem is that mediainfo reports something ? does it cause any actual problems or is he just crazy ? [03:52] <burek> :))) [03:52] <Compn> ask him, because honestly i cant tell from his post [03:53] <burek> so, his mediainfo shows 601 because ffmpeg didn't write that info in the atom [03:53] <burek> if i understand correctly [03:53] <burek> I mean, i dont want to ask him if he is crazy [03:54] <burek> he did help other guy in the forum while waiting for his answer :) [03:54] <burek> the least i could be is to be polite :) [03:54] <Compn> yes, i think your answer correct [03:54] <Compn> but ask him if that ffmpeg non-atom is causing real problems [03:54] <burek> ok, I will, thanks a lot for the explanation [04:12] <Compn> well , probably i am wrong [04:13] <Compn> most of these people that talk about bt709 and such are ... like the audiophile people [04:13] <Compn> they have no clue what they need, but they saw it on a box once so now thats the most important feature for thier cat videos [04:14] <Compn> and obviously, they want the bt709 instead of the bt 601 because ... 709 is a higher number. [04:46] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:69fb605ad5e0: mpc8: check stream count before accessing stream 1. [04:46] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:b53ed19aa74c: lcldec: Check length before unsigned subtraction. [04:46] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:46cb61819d86: gifdec: check that the last keyframe exists and has been successfully parsed. [04:46] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:7e059c9c5d78: mpeg4videoenc: check w,h to be within the supported range. [09:01] <burek> Compn lol :D [09:32] <EricAhn> I'm going to test from HLS to RTp stream. [09:33] <EricAhn> But, I got an error, "AAC with no global headers is currently not supported.", more details at http://pastebin.com/EajeP5bQ [09:33] <EricAhn> did not supported ffmpeg/ffserver HLS(H.264/aac) => H.264/aac ? [09:34] <EricAhn> I found old irc log, ( http://lists.ffmpeg.org/pipermail/ffmpeg-devel-irc/2010-May/000107.html ) [09:36] <EricAhn> also, http://web.archiveorange.com/archive/v/yR2T4ccIkwqdDXgCRZm0 read, but I'm not sure how much related about this [10:23] <cone-871> ffmpeg.git 03Carl Eugen Hoyos 07master:fc50175ba21f: Refuse to mux tta into matroska, the output file is broken. [11:40] <cone-871> ffmpeg.git 03Paul B Mahol 07master:e65046bf836b: lavfi/earwax: remove config_input() [11:58] <cone-871> ffmpeg.git 03Paul B Mahol 07master:056664ff25c6: lavc/tta: remove nonsense s->avctx indirection, use avctx directly [12:11] <durandal_1707> michaelni: why matroska muxer writes not matroska tags? [12:14] <durandal_1707> instead it writed riff/wav ones I do not think this is valid [12:28] <cone-871> ffmpeg.git 03Stefano Sabatini 07master:2ed0803c6c99: lavu/eval: extend if/ifnot functions to accept a third parameter [12:28] <cone-871> ffmpeg.git 03Stefano Sabatini 07master:2b207bab196d: doc/eval: substitute if/then/else construct with an example making use of boolean expression composition [12:28] <cone-871> ffmpeg.git 03Stefano Sabatini 07master:ca1bc188f4e7: doc/eval: fix documentation for time() function [12:28] <cone-871> ffmpeg.git 03Stefano Sabatini 07master:c89a8ee23d7c: doc/eval: fix/review the section about SI prefixes and usage [13:01] <wm4> why is the max_analyze_duration message printed as warning? [13:01] <wm4> how is that a warning? [13:02] <cone-871> ffmpeg.git 03Paul B Mahol 07master:9ddf5631e319: matroskaenc: support TTA muxing [13:02] <cone-871> ffmpeg.git 03Paul B Mahol 07master:ee8d4a414288: matroskaenc: fix -codec copy with TTA [13:03] <wm4> durandal_1707: have you found out how the length of the last sample for tta in mkv is calculated? [13:03] <durandal_1707> wm4: nope, ask matroska creators [13:17] <wm4> avio_open2() sure is a wtf [13:18] <wm4> so I can't use av_opt directly in the context [13:18] <wm4> but I have to create an AVDictionary first? [13:18] <cone-871> ffmpeg.git 03Paul B Mahol 07master:dd5689a3bdaa: matroskadec: export codec bits_per_coded_sample [13:20] <wm4> and it mutates the AVDictionary you pass... wat [13:22] <saste> wm4, about the AVDict change, anton insisted with that design because it considered that "simpler" for the user [13:23] <wm4> sigh [13:23] <saste> but again we are not the authors of that API, ask/complain to libav [13:24] <wm4> anton = elenril, right? [13:24] <durandal_1707> whois is your friend [13:39] <saste> wm4, and i think you *can* use av_opt in the AVIO context, once it was created [13:40] <wm4> but then it already has connected [13:43] <saste> wm4, what's the problem with specifying options in the dictionary? [13:43] <wm4> it's suddenly a different API, and different memory management [13:44] <wm4> also it always forces you to use strings [13:44] <saste> wm4, i see [13:45] <saste> i can't remember (or never knew) why a more avcodec_open()-like API was choosen [13:45] <saste> *was not choosen [13:45] <saste> i assume there were good reasons for that [13:45] <wm4> <wbs> but you can't easily pass parameters before the file/stream is opened [13:45] <wm4> <wbs> otherwise you'd need to split it into a "parse url, allocate contexts but don't actually open it" step [13:45] <wm4> <wbs> after which the caller could set options [13:45] <wm4> <wbs> and then tell it to actually start the connection [13:46] <wm4> so at least it's easier to implement [13:48] <saste> yes [14:18] <cone-871> ffmpeg.git 03Martin Storsjö 07master:e90820d4f815: rtp: Make sure priv_data is set before reading it [14:18] <cone-871> ffmpeg.git 03Martin Storsjö 07master:57ed8debb9b9: wmv2: Propagate the wmv2 idct permutation type to the dsputils context [14:18] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:40c27504a5fd: Merge commit '57ed8debb9b9cc565cc6e9f98c5b5cbb9f69097c' [14:18] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:1d2abca7e1f0: wmv2enc: drop setting of idct_algo [14:27] <cone-871> ffmpeg.git 03Martin Storsjö 07master:932117171f32: rtp: Make sure the output format pointer is set [14:27] <cone-871> ffmpeg.git 03Martin Storsjö 07master:4a4a7e138c92: rtpenc_chain: Use the original AVFormatContext for getting payload type [14:27] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:b4581e9394c9: Merge commit '4a4a7e138c92901e04db46a6b05cc6948023e5f5' [14:34] <cone-871> ffmpeg.git 03Diego Biurrun 07master:528878ee7b37: openbsd: configure: Stop enabling PIC by default [14:34] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:9c555bca68cf: Merge commit '528878ee7b377e23a194d7c801571d97793047e0' [14:39] <cone-871> ffmpeg.git 03Luca Barbato 07master:7a95afe433b2: doc: fix dependencies in pod generation [14:39] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:70b0aeba001c: Merge commit '7a95afe433b2a692f490b98948c082e62ffc1d27' [14:45] <TimNich> use hit a make issue using my usual build set up with b4581e939 [14:58] <TimNich> OK so it was a one off hiccup! Sorry for the noise. [15:20] <EricAhn> did ffserver support from HLS(means that using ffmpeg input) to rtp(H264/AAC)? [15:54] <cone-871> ffmpeg.git 03Mans Rullgard 07master:e9d817351b28: dsputil: Separate h264 qpel [15:54] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:fc13a89654b5: Merge remote-tracking branch 'qatar/master' [15:56] <durandal_1707> when will this nonsense stop? [15:57] <j-b> mever [15:58] <durandal_1707> all the time something is renamed/moved around/etc ... [15:58] <durandal_1707> i just want to see end of tunel [15:58] <Compn> well [15:58] <nevcairiel> when everything is neatly seperated in small reusable chunks [15:58] <nevcairiel> instead of one big hunk of everything [15:59] <Compn> i think dark_shikari mentioned that our h264 decoder was not as fast as other software h264 decoders [15:59] <durandal_1707> what, every test on web show it is fastest [15:59] <Compn> so maybe a huge cleanup, and shuffle out bits, then merge x264 optimizaitons back , then tweak the decoder ... [16:00] <durandal_1707> hello it is 2013! too late [16:00] <Compn> for single core decoding ? [16:01] Action: Compn tries to find benchmarks [16:01] <wm4> who cares about single core? [16:01] <wm4> even phones have multicores these days [16:02] <nevcairiel> more single core speed = less power used [16:02] <nevcairiel> even translates to multicore [16:05] <cone-871> ffmpeg.git 03Paul B Mahol 07master:7b007a7c1fad: flic: do not set sample_fmt [16:05] <cone-871> ffmpeg.git 03Paul B Mahol 07master:eb567a79990b: eacdata: do not set sample_fmt [16:05] <wm4> sure, but a fast single core decoder probably still won't compete against ffmpeg, even if ffmpeg single-core is slower [16:05] <Compn> look i'm just coming up with one guess as to the h264 split [16:06] <Compn> :P [16:06] <Compn> it was always a hope that x264 would be merged [16:06] <Compn> i know x264 guys dont want to hear that :P [16:06] <wm4> why would you merge it [16:06] <Compn> since its kind of hard to mix gpl and lgpl code in one file , making you use a ton of ifdefs ... [16:07] <wm4> just to increase the amount of code the decoder depends on? [16:07] <nevcairiel> x264 is an encoder [16:07] <wm4> *encoder [16:07] <nevcairiel> it wouldnt help the decoder much [16:07] <wm4> if it works fine stand-alone, leave it be [16:07] <wm4> better fight NIH inside of ffmpeg [16:07] <Compn> nevcairiel : you got the latest coreavc v libavcodec benchmarks ? [16:07] <durandal_1707> wm4: like what? [16:07] <Compn> searching doom9 is giving me a headache [16:07] <nevcairiel> i agree, putting such a big encoder project into ffmpeg just for the sake of doing it doesnt help :p [16:08] <nevcairiel> Compn: no [16:08] <wm4> durandal_1707: TS demuxer [16:08] Action: durandal_1707 still confused [16:10] <Compn> nevcairiel : true, there might be one or two things. maybe they already got ported [16:11] <Compn> do a search for loren in git log ... [16:12] <durandal_1707> there are matroska samples with DTS/EXPRESS and/or DTS/LOSSLESS? [16:12] <nevcairiel> i have never seen one with dts express in it, but lossless is common enough [16:13] <durandal_1707> nevcairiel: have link to sample? [16:13] <nevcairiel> no [16:28] <michaelni> when everything is split up in small parts they will make a party and then start merging it all back together and 5-10years later the cycle repeats [16:29] <michaelni> split up related parts so they become inconsistent, then merge back to make it consistent [16:30] <michaelni> then again split up because its again too much [16:34] <Compn> lol [16:35] <nevcairiel> if the only relation is that its for video decoding, its not a very strong relation =p [16:35] <Compn> i'm surprised there isnt a /h264/ yet [16:35] <Compn> it has more than a dozen files already :P [16:36] <Compn> two dozen... [16:36] Action: durandal_1707 still confused [16:36] <nevcairiel> counting all the arch specific ones? [16:37] <Compn> no, i'd rather not [16:37] <Compn> lots of code in ffmpeg [16:37] <Compn> to be sure [16:38] <Compn> so when are we going to merge into the linux kernel ? [16:38] <michaelni> nevcairiel, example 1 from "yesterday": change the array used for B frame MC functions to a different type than the one used for P frames [16:38] <nevcairiel> i would ask why they have to be of the same type? [16:39] <Compn> consistency [16:39] <michaelni> consistency & ease of use [16:39] <Compn> maybe optimization too... [16:39] <durandal_1707> nobody measured performance implications? [16:39] <michaelni> you wouldnt want to mistakely mix the types [16:39] <Compn> and performance, plus compiler optimization, plus platform might have problems with different pieces of code ... [16:40] <durandal_1707> they usually only measure micro optimizations .... [16:40] <michaelni> nevcairiel, example 2 from "yesterday": the wmv2 idct (which is identical to the mpeg2 reference idct) was split out of the framework for mpeg idcts [16:41] <Compn> and then if you are doing find + replace grep bugs ... it wont show up :) [16:41] <nevcairiel> if its identical, why does it exist? there must be some differences =p [16:41] <michaelni> nevcairiel, ? [16:41] <Compn> michaelni : nevcairiel means duplicate code [16:41] <Compn> "why arent they merged if they are so similar" [16:41] <nevcairiel> if you say its identical to the mpeg2 idct [16:41] <nevcairiel> why isnt the mpeg2 idct used [16:41] <michaelni> nevcairiel, theres only one integer "reference" idct [16:42] <michaelni> and thats not part of the mpeg/jpeg idct framework anymore now [16:42] <nevcairiel> if its called wmv2 idct and only used for wmv2, its hardly something thats reused [16:43] <michaelni> yes, split up for one reason, tomorrow merge for another like consistency [16:44] <av500> you can always not merge it :) [16:44] <michaelni> and it was usefull for debuging / testing non wmv2 files that is if they decode correctly/better with it or any other of the idcts [16:44] <michaelni> av500, yes but that makes future merges harder [16:45] <michaelni> iam lazy [16:45] <michaelni> less work is better [16:45] <nevcairiel> maybe i'm not around long enough, but i havent seen stuff moving from a to b and then back from b to a all that often [16:45] <nevcairiel> and to me, splitting up big blobs into small pieces is always good in any software [16:46] <michaelni> nevcairiel, spliting the idct in a seperate file: good [16:46] <michaelni> using a seprate incompatible framework for it: bad [16:46] <michaelni> thats IMHO [16:49] <durandal_1707> are they comitting without running fate test before ? [16:49] <nevcairiel> you mean like you did? :P [16:51] <durandal_1707> nevcairiel: my commits where small and trivial... [16:51] <nevcairiel> still broke fate :) [16:52] <durandal_1707> yes, but small < big [16:54] <durandal_1707> i think BBB is pushing some very old commits that where previoulsy rejected in days before split [16:55] <EricAhn> I'm going to test from HLS to RTp stream. But, I got an error, "AAC with no global headers is currently not supported.", more details at http://pastebin.com/EajeP5bQ did not supported ffmpeg/ffserver HLS(H.264/aac) => H.264/aac ? [17:15] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:c071b006436d: mpeg12demux: Fallback to startcode for stream type identification. [17:25] <wm4> does avio have anything to reconnect a broken network connection? [17:25] <wm4> or do you have to recreate the avio? [17:47] <EricAhn> wm4 : ping [17:47] <wm4> did you want wbs? [17:47] <EricAhn> wbs means? [17:48] <wm4> I thought you typoed my nick [17:48] <wm4> or his [17:49] <EricAhn> wm4 : I have a question about ffserver. [17:49] <wm4> I'm not an expert in ffserver [17:49] <EricAhn> If i explain something, May u look into? [17:50] <EricAhn> related rtp protocol [17:50] <wm4> as much as anyone on this channel might or might not [17:57] <Compn> ironically, wbs is the guy to ask about rtp :P [17:58] <Compn> EricAhn : did you ask on ffserver-user mailing list? thats the proper place for your question [17:58] <Compn> this channel is more for development of the programs, not user questions :) [17:58] <EricAhn> I found "http://ffmpeg.org/pipermail/ffmpeg-devel/2010-May/096088.html" in ffmpeg-devel [17:58] <durandal_1707> that is 3 years old [17:58] <EricAhn> Compn : I have no idea. [17:59] <EricAhn> Compn : I want to know if that feature didn't implement . [18:00] <EricAhn> Compn : and going to implement that feature. [18:00] <durandal_1707> wtf i did [18:00] <EricAhn> duradal_1707 : ? [18:01] <wm4> wut [18:01] <wm4> you cleared the banlist [18:01] <durandal_1707> omg [18:01] <wm4> now all these trolls will flood this channel! [18:02] <Compn> durandal_1707 : are you not good with computer ? [18:02] <Compn> hehe [18:03] <Compn> EricAhn : ah. [18:03] <EricAhn> durandal_1707 : In my test ffserver.conf, I success only video channel. [18:03] <Compn> EricAhn : i'm not sure who maintains ffserver right now. maybe michael knows [18:03] <Compn> or maybe its in maintainers file [18:05] <EricAhn> if it have a video/audio configuration, H.264/AAC in ffserver.conf, failed play. [18:05] <EricAhn> wbs : ping [18:09] <saste> Compn, bcoudurier is the nominal maintainer for ffserver, but in truth nobody is maintaining it for real [18:11] <michaelni> if someone wants to work on ffserver that person should probably become maintainer of ffserver [18:12] <durandal_1707> .nobody wants. [18:13] <saste> durandal_1707, i wonder why [18:13] <michaelni> its simple people prefer if the code they need working doesnt work [18:14] <durandal_1707> ENODEVS [18:15] <saste> that would explain why ffmpeg has so many bugs :) [18:15] <saste> still there is some traffic going on on ffserver-user [18:15] <saste> i was against the ffmpeg-user -> ffserver-user split [18:15] <Compn> michaelni : should we send eric to ask libav then ? :P [18:17] <saste> what about merging again ffserver-user -> ffmpeg-user? [18:17] <saste> i find a bit silly to have a separate ML for each tool [18:17] <michaelni> libav can maintain ffserver for me, yes, sadly they dont either [18:17] <saste> and most issues are shared amongst tools [18:18] <durandal_1707> saste: i agree, split between tools is little silly, where is ffplay and ffprobe? [18:18] <saste> llogan, ^^ [18:20] <Compn> i think EricAhn is just asking of hyc's patch was ever committed [18:20] <Compn> have to read gitlog for that [18:21] <michaelni> if theres a good patch that wasnt commited ping it and we can review and or comit it [18:21] <EricAhn> Compn : hm okay, will review it [18:22] <michaelni> EricAhn, are you a developer ? do you want to maintain ffserver ? (there are MANY users who would love you then) [18:22] <EricAhn> michaelni : Yeah, I'm developer, [18:23] <michaelni> good and you want to try to maintain or work on ffserver ? [18:23] <EricAhn> michaelni : ^^ ffserver hm, It's awesome [18:24] <michaelni> great \o/ [18:24] <michaelni> send a patch that adds you to MAINTAINERS [18:24] <durandal_1707> michaelni: dont be silly [18:24] <EricAhn> ^^; [18:24] <EricAhn> MAINTAINERS? [18:24] <michaelni> the file in git [18:26] <durandal_1707> EricAhn: are you from uk? [18:27] <EricAhn> Who should i send the patch? [18:27] <michaelni> ffmpeg-devel [18:27] <EricAhn> durandal_1707 : this is south of korea [18:27] <EricAhn> ffmpeg-devel mailing list? [18:27] <durandal_1707> yup [18:28] <EricAhn> oh I got [18:28] <EricAhn> it [18:28] <saste> EricAhn, alternatively you may create a github repo [18:28] <saste> work there, and send patches for review [18:29] <saste> or ask us to pull changes [18:29] <EricAhn> saste : oh, It's good idea. [18:29] <saste> you may need to get used to the ffmpeg development workflow [18:31] <EricAhn> okay I found https://github.com/FFmpeg/FFmpeg [18:46] <wm4> "Please provide a failing command line including complete, uncut console output." [18:47] <wm4> I wish cehoyos would stop this bullshit [18:53] <Compn> wm4 : why dont you volunteer some time to clear bug reports instead of complaining how others do it ? [18:53] <wm4> Compn: it was a bug reported by me [18:54] <durandal_1707> wm4: ignore him, he have list of text he copy paste all the time [18:57] <saste> wm4, the bug should be easy to fix, do you want to try it? [18:57] <saste> basically you need to add a few missing checks and do some string bokkeeping [18:58] <saste> i'll try to have a look at the issue since i'm partially responsible for that code, but i have a lot more work to do atm [19:08] <saste> ^^ the right question is: why the options are still documented if they have been removed? [19:11] <wm4> old version? [19:12] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:b75ac7c71cc8: lavc: include timebase in avcodec string at debug level. [19:12] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:bee044d7c261: ffmpeg: copy tmcd track timebase parameters [19:12] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:9362f31b5551: movenc: Calculate fps for tmcd without intermediate step. [19:12] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:55d66b27902d: movenc: check that fps for tmcd is within encodable range. [19:25] <wm4> hm wget actually accepts both blank domain name, and domain name + ":" + port [21:12] <cone-871> ffmpeg.git 03Jason 07master:a717fa84ede9: lavu/timecode: Allow drop frame mode for 60000/1001 fps [21:12] <cone-871> ffmpeg.git 03Jason 07master:77b740ac0ac4: lavu/timecode: fix time code calculation for 60000/1001 drop frame [21:56] <michaelni> Daemon404, i see IOC failures without linenumbers (or iam blind) [22:00] <Daemon404> weird [22:01] <Daemon404> let me manually test [22:01] <Daemon404> oh rsync mess up [22:01] <Daemon404> files were missing [22:02] <Daemon404> itll be fixed next run [22:02] <Daemon404> ignore it [22:02] <michaelni> ok [22:05] <durandal_1707> michaelni: i see same gmc_mmx in libavcodec/x86/dsputil_mmx.c [22:13] <durandal_1707> michaelni: how to get rid of av_reverse warning in lavu? [22:13] <durandal_1707> where ff_reverse should be put? [22:16] <michaelni> durandal_1707, what warning ? [22:17] <durandal_1707> av_reverse in lavu/eval [22:17] <durandal_1707> well array is not really mandatory ... [22:17] <durandal_1707> i doubt it is big performance drop [22:21] <michaelni> the attribute deprecate could be droped [22:23] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:cf48b006400e: cavsdec: check for value in get_ue_code() [22:23] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:fc8e8e5bef32: h264_qpel: put cpuflags checks back. [23:18] <cone-871> ffmpeg.git 03Michael Niedermayer 07master:c10350358da5: gifdec: gif_copy_img_rect: Fix end pointer [00:00] --- Fri Jan 25 2013

1 0

ffmpeg.log.20130124
by burek 25 Jan '13

25 Jan '13

[00:24] <antonello> hi, i would like to modify the sample decoding_encoding. I want to transcode a file. this is my code http://pastebin.com/23xbZ3gV [00:25] <antonello> the step of code are 1) decode file in raw format , 2) convert the sample format 3) encoding in final format [00:26] <antonello> at this time the conversion sample_format fail [00:26] <antonello> can you help me ? [00:32] <llogan> antonello: not many API users here. might want to try libav-user mailing list http://ffmpeg.org/mailman/listinfo/libav-user [00:32] <llogan> although durandal_1707 occasionally answers API stuff here [00:36] <antonello> ook thanks . i post my problem in mailing list..... i try chat way because tomorrow I'll have 'a meeting for my thesis but my program does not work [00:37] <llogan> then i hope you find an answer and don't fail your thesis like i did [00:37] <llogan> actually i just never finished that one [00:39] <llogan> alternatively you can wow them with a mandelbrot: ffplay -f lavfi mandelbrot [00:43] <antonello> I would like to surprise with a transcoding :P [00:51] <teratorn> antonello: "does not work" is uselessly vague [00:54] <teratorn> antonello: c= NULL; ????? [00:54] <antonello> it's vague because , the sample_format convertion fail and I haven't any error code [00:54] <teratorn> i don't see how audio_encode could *not* segfault, if you're even getting that far [00:55] <teratorn> you're overwriting your passed in `c' variable to NULL in several functions - i fail to see how this has any hope of working [00:57] <teratorn> antonello: you're using fread() manually to read an audio file... why? use ffmpeg to read audio files [00:59] <antonello> I'm modify the decoding_encode example [00:59] <teratorn> antonello: it's just a huge amout of messy code [01:01] <teratorn> antonello: you should learn about avio* functions and AVFormatContext objects [01:01] <antonello> I can't find documentation about library , I wrote this code by looking at the examples [01:02] <teratorn> antonello: the documentation is in the header files and the source code :) [01:02] <teratorn> antonello: there aren't very many good examples [01:03] <antonello> I saw :( [01:04] <antonello> do you think my code can work ? [01:05] <teratorn> antonello: sure [01:05] <teratorn> antonello: if you fix all the bugs [01:05] <teratorn> >;) [01:07] <teratorn> antonello: I would probably try making it work with data that does not require SRC first [01:08] <teratorn> e.g. raw -> aac keeping the same sample rate, same number of channels (mono!) [01:08] <teratorn> start with the simplest possible transcode and work your way up from there [01:09] <antonello> the sample encode without src is che encode example ! [01:14] <Demon_Fox> Anyone have some decent test files for testing audio? [01:43] <juanmabc> lol, mplayer "]" speed up [01:45] <llogan> Demon_Fox: testing it in what way? [01:47] <Demon_Fox> Variety of sounds for audio codec comparison at different bitrates [01:48] <Demon_Fox> Then I will write a little C program to generate a bash script for 100 test files [01:50] <llogan> http://www.rarewares.org/test_samples/ [01:50] <llogan> http://tech.ebu.ch/publications/sqamcd [01:51] <Demon_Fox> thanks [01:51] <llogan> http://wiki.hydrogenaudio.org/index.php?title=ABX [01:51] <Demon_Fox> What is a ww file? [01:52] <llogan> wavpack [01:52] <llogan> lossless audio format [01:53] <llogan> what encoder(s) are you testing? [01:59] <Demon_Fox> ffmpeg [01:59] <Demon_Fox> then [01:59] <Demon_Fox> oops [01:59] <Demon_Fox> I mean [01:59] <Demon_Fox> libvorbis [01:59] <Demon_Fox> speex [01:59] <Demon_Fox> nero aac [01:59] <Demon_Fox> faac [02:00] <Demon_Fox> lame [02:00] <Demon_Fox> ac3 [02:00] <Demon_Fox> And then all of those at 5kb/s steps from 10-95kb/s per channel [02:02] <Demon_Fox> maybe even flaac (even though it is lossless) for size comparisons [02:22] <llogan> Demon_Fox: see http://d.hatena.ne.jp/kamedo2/ [02:23] <Demon_Fox> Are you from japan? [02:23] <llogan> no, and i don't speak/read japanese. [02:23] <Demon_Fox> Also what is qaac? [02:24] <llogan> https://sites.google.com/site/qaacpage/ [02:24] <llogan> don't forget to try fdk-aac [02:25] <llogan> see https://ffmpeg.org/trac/ffmpeg/wiki/AACEncodingGuide [02:25] Action: llogan is afk for a while [02:28] <Demon_Fox> Wonder why this is the first time I have ever heard of fdk aac? [02:28] <klaxa> you don't idle in #ffmpeg enough [03:22] <Demon_Fox> Are there any places to get free instrumental? [03:23] <klaxa> http://www.jamendo.com/en/ ? [03:30] <burek> morgan.freenode.net [03:30] <burek> Morgan Freenode :) [03:31] <Demon_Fox> klaxa, Is it legal? [03:35] <llogan> Demon_Fox: did you even look at the site? [03:35] <Demon_Fox> yes [03:35] <Demon_Fox> It said for private use [03:38] <llogan> check the license for each track. all that i looked at are creative commons licenses. [03:40] <llogan> you can also license tracks if your usage does not correlate to free usage according to each license [03:56] <Demon_Fox> ok [04:47] <John__> Hi [04:48] <John__> I'm trying to record an embedded video on a webpage using ffmpeg with the command: ffmpeg -f alsa -ac 2 -i hw:0,0 -f x11grab -r 30 -s 712x366 -i :0.0+229,186 -acodec pcm_s16le -vcodec libx264 -preset ultrafast -crf 0 -threads 0 output.mkv [04:49] <John__> But it has an error: http://pastebin.com/X3N2Mvbn [04:50] <John__> I'm not sure what's the problem may be but it's having trouble recording a certain region of my computer screen. [04:51] <John__> If I have to take a guess, it would be that ffmpeg was not compiled and I'm using an older version so this function might not work properly. [04:57] <John__> It also could mean that the ffmpeg doesn't understand the coordinate or the window is not there and it can't simply record. [04:59] <John__> I've tried to mimic the size of the player embedded using another window to cover the width/height. Then I closed the window. This might cause some issue in regards recognizing the window being closed. [04:59] <John__> I haven't taken any approach on trying to debug this problem because there's little known causes base on the error it found. [05:01] <John__> An alternative would be using ffcast2, but I didn't met the requirements. [12:23] <Diogo> hi this is possible using ffmpeg hide Encoding settings from video metadata..? [12:24] <Diogo> ? [13:08] <samek_> Hi, I've got several parts of the same video, how can i "glue" them together ? I tried with multiple -i for example ffmpeg -i part01.ts -i part02.ts out.mp4 but the out.mp4 only holds first part [13:11] <gartral> hey all, I'm trying to get ffmpeg to record a game with the game A/V and my voice, i am not having any luck with this at all, I'm on Ubuntu Studio 12.10 with ffmpec/avconv 0.8.4. using the script from https://docs.google.com/file/d/0B7JCjpvY2cExVEFqX1Uxcjl3ZWM/edit along withh the insights from http://www.linux.com/learn/tutorials/367395-weekend-project-record-from-sky… any ideas? upon recording my video is grea [13:20] Action: gartral is off to make coffee while he waits for a response [13:21] <JEEB> gartral, ubuntu uses a fork of ffmpeg called libav so you'd probably want to poke #libav with it before someone starts telling this same thing to you in other words :) Also your text cut in the middle because of IRC line length limitations [13:22] <JEEB> (some people like to just ramble about it or so) [13:22] <gartral> JEEB: oh, thanks, I'll crop the post and repost here and in #libav [13:24] <durandal_1707> gartral: or deinstal libav and install ffmpeg and ask here :) [13:24] <JEEB> I wish it was as easy like that :/ [13:25] <durandal_1707> there are binaries one can install [13:25] <JEEB> yes, but you can't exactly "uninstall libav" [13:25] <gartral> durandal_1707: already explored that route, if I remove libav it wants to remove quite literally half the OS [13:25] <JEEB> unless you want to lose everything that depends on libav* [13:25] <JEEB> the libraries [13:25] <durandal_1707> just because ubuntu does not have ffmpeg package does not mean one is disallowed to use ffmpeg [13:26] <JEEB> yes, but you just said "uninstall libav", which would mean uninstalling all of the things it installs [13:26] <JEEB> which means a lot of libraires [13:26] <JEEB> which are depended upon [13:26] <JEEB> libav-tools package might be remove'able [13:26] <JEEB> as it contains avconv/the old ffmpeg/avprobe/avserver/avplay methinks? [13:27] <JEEB> and yes, you can just use an ffmpeg binary naturally, or build one yourself [13:27] <durandal_1707> ok you are free to use static binary and leave libav to exist on your hdd [13:27] <JEEB> although to be honest we don't even know his problem yet :D [13:27] <JEEB> since his line cut off [13:27] <gartral> lol [13:28] <gartral> hang on, I'm working it.. irssi doesn't like editing posts [13:28] <JEEB> yeah, irssi is known for just either sending a long message and it gets truncated on the server side, or it truncates it itself without telling to you [13:29] <gartral> x.x [13:30] <gartral> my issue is that I'm trying to record game AV and my mic, using a script from https://docs.google.com/file/d/0B7JCjpvY2cExVEFqX1Uxcjl3ZWM/edit and insight from http://www.linux.com/learn/tutorials/367395 [13:31] <JEEB> and the actual issue is..? [13:31] <JEEB> as in terminal log and so forth? [13:31] <gartral> that my sound is non-existant. [13:32] <gartral> and here's the terminal output http://paste.ubuntu.com/1565915/ [13:36] <JEEB> you should use the avconv command with libav [13:36] <JEEB> the ffmpeg binary in there was left to rot [13:36] <JEEB> when elenril rewrote parts of it [13:36] <gartral> how much of the script am i going to have to change? [13:37] <JEEB> no idea, but try just switching from 'ffmpeg' to 'avconv' first of all [13:37] <JEEB> and you'd have to make the other changes in any case because since ffmpeg's parts were rewritten those updates are in both ffmpeg's ffmpeg and libav's avconv [13:38] <gryffus> what are the legal problems about implementing ffmpeg in commercial HTPC built on mythtv please? I live in middle europe and couldn't find any information about it on google [13:39] <JEEB> from ffmpeg's side the only problem is the software license, which is either LGPL or GPL depending on which you pick during configuration when compiling [13:40] <JEEB> you'd have to abide the software license that you need depending on the configuration you need/select [13:43] <gryffus> JEEB: i am curious more about the codec licences... [13:43] <JEEB> that is why I said 'from ffmpeg's side" [13:44] <gryffus> so we can't be sued that we implemented ffmpeg in our commercial player? I was told that i need to buy licences for codecs... [13:44] <JEEB> ugh, don't misread me. Also IANAL [13:45] <gryffus> oh, i see... [13:45] <JEEB> ffmpeg only has a software license. Distributing binaries that implement possible things that might need to be licensed to distribute is a whole separate issue [13:45] <gryffus> JEEB: ok, that's i wanted to know. Let's do more searching [13:45] <gryffus> thanks :) [13:46] <JEEB> check with a lawyer if you're really worried [13:46] <JEEB> that's all I can say [13:47] <JEEB> also Dolby does not license open source, and as far as I know neither does DTS [13:47] <JEEB> so getting a license for formats by those people might be a hard task :P [13:47] <gryffus> JEEB: here in czech republic we have not many lawyers specialised on software/IP licensing [14:22] <fling> how to join 2 mono waves into single stereo flac? [14:22] <fling> I have left.wav and right.wav and I need stereo.flac :p [14:23] <JEEB> there was a filter for that [14:23] <JEEB> if I recall correctly [14:24] <saste> fling, amix/apad [14:25] <fling> saste: why apad? [14:26] <saste> fling, because i was confused [14:26] <saste> i meant "amerge" [14:27] Action: fling is reading [14:30] <fling> saste: is this right? > ffmpeg -i "amovie=left.wav [l] ; amovie=right.wav [r] ; [l] [r] amerge" stereo.flac [14:30] <saste> fling, looks so [14:30] <fling> great! thanks [14:30] <saste> uhm wait [14:31] Action: fling is waiting [14:31] <saste> you should either use filter_complex, or use -f lavfi [14:31] <saste> the first option seems simpler [14:31] <Diogo> hi this is possible using ffmpeg hide Encoding settings from video metadata..? [14:32] <Diogo> this string : Encoding settings : cabac=0 / ref=5 / deblock=1:0:0 / analyse=0x1:0x111 / me=umh / subme=8 / psy=1 / psy_rd=1.00:0.00 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=0 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=-2 / threads=1 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / bluray_compat=0 / constrained_intra=0 / bframes=0 / weightp=0 / keyint=250 / keyint_min=50 / scenecut=40 / intra_r [14:32] <Diogo> efresh=0 / rc_lookahead=50 / rc=2pass / mbtree=1 / bitrate=200 / ratetol=1.0 / qcomp=0.60 / qpmin=0 / qpmax=69 / qpstep=4 / cplxblur=20.0 / qblur=0.5 / vbv_maxrate=300 / vbv_bufsize=300 / nal_hrd=none / ip_ratio=1.40 / aq=1:1.00 [14:32] <fling> saste: ? ffmpeg -filter_complex -i "amovie & [14:32] <fling> Diogo: pipe output to second ffmpeg [14:32] <saste> fling, ffmpeg -i left.wav -i right.wav -filter_complex "[0:a][1:a]amerge" OUT [14:33] <fling> saste: oh! thanks :D [14:33] <Diogo> fling: i don't understand? [14:33] <Diogo> pipe output? [14:33] <saste> or with -f lavfi: ffmpeg -f lavfi -i "amovie=left.wav [l] ; amovie=right.wav [r] ; [l] [r] amerge" OUT [14:34] <JEEB> Diogo, no [14:35] <JEEB> also I still don't understand people who want to remove those settings [14:35] <relaxed> it's their secret sauce! [14:35] <fling> saste: Unrecognized option 'filter_complex' [14:35] <fling> saste: do I need a configure option or upgrade? [14:36] <saste> fling, upgrade I think [14:36] <fling> saste: I have 0.10.3 [14:36] <fling> ok [14:36] <saste> audio filters are recent addition [14:36] <fling> to 1.1.1? [14:36] <relaxed> latest git version is always recommended. [14:37] <fling> relaxed: ok [14:37] <saste> >= 0.11 should be fine [14:37] <fling> is it just always stable in master? [14:37] <JEEB> ffmpeg is never stable [14:37] <fling> hehe [14:37] <JEEB> most multimedia projects are never stable [14:38] <durandal_1707> dead multimedia projects are stable [14:38] <fling> durandal_1707: which ones? [14:38] <saste> JEEB: because they don't want to make people know which components are using for the streaming [14:38] <JEEB> saste, d'uh [14:39] <JEEB> if you're worried about that you have problems with something else :P [14:39] <JEEB> not to mention it'd only tell them that libx264 was used somewhere along the road [14:39] <rajeev> is latest ffmpeg comes with multiple threading enabled for libx264 or not? [14:39] <JEEB> if you have built both with threading enabled, yes [14:40] <JEEB> libx264 by default has threading and so does ffmpeg's libx264.c encoder [14:40] <LexSfX> hello. i'm trying to do this: ( ffmsindex.exe "Wow-64 2013-01-22 22-12-35-69.avi" ) where "Wow-64 2013-01-22 22-12-35-69.avi" is a 402GB file. according to Process Explorer, disk I/O seems to be running consistently >100MB/s, and has been running for a couple hours now. however, the ffmsindex progress output in the command console is still 0%. will this ever reach 1%? according to my [14:40] <LexSfX> calculations, 1% of the file (~4GB) should be done every ~40 seconds. [14:41] <rajeev> am using ffmpeg for my trans coding and see some beard behavior, If with the previous versions of ffmpeg and libx264 i do transcoding it takes x FPS to complete it. when with the latest ffmpeg which has lot of assembler modifications should improve my trans coding time together with FPS. I did see my FPS goes about 30% better compared to the previous trancode but Throughput which is (MBPS per CPU percentage) goes higher by 45%. which is [14:41] <rajeev> Could someone suggest me the reason behind it? Looking forward for the response. He is my ffmpeg command line with new and old ffmpeg+libx264 [14:41] <JEEB> LexSfX, that's a problem for ffms2/ffmpegsource project rather than here :) [14:41] <rajeev> #ffmpeg -y -i 3G1PFLuTrgM.flv -vcodec libx264 -coder 0 -flags +loop -cmp +chroma -me_method epzs -subq 2 -me_range 16 -bf 0 -refs 2 -trellis 0 -x264opts no-mixed-refs:no-8x8dct:no-mbtree:weightp=0:b-adapt=0:no-deblock=0:partitions=i8x8,i4x4:no-weightb:ipratio=1.41 -threads 1 -wpredp 0 -rc_lookahead 0 -g 250 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -b_strategy 1 -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -crf 28 /tmp/test1.flv& [14:42] <LexSfX> JEEB: where should i ask that, specifically? sorry [14:42] <rajeev> frame= 5085 fps= 38 q=26.0 Lsize=16394kB time=00:03:32.08 bitrate= 633.2kbits/s video:16294kB audio:0kB global headers:0kB muxing overhead 0.612624% [libx264 @ 0x24ceb40] frame I:147 Avg QP:24.76 size: 8813 [libx264 @ 0x24ceb40] frame P:4938 Avg QP:30.46 size: 3116 [libx264 @ 0x24ceb40] mb I I16..4: 58.7% 0.0% 41.3% [14:42] <rajeev> Results with Old ffmpeg+libx264 trancode (10 concurrent Trancode per session) [14:42] <rajeev> frame= 5085 fps= 32 q=26.0 Lsize= 16140kB time=00:03:32.08 bitrate= 623.4kbits/s video:16040kB audio:0kB global headers:0kB muxing overhead 0.622325% [libx264 @ 0x3297b40] frame I:147 Avg QP:24.76 size: 8813 [libx264 @ 0x3297b40] frame P:4938 Avg QP:30.59 size: 3064 [libx264 @ 0x3297b40] mb I I16..4: 58.7% 0.0% 41.3% [14:45] <rajeev> ok give me a sec [14:47] <rajeev> can u see it? [14:47] <durandal_1707> :) no [14:47] <durandal_1707> post link [14:47] <rajeev> i pasted it at www.pastie.org [14:47] <durandal_1707> post link to it [14:48] <rajeev> http://pastie.org/5847181 [14:48] <rajeev> this is output of single transcode [14:50] <rajeev> can u see now? [14:56] <rajeev> @fflogger can you see now? [14:56] <rajeev> any suggestions please [14:57] <rajeev> when i try to grab threads i could see encoding thread is using maximum cpu but there are some other threads those are taking lot of cpu [14:57] <rajeev> as well [14:57] <rajeev> i started looking at the jstack trace and couldn't find those threads [14:58] <rajeev> in stack trace [15:01] <rajeev> any suggestions @ <JEEB> [15:31] <EricAhn> did ffserver support from HLS(means that using ffmpeg input) to rtp(H264/AAC)? (I asked "ffmpeg-devel" channel, I think do correct in "ffmpeg" channel ,) [16:02] <Diogo> hi i don't know the difference between -keyint_min 50 and -movflags faststart ?? [16:02] <Diogo> key_int but information about the video every 5 seconds right? [16:52] <LukaszF> Hello, can anybody help me? I have some RAW AAC frame(s) and have trouble to decode it. [16:52] <LukaszF> I mean by calling avcodec_decode_audio4 from libav [16:53] <LukaszF> Frame itself was captured from IP Camera AAC 128kbps, 2 channels, 44100kHz [16:54] <LukaszF> thats all i know, after this i have some bitstream with tag aac adnd approx size of 400bytes [16:55] <LukaszF> no matter what i do return value is -1 and got_frame_ptr from avcodec_decode_audio4 is 0... [16:55] <LukaszF> any idea how to understand what is going wrong? [16:57] <LukaszF> developing on windows with libavcodec 54. 77.100 / 54. 77.100 [16:57] <LukaszF> and using AV_CODEC_ID_AAC and avcodec_find_decoder [16:58] <durandal_1707> you dont get any error message ? [16:58] <LukaszF> actualy i have [16:58] <LukaszF> [aac @ 0ef3f640] Number of bands (11) exceeds limit (3). [16:59] <durandal_1707> does same sample works with ffmpeg? [16:59] <LukaszF> you mean with ffmpegcmd application? [17:00] <durandal_1707> yes [17:00] <LukaszF> dont know how to check it? [17:02] <durandal_1707> i dont have sample... [17:02] <LukaszF> i can provide you with one frame, from that camera [17:02] <durandal_1707> LukaszF: can other app read it? [17:02] <LukaszF> actualy i have recorded it to separete files [17:03] <durandal_1707> each frame in separate file or? [17:03] <LukaszF> it is RAW frame, i am not even able to tell what it really is [17:03] <LukaszF> it isn;t contained in any container [17:04] <LukaszF> all i know that we have set camera to capture sound in aac format, and from our server we have stream of such frames [17:04] <LukaszF> beginning with four random bytes, and AAC tag after it [17:05] <EricAhn> what does "AAC with no global headers is currently not supported." mean in ffserver? [17:05] <durandal_1707> that it is not supported [17:06] <durandal_1707> LukaszF: upload one such frame [17:06] <durandal_1707> and give me link [17:06] <LukaszF> okey, give me a sec [17:08] <LukaszF> should i paste link here? [17:16] <durandal_1707> LukaszF: looks like it have some kind of extra header [17:16] <LukaszF> if i remove this header can it be decoded? [17:17] <durandal_1707> try skipping first 8 bytes [17:17] <LukaszF> ok [17:27] <LukaszF> now i got strange result [17:27] <LukaszF> error string [17:27] <LukaszF> [aac @ 0e95f620] Reserved bit set. [17:27] <LukaszF> but what is more strange return value from decode function is very big negative value [17:28] <LukaszF> like -1094995529... [17:29] <LukaszF> i am was trying with and without adding FF_INPUT_BUFFER_PADDING_SIZE to allocated buffer size for data [17:29] <LukaszF> i wonnder if it is important to add FF_INPUT_BUFFER_PADDING_SIZE to data buffer... [17:30] <LukaszF> any way result is the same [17:33] <durandal_1707> for decoding single frame? [17:34] <LukaszF> yes [17:34] <durandal_1707> just keep skipping +4 bytes [17:34] <LukaszF> hm ok i try that [17:34] <durandal_1707> or +1 [17:34] <LukaszF> hm ok [17:34] <shahriman> guys I have been trying to set psy-trellis for libx264 encoder from command line, but the specified format, i.e. -x264opts ...:psy-rd="1.5:0.2":... is not working [17:35] <LukaszF> actualy just tried -4 ;) [17:35] <shahriman> i.e. ffmpeg fails to set the option [17:35] <durandal_1707> and error is? [17:35] <shahriman> from code it looks like it's just a string that's passed to libx264, but ffmpeg considers that ":" to be a param seperator [17:36] <shahriman> [libx264 @ 0x15da960] bad option '0.2': '1' [17:36] <durandal_1707> and you cant escape it [17:36] <brontosaurusrex> shahriman: whats your exact command line? [17:36] <shahriman> i tried escaping it [17:36] <shahriman> doesn't help [17:36] <durandal_1707> it could be regression [17:37] <durandal_1707> saste: ^ [17:37] <shahriman> ffmpeg -y -copyts -i input -pix_fmt yuv420p -map 0:0 -vcodec libx264 -b:v 2810000 -preset medium -x264opts min-keyint=24:keyint=72:vbv-bufsize=12500:vbv-maxrate=3750:qpmin=5:psy-rd="1.5:0.2":aq-strength=1.5 -level 3.1 -profile:v high -pass 2 -passlogfile passlog -r 2997/125 -vframes 5000 -f mov output.mov [17:38] <beastd> shahriman: :psy-rd="1.5\:0.2": <- not working? [17:38] <shahriman> no [17:38] <EricAhn> durandal_1707 : did not implemented RTSP(H.264, AAC) in ffserver? [17:38] <beastd> i mean with both the "" and the \ inside [17:38] <shahriman> beastd: hey man how are you? remember me? [17:38] <shahriman> beastd: doesn't help [17:38] <beastd> shahriman: yes, i do! long time no see :) [17:39] <shahriman> was there a time that this was working? [17:39] <saste> shahriman, it's a known issue, and we have a workaround for that [17:39] <shahriman> Sashmo: what's that? [17:39] <shahriman> saste: ^ [17:40] <saste> shahriman, In filter and psy-rd options that use ":" as a separator themselves, use "," instead. They accept it as well since long ago but this is kept undocumented for some reason. [17:41] <shahriman> ok, trying [17:41] <saste> this is from the libx264 docs in ffmpeg-codecs(1) [17:45] <brontosaurusrex> hm, iam not even using -x264opts ... [17:45] <shahriman> saste: awesome. works. thanks a lot. [17:45] <shahriman> that's why you are brontosaurusrex [17:45] <shahriman> modernize of go extinct :P [17:46] <brontosaurusrex> no, no, its a double - fake animal, you missed the point completely [17:46] <beastd> saste: i am sure i am missing something but i would have expected the string is parsed using av_get_token [17:47] <saste> beastd, i raised the point when the patch was published on ML [17:47] <saste> but then for whatever reason michaelni preferred to push that variant [17:47] <beastd> saste: must be some reason [17:47] <saste> note that libav is working on a similar option right now (but - of course - with a different name) [17:47] <saste> that will make use of escaping [17:47] <beastd> av_get_token escaping or sth else? [17:48] <saste> beastd, yes [17:49] <brontosaurusrex> actually core x264 presets was my idea, so without me you would still be writing 7 line long silly command lines [17:50] <brontosaurusrex> so .... [17:54] <brontosaurusrex> :P [17:54] <stephanedev> hi everybody. i have been studying the ffmpeg API examples and i don't really understand how frame->pts is computed. my understanding was that frame->pts is in codec_ctx->time_base unit, but in the example it is rescaled using the stream time base. why is that? [17:56] <stephanedev> i am talking about the muxing.c example (http://ffmpeg.org/doxygen/trunk/doc_2examples_2muxing_8c-example.html) [17:58] <stephanedev> the line is: frame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base); [17:58] <durandal_1707> EricAhn: rtsp should be supported.... [17:59] <stephanedev> if the intended frame rate is 25 fps, i thought setting the codec_ctx->time_base to {1, 25} and incrementing frame->pts by 1 for every frame was the way to go, but it seems it's not [18:01] <stephanedev> shouldn't the timestamp rescaling occur on the packet timestamp before writing it to the stream? [18:02] <LukaszF> durandal_1707: I have skipped 16 first bytes [18:03] <LukaszF> after that i have return value of avcodec_decode_audio4 == 1...sigh that means 1 byte does it? [18:03] <LukaszF> any way got_frame_ptr is still 0.... [18:03] <LukaszF> i lost hope to decode this... [18:06] <durandal_1707> hmm, ask camera guys how is aac saved.. [18:06] <LukaszF> already done it, but may wait a while for answer [18:06] <durandal_1707> EricAhn: there is no example on google how to stream rtsp with ffmpeg? [18:07] <EricAhn> duradal_1707 : there is no perfect sample. [18:07] <LukaszF> thanks for your time [18:08] <LukaszF> but maybe i can do one more thing, waht could be command for ffmepg to extract aac stream from mpeg4 container? [18:08] <EricAhn> durandal_1707 : referenced http://ffmpeg.org/trac/ffmpeg/wiki/StreamingGuide [18:08] <LukaszF> because our cameras can record to ordinary container, maybe i could compare those outputs [18:09] <durandal_1707> good idea [18:10] <durandal_1707> -vn -c copy -f u8 file could save just "raw" frames [18:10] <LukaszF> ok i try that [18:12] <LukaszF> wait file in your command means output or input file? [18:16] <durandal_1707> output [18:17] <durandal_1707> -i input -vn ... [18:17] <LukaszF> ye got it [18:18] <LukaszF> yes it hasn't got any aac tags or such... [18:18] <LukaszF> is it possible to write single frame? [18:19] <durandal_1707> add -frames 1 [18:19] <LukaszF> perfect thanks [18:20] <LukaszF> now i can test my decoding function maybe i fucked up something else [18:22] <EricAhn> durandal_1707 : It's my ffmpeg "./ffmpeg -i http://qthttp.akamai.com.edgesuite.net/iphone_demo/Video_Content/npr/blindp… -vcodec copy -acodec copy http://10.10.250.122:8090/feed1.ffm" [18:30] <LukaszF> Strange i am unable to decode those aac frame extracted form mp4 container. [18:30] <LukaszF> If i understand right aac frames ARE in fact independent ( no like frames in mpeg4 )? [18:31] <LukaszF> And it should be possible to decode just one? [18:31] <LukaszF> or am i in woods totally? [18:38] <durandal_1707> LukaszF: even first frame? [18:38] <durandal_1707> have you checked with other player that mp4 file plays fine? [18:39] <LukaszF> mp4 file works just fine in media player classic [18:39] <LukaszF> after that i extracted whole aac stream, and it works in winamp [18:40] <LukaszF> I thought that aac frames even first one are the same [18:40] <durandal_1707> you also need parser iirc [18:41] <LukaszF> So those frames are not actually binary packed sound data? [18:41] <LukaszF> but has some meta information? [18:42] <durandal_1707> parser is for spliting packets [18:43] <durandal_1707> but maybe in your case frame are just in some weird container [18:46] <LukaszF> no, it came from ordinary mpeg4 container [18:46] <LukaszF> i have something like mp4Explorer from codeplex [18:46] <LukaszF> and it recognize 2 trak just fine [18:47] <LukaszF> one for video and one for audio [18:47] <LukaszF> i Don actualy get that idea with parser [18:48] <LukaszF> If i got one frame from ffmpeg called on extracted aac stream, i shouldn't need any parser? [18:55] <durandal_1707> you should be able to decode aac from mp4 using lavf to demux acc from mp4 and lavc to decode aac into pcm [19:00] <LukaszF> I was wondering, what is difference between ffmpeg - i test.mp4 -vn -c copy -f u8 test.aac [19:00] <LukaszF> and [19:01] <LukaszF> ffmpeg -i test.mp4 -vn -acodec copy test.aac [19:01] <LukaszF> because result are strange i both case i have playable aac file [19:02] <LukaszF> but first one is on some default options and is recognized as mono 128kbps [19:02] <LukaszF> and second one is playing normally as should in stereo and such [19:02] <durandal_1707> the last have actual header at start. the first whan in just extracted frames [19:03] <LukaszF> ow i thought so [19:03] <durandal_1707> -f u8 is actualy muxer for raw unsigned 8byte pcm [19:05] <LukaszF> so in first case in test.aac there are pcm frames? [19:05] <durandal_1707> no [19:05] <durandal_1707> aac frames are copied [19:19] <LukaszF> got damn [19:19] <LukaszF> i miscalculated some bytes [19:19] <LukaszF> just decoded that stupid frame from ffmpeg extracted stream [19:20] <LukaszF> now i have such question [19:20] <LukaszF> that decoded frame was from ffmpeg -i test.mp4 -vn -acodec copy -frames 1 test.aac [19:20] <LukaszF> wich means that i had necessary data to initialize ffmeg structures [19:21] <LukaszF> i suspect that those raw frames from camera may not have such information as in ffmpeg - i test.mp4 -vn -c copy -f u8 -frames 1 test.aac , what can i do then... [19:22] <durandal_1707> guess them? dunno because i dont have camera here..... [19:23] <LukaszF> ok thank you for your help, i try to tinker with it until i receive some information from camera manufacturer [19:34] <Guest17813> is the current ffmpeg is better CPU intensive compared to old version? [19:53] Last message repeated 1 time(s). [21:00] <stephanedev> could anyone explain why in the ffmpeg muxing.c example (http://ffmpeg.org/doxygen/trunk/doc_2examples_2muxing_8c-example.html) the frame pts is scaled from codec time base to stream time base (i.e. before going through the encoder)? i thought this conversion should happen for the packet pts before writing it to the stream. [21:13] <teratorn> stephanedev: yes, i believe usually you are correct [21:14] <stephanedev> teratorn: thank you. do you have any idea why it's done differently in muxing.c? [21:14] <teratorn> stephanedev: i only see one call to av_rescale_q() in there, and it's not converting the timestamps [21:14] <teratorn> its incrementing [21:14] <teratorn> by a constant [21:14] <stephanedev> a constant (1) scaled from codec time base to stream time base [21:14] <teratorn> correct [21:15] <stephanedev> why is it scaled at the frame level (before encoding) instead of at the packet level (after encoding and before muxing)? [21:16] <Mavrik> because encoder can buffer frames and reorder them when creating B-frames [21:16] <teratorn> the pts aren't being scaled, they're being incremented by a constant [21:16] <Mavrik> so you really want to pass the correct frame PTS to the encoder [21:16] <teratorn> equivalent to 1 codec time unit [21:16] <Mavrik> so it returns you proper DTS/PTS pair [21:17] <teratorn> stephanedev: so im unsure if that example is correct or not. it produces good output for you? [21:17] <stephanedev> thank you both teratorn and Mavrik for your help. i am still not sure to understand though [21:18] <stephanedev> the line is: rame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base); [21:18] <stephanedev> if i understand correctly, it scales the frame pts from codec time base to stream time base [21:19] <stephanedev> why would the encoder care about the stream time base? [21:19] <teratorn> no, it doesn't scale the pts, it increments by a constant [21:19] <stephanedev> doesn't it use only the codec time base? [21:19] <teratorn> i don't know what is hard to understand about this [21:19] <Mavrik> encoder doesn't care about time base [21:19] <Mavrik> whoever passes the frames to encoder and gets them back from it does though [21:19] <teratorn> pts = av_rescale_pts(pts, codec_time_base, stream_time_base) would scale the pts [21:19] <Mavrik> instead of focusing on that line [21:20] <Mavrik> rather figure out WHY is that even done [21:20] <Mavrik> it's not the rescaling that's the key [21:20] <stephanedev> sorry if i'm unclear, i will try to explain more [21:20] <teratorn> you're not unclear, you're simply wrong [21:20] <teratorn> :) [21:20] <teratorn> adding a constant isn't the same as scaling according to a ratio [21:20] <teratorn> ok? [21:21] <stephanedev> but why not adding 1? why does it have to be dependent on the stream time base? [21:21] <teratorn> because the pts values are copied on to the pkt, and will be correct for the stream they are going to be written to, I guess [21:21] <teratorn> normally I scale pkt pts values after encoding, but I guess that is not a requirement (?) [21:22] <teratorn> according to what Mavrik is saying [21:22] <Mavrik> nope [21:22] <Mavrik> the encoder doesn't really do anything with PTS [21:22] <Mavrik> except to remember them [21:22] <stephanedev> that's how i thought it works too, i.e. the call to rescale is done on the packet [21:22] <Mavrik> so it can correctly set PTS/DTS [21:23] <stephanedev> ok i see [21:23] <Mavrik> I usually scale before encoding though, less messy [21:23] <Mavrik> also, I only have to scale PTS and not bother with DTS (since frames before encode don't have DTS) [21:23] <stephanedev> let me run my test again, but it seems there is a difference in the generated video though [21:23] <Mavrik> matter or preference :) [21:23] <stephanedev> yes it's simpler [21:25] <stephanedev> i found a difference in the video size depending on where i do the scaling, but i may have done something wrong, i will try again [21:26] <stephanedev> in case you leave before i am back, i just want to thank you very much for your help and explanations Mavrik and teratorn [21:26] <teratorn> stephanedev: np :) [21:27] <teratorn> timestamps are a bitch [21:27] <Mavrik> yeah :\ [21:38] <stephanedev> i ran a simple test program and still have a difference in the generated video depending on where i do the scaling. both videos play, but the one with scaling on frame pts is 24 MB whereas the one with scaling on packet.pts and packet.dts is 1.67 MB [21:40] <stephanedev> the test consists in loading a set of images (frames from the big buck bunny video) and encoding them at 25 fps with H264 and writing them to mkv file [21:42] <stephanedev> the first test configuration is similar to muxing.c: frame pts initialized to zero and incremented by av_rescale_q(1, videostream->codec->time_base, videostream->time_base); for every frame [21:44] <stephanedev> second test configuration is based on the same code as first configuration. the 2 differences are that frame pts is incremented by one for every frame and packet pts and dts are scaled before writing to the file as follows: packet.pts = av_rescale_q(packet.pts, codec_context->time_base, videostream->time_base); [21:46] <stephanedev> do you have any clue as to why the generated videos are different and especially why one is so much bigger than the other? [21:47] <Mavrik> that looks like something goes very wrong with encoding [21:47] <Mavrik> either your quality settings are wierd [21:47] <Mavrik> or encoder is dropping your frames [21:48] <stephanedev> Mavrik: thank you. i didn't set any codec specific option [21:49] <stephanedev> there are 804 frames at 25fps, and both generated videos are 32s long [21:49] <Mavrik> did you actually LOOK at them? [21:50] <stephanedev> yes, they play fine [21:50] <stephanedev> both with vlc and ffplay [21:50] <Mavrik> quality? [21:51] <stephanedev> the quality is similar as far as i can tell [21:51] <stephanedev> i don't see any obvious difference [21:51] <Mavrik> then obviously you're either looking at wrong file [21:51] <Mavrik> writing to wrong file [21:52] <Mavrik> or looking at source file [21:52] <Mavrik> or not writing output at all [21:52] <Mavrik> :P [21:52] <stephanedev> i checked the file timestamps [21:52] <Mavrik> having two files with 10x size difference [21:52] <stephanedev> yeah, that's weird [21:52] <Mavrik> with "similar quality" is rather impossible [21:52] <llogan> maybe it would be more obvious if you show the console output of "ffmpeg -i input1 -i input2" [21:53] <llogan> (via pastebin.com or similar) [21:53] <stephanedev> llogan: thanks, i'll do that right away [21:57] <stephanedev> everything is the same but the bitrate: 436 kb/s for the small file, 6266 kb/s for the large file. but in both test configurations, i set the codec context bit rate to 400 kb/s [21:58] <stephanedev> so it looks like scaling the frame pts to the stream time base before encoding somehow affects the bitrate? [22:02] <stephanedev> in case you want to check the full output of the command you suggested: http://pastebin.com/beM1hvX2 [22:09] <stephanedev> i tried by using vp8 codec instead of h264 and something similar happened: one video is ~1 MB, the other one is ~7 MB [22:12] <stephanedev> again, ffmpeg -i shows that the bitrate is different: 423 kb/s and 1883 kb/s [22:12] <stephanedev> in the code, codec context bitrate is set to 400000 [23:19] <stephanedev> i tried on linux and i get the same behavior. i wrote a simple test program to reproduce the issue [23:20] <stephanedev> it loads images from a directory and generates a video. a command line option allows to choose whether to scale the frame timestamp or the packet timestamp to the stream time base [23:59] <stephanedev> just to summarize my issue/misunderstanding/confusion/(error?): i get different video files (notably the size) depending on what the time base conversion is applied to (frame pts or packet pts). if anyone wants to reproduce the issue, here is the code i used (sorry, it is not really clean): http://pastebin.com/PQ1aUdgf [00:00] --- Fri Jan 25 2013

1 0

ffmpeg-devel.log.20130123
by burek 24 Jan '13

24 Jan '13

[00:07] <saste> or.. i could extend if/ifnot to take an additional param... [00:28] <Darkarnium> win 2 [00:29] <Darkarnium> Whoops [04:54] <highgod> Hi,I want to ask a question. I get the mpeg2 video format(4:2:0 or 4:2:2) in the extra data. Is there any condition that the mpeg2 doesn't has the extra data? If under this condition, how can I get the video format? Thanks all. [05:13] <michaelni> highgod, the extradata is set from the parser or avformat, a user application might use avcodec without these or they might both fail to set extradata [05:13] <michaelni> in that case it would be in some frame (and data before that frame could not be decoded) [05:20] <highgod> Thanks, michaelni, I submit the dxva2 patch, and I want to get the video format before decode start. If the video is not 4:2:0, I will stop the dxva2 process. I have get the format from extra data reference the mpeg2video decoder that read the extra data. in AVCodecContext. But the latest hendrik reply said that I should check the mpeg2video without the extra data.I don't know how to do [05:20] <highgod> it. Thanks [05:29] <wm4> I don't know too much about video formats, but I always thought you can't know anything before inspecting the first video packet [05:29] <wm4> and even then, later packets may reconfigure the stream completely [05:29] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:e49c2aab80cb: truemotion2: clear the token array if its initialization fails. [05:29] <cone-106> ffmpeg.git 03Michael Niedermayer 07master:713dea584b11: truemotion2: use av_mallocz() [05:40] <highgod> wm4:I inspect the extra data.I just wang to get the format.Is there anyway to do this process?Thanks. [05:42] <Darkarnium> Hey all, just a quick question regarding command line arguments to demuxers / muxers. It looks like it's all handled via the options 'field' of an AVClass for the (de)muxer. Is that correct? [05:49] <cone-106> ffmpeg.git 03Xi Wang 07master:2f014567cfd6: rtpenc: fix overflow checking in avc_mp4_find_startcode() [05:49] <cone-106> ffmpeg.git 03Xi Wang 07master:902cfe2f74d7: rtmp: fix multiple broken overflow checks [05:49] <cone-106> ffmpeg.git 03Xi Wang 07master:4e692374f796: rtmp: fix buffer overflows in ff_amf_tag_contents() [10:54] <durandal_1707> uh, why LJPG in riff have two different codecs? [12:23] <cone-16> ffmpeg.git 03Paul B Mahol 07master:92d874cb787d: flacenc: include correct header [12:23] <cone-16> ffmpeg.git 03Paul B Mahol 07master:6f465c6219b6: dcaenc: remove unused header [12:24] <cone-16> ffmpeg.git 03Paul B Mahol 07master:be38aaba4a8e: ffv1enc: include correct header [12:24] <cone-16> ffmpeg.git 03Paul B Mahol 07master:a2dd988bf83b: jpeglsenc: add missing put_bits.h header [12:24] <cone-16> ffmpeg.git 03Paul B Mahol 07master:205b2d2a9822: adcpmenc: remove unused header [12:39] <durandal_1707> lol why ffplay still does not support -af, wasnt there some patch for this? [13:07] <Compn> durandal_1707 : iirc theres jpegls and ljpeg [13:08] <Compn> different lossless jpeg formats [13:08] <Compn> but i dont know if its correct in riff, i didnt look [13:59] <cone-16> ffmpeg.git 03Ronald S. Bultje 07master:55aa03b9f8f1: floatdsp: move vector_fmul_add from dsputil to avfloatdsp. [13:59] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:b1b870fbd718: Merge commit '55aa03b9f8f11ebb7535424cc0e5635558590f49' [14:11] <cone-16> ffmpeg.git 03Ronald S. Bultje 07master:42d324694883: floatdsp: move vector_fmul_reverse from dsputil to avfloatdsp. [14:11] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:6e6e1708984e: Merge commit '42d324694883cdf1fff1612ac70fa403692a1ad4' [14:22] <cone-16> ffmpeg.git 03Ronald S. Bultje 07master:5959bfaca396: floatdsp: move butterflies_float from dsputil to avfloatdsp. [14:22] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:24604ebaf85b: Merge commit '5959bfaca396ecaf63a8123055f499688b79cae3' [14:36] <cone-16> ffmpeg.git 03Ronald S. Bultje 07master:d56668bd8007: floatdsp: move scalarproduct_float from dsputil to avfloatdsp. [14:36] <cone-16> ffmpeg.git 03Diego Biurrun 07master:73b704ac609d: arm: Add some missing header #includes [14:36] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:8102f27b5b3d: Merge commit '73b704ac609d83e0be124589f24efd9b94947cf9' [15:59] <cone-16> ffmpeg.git 03Tim Walker 07master:99ccd2ba10ea: mlp: store the channel layout for each substream. [15:59] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:fa36270c4c68: Merge commit '99ccd2ba10eac2b282c272ad9e75f082123c765a' [16:11] <cone-16> ffmpeg.git 03Tim Walker 07master:3ffcccb4fbaa: mlpdec: TrueHD: use Libav channel order. [16:11] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:cd6a8618b1f5: Merge commit '3ffcccb4fbaae4d5ad775506f1f2761f2029affa' [16:34] <cone-16> ffmpeg.git 03Tim Walker 07master:1fd2deedcc64: mlpdec: set the channel layout. [16:34] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:1cf6f6f3daa9: Merge commit '1fd2deedcc6400e08b31566a547a5fac3b38cefb' [16:41] <cone-16> ffmpeg.git 03Tim Walker 07master:b0c7e3ffd09f: mlp_parser: account for AVCodecContext.request_channels when setting the channel layout. [16:41] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:86b892af199d: Merge commit 'b0c7e3ffd09f68022429920da483ad7abed84aa1' [16:49] <cone-16> ffmpeg.git 03Tim Walker 07master:ed1b01131e66: mlp: implement support for AVCodecContext.request_channel_layout. [16:49] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:826188d27acd: Merge commit 'ed1b01131e662c9086b27aaaea69684d8575fbea' [16:57] <cone-16> ffmpeg.git 03Tim Walker 07master:40cf1add12fa: mlp_parser: cosmetics: re-indent. [16:57] <cone-16> ffmpeg.git 03Ronald S. Bultje 07master:0ee8293a77a6: vp3dsp: don't do aligned reads on input. [16:57] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:0ff21c316534: mlpdec: remove disabled code, leftover from merge [16:57] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:516f30ed2082: Merge commit '0ee8293a77a6afad161e91ce1d43c4a57ce33a6a' [17:04] <cone-16> ffmpeg.git 03Ronald S. Bultje 07master:2e4bb99f4df7: vorbisdsp: convert x86 simd functions from inline asm to yasm. [17:04] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:b90ab2b99365: Merge commit '2e4bb99f4df7052b3e147ee898fcb4013a34d904' [17:15] <cone-16> ffmpeg.git 03Paul B Mahol 07master:a793a587df05: lavfi/swapuv: support all planar yuv pixel formats [17:25] <antonello> hallo , I am doing an audio transcode , Can help me ? [17:28] <durandal_1707> antonello: join #ffmpeg and ask there [17:29] <antonello> ok txt [17:36] <durandal_1707> michaelni: sox have bunch of biquad filters, i want to port to lavfi [17:37] <durandal_1707> question is should i put filters in separate files? , probably not [17:38] <durandal_1707> all filters actually use same "biquad function" with differen parameters... [17:39] <michaelni> if they differ just in parameters then same file [17:39] <michaelni> unless the code stays very similar (then multiple could make sense for easier updating) [17:40] <durandal_1707> well i do not want to use structure in function as that makes is slower [17:48] <cone-16> ffmpeg.git 03Diego Biurrun 07master:88bd7fdc821a: Drop DCTELEM typedef [17:48] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:ac8987591f55: Merge commit '88bd7fdc821aaa0cbcf44cf075c62aaa42121e3f' [17:57] <cone-16> ffmpeg.git 03Ronald S. Bultje 07master:32ff6432284f: dsputil: remove avg_no_rnd_pixels8. [17:57] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:6d1e9d993a55: Merge commit '32ff6432284f713e9f837ee5b36fc8e9f1902836' [18:01] <durandal_1707> does somebody know important reasons of current dsputil split surge? [18:02] <wm4> apparently it's a big clusterfuck [18:03] <durandal_1707> so only cosmetics? [18:04] <durandal_1707> yes it is really big clusterfuck.... [18:05] <cone-16> ffmpeg.git 03Ronald S. Bultje 07master:2bef1a83d7a3: dsputil: remove some never-assigned function pointers from the struct. [18:05] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:70270f27538f: Merge commit '2bef1a83d7a3ca60b161ab90bf4d4338e5def5e7' [18:08] <wm4> durandal_1707: I think they want to decouple decoders from what they perceive as big interlocked clusterfuck [18:08] <wm4> and then delete the clusterfuck [18:09] <wm4> but I don't know if it's more than just a cleanup [18:09] <durandal_1707> it should provide smaller binary if you disable most of code [18:11] <cone-16> ffmpeg.git 03Ronald S. Bultje 07master:2612c4dc78e1: dsputil: remove 9/10 bits hpel functions. [18:11] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:f900fd5d31dc: Merge commit '2612c4dc78e1750c4653bf5a9f2cbe95f7b4ed6e' [18:17] <saste> michaelni, about the image2 updatefirst option, I cannot understand what's useful for [18:22] <michaelni> saste, if you have a webserver and serve a image from it, with update first a ffmpeg running on the server could update that image [18:22] <michaelni> IIRC [18:25] <saste> michaelni, i don't think i understand the code, or the code in img2enc.c is wrong [18:25] <saste> why should av_get_frame_filename() fail? [18:25] <cone-16> ffmpeg.git 03Ronald S. Bultje 07master:baf35bb4bc4f: dsputil: remove one array dimension from avg_no_rnd_pixels_tab. [18:25] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:1e7a92f219a8: Merge commit 'baf35bb4bc4fe7a2a4113c50989d11dd9ef81e76' [18:25] <saste> also in that case we have no guarantee that "filename" contains a valid string [18:27] <michaelni> it fails because theres no %d in it [18:27] <michaelni> when you update always the same file, you dont need/want a %d [18:28] <saste> michaelni, i see [18:28] <saste> thanks [18:29] <cone-16> ffmpeg.git 03Martin Storsjö 07master:2e4130037cf3: float_dsp: Include config.h for redefining restrict [18:29] <cone-16> ffmpeg.git 03Xi Wang 07master:3cff53369acd: rtmp: fix multiple broken overflow checks [18:29] <cone-16> ffmpeg.git 03Xi Wang 07master:ecb918e5f0a4: rtmp: fix buffer overflows in ff_amf_tag_contents() [18:29] <cone-16> ffmpeg.git 03Xi Wang 07master:cf29f49d8ae0: rtpenc: fix overflow checking in avc_mp4_find_startcode() [18:30] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:8b8ac78ef3a1: Merge remote-tracking branch 'qatar/master' [18:33] <cone-16> ffmpeg.git 03Stefano Sabatini 07master:ca68acd7ac37: lavf/segment: mark #EXT-X-ENDLIST in M3U8 only at the end of stream [18:33] <cone-16> ffmpeg.git 03Stefano Sabatini 07master:f4e4e5b84162: lavf/segment: drop now pointless segment_list_close() function [18:56] <cone-16> ffmpeg.git 03Stefano Sabatini 07master:bb539eb13ab0: lavf/segment: avoid failure in case -flags +live and -segment_times/frames are specified [18:56] <cone-16> ffmpeg.git 03Stefano Sabatini 07master:e3197257171c: lavf/segment: exit immediately in case of invalid stream specifier [19:23] <durandal_1707> michaelni: aselect filter that select samples based on time interval is exactly what i'm looking for, i just want that timestamps are keept intact [19:30] <cone-16> ffmpeg.git 03Micah Galizia 07master:d4e5d8d4c264: lavf/hls: broker HTTP options [19:35] <durandal_1707> no wonder nobody cares for lavfi when even creators do not understand it [19:35] <saste> durandal_1707, what's the matter? [19:36] <wm4> lol [19:36] <durandal_1707> i want to silence portion of audio [19:36] <durandal_1707> using aselect and volume for example [19:36] <durandal_1707> other parts of audio should be intact [19:37] <saste> durandal_1707, in short, you can't [19:37] <Compn> cant mute ? [19:37] <saste> durandal_1707, adding support to dynamic expression in volume should be easy [19:37] <durandal_1707> so some combination of aselect,avolume,abuffer,asplit [19:37] <Compn> like mplayer's edit decision list ? :P [19:37] <saste> then you can script stuff .e.g. with sendcmd (or a complicated expression) [19:37] <Compn> " edit decision list " [19:37] <durandal_1707> saste: adding extra option for every filter is boring task [19:38] <wm4> Compn: that's quite a special case [19:38] <Compn> you want to be able to insert and remove filters at will [19:38] <durandal_1707> you mean i wate 30 minutes to send cmd? [19:38] <Compn> wm4 : what? edl ? what about filtering a logo that only appears on screen for 5 minutes every 15 minutes ? [19:39] <durandal_1707> adding extra option in each filter that will parse command is overkill [19:39] <wm4> well, edl allows you to mute - but AFAIK that's a hardcoded special case, and you can't do anything else in edls [19:39] <Compn> you can skip video in edl [19:39] <saste> durandal_1707, it's not that hard if you get used [19:39] <Compn> seek that is [19:39] <Compn> seek and mute [19:39] <durandal_1707> saste: it is wrong.... [19:40] <Compn> a very simple edl system. but other edl applications have more options. this is just mplayer limitation.... [19:40] <wm4> I think you need to provide a stable and simple interface to lavfi [19:40] <wm4> so at least _users_ can use it [19:40] <durandal_1707> it is not hard to clusterfuck on big pile of hack .. [19:41] <saste> durandal_1707, you add more options, or you add more filters [19:41] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:6747b0be9b2a: mlp: fix channel order. [19:41] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:8071288601cc: mlpdec: move ch_assign setup code down to after ch layout is set. [19:41] <Compn> there was another use case for turning filters on at random, but i've forgotten it now. [19:41] <wm4> Compn: well, in theory you could design an EDL system that allows execution arbitrary slave mode commands, and that's as far as mplayer could get, probably [19:41] <saste> sometimes it is easier to add options, or filters, depending on various constraints (efficiency, ease of use, etc.) [19:41] <wm4> personally, I think a lua shell for lavfi would solve this [19:42] <wm4> you could do things like slicing audio with Lua functions, and explicitly filter them, then concat the result [19:42] <saste> durandal_1707, for example i completely forget that i wanted to implement audio fading through expression evaluation [19:42] <durandal_1707> saste: so can you explain what new filter would do? [19:42] <saste> (using volume) [19:42] <durandal_1707> saste: good, because it is bad idea [19:42] <michaelni> the correct solution is to have a filter that can enable or disable another [19:42] <michaelni> if thats what is wanted [19:43] <saste> michaelni, how? [19:43] <Compn> -af spigot:volume [19:43] <wm4> sounds like a clusterfuck solution [19:43] <Compn> lol [19:43] <wm4> I mean, filter disabling other filters? [19:43] <saste> wm4, can you help with implementing a lavfi lua scripting filter? [19:43] <durandal_1707> there is asplit- it clones [19:44] <durandal_1707> there is aselect - it select frames/samples [19:44] <Compn> saste : would you commit lua shell if he did ? [19:44] <wm4> saste: actually I'm somewhat interested, but I have other things to do, and it'd probably break in a month due to API changes [19:44] <Compn> wm4 is here to troll, not to develop ;P [19:44] <wm4> Compn is here to troll, not to develop ;P [19:45] <Compn> nice comeback , your mom! [19:45] <michaelni> like -vf switch(volume=0,ts>5,ts<9) [19:45] <wm4> Compn: no u [19:45] <gnafu> "So's your face." [19:45] <michaelni> but the whole lavfi parser is flawed, it cant even parse anything but flat syntax [19:45] <saste> durandal_1707, asplit[in1][in2]; [in1]aselect=EPXR[out1];[in2]aselect=!EXPR[out2]; [out1][out2]ainterleave [19:46] <durandal_1707> i dont want ainterleave i want acat [19:46] <michaelni> saste, good example why its not useable [19:46] <michaelni> i mean the syntax [19:46] <saste> durandal_1707, or you extend *select to send the samples to another "else" output [19:47] <saste> michaelni, about flat syntax -> not true [19:47] <saste> it's a mess because escaping, but nothing i can't do about [19:47] <michaelni> yes [19:47] <saste> we can still design an alternative language (for file scripting) [19:47] <michaelni> its flat or not useable [19:47] <michaelni> yes [19:47] <michaelni> we should [19:47] <michaelni> also doesnt need to be just for files [19:47] <wm4> no, you should not design another language [19:47] <saste> about the switch thing: we (almost) already have it, it's the sendcmd [19:48] <michaelni> wm4, language is a bad term here [19:48] <wm4> the commands stuff is even worse, because it allows only flat strings, and unless with graph parsing, everyone is absolutely forced to use it [19:48] <saste> alternatively: we provide scripting, and the user builds the filtergraph using the wrapping language [19:49] <durandal_1707> atime=30,40,60,80[a1][a2][a3][a4][a5],][a1]afade[out0],.... [19:49] <saste> all solutions imply someone doing work, of course [19:49] <michaelni> i just want filter(foo(bar=5)) instead of filter=\\\foo\\\\\:\\\\;\\\\\;;;\\\\\:bar\\\\\\=\\\\\\\5\\\\\whatever [19:49] <wm4> anyway, people inventing their own powerful ad-hoc languages almost always ends in tears [19:49] <michaelni> wm4, YES [19:49] <michaelni> i want the insane syntax fixed not a new language [19:50] <durandal_1707> so use C syntax [19:50] <wm4> add clang as ffmpeg dependency! [19:50] <durandal_1707> wm4: so what you propose? [19:51] <wm4> durandal_1707: if you really want to allow complex filter graphs, use an external, existing, well-proven scripting language, and make lavfi available to it [19:52] <durandal_1707> lua, now i'm not sure I'm right person for it... [19:52] <durandal_1707> what would get exported? [19:54] <wm4> I suppose you would use the scripting language to build a filter graph [19:54] <wm4> so, just that [19:56] Action: wm4 suggests looking at avisynth/vsynth [19:57] <saste> durandal_1707, also now that i think at it, afade has the same limitations of fade, that's why i wrote (and never finished) multifade [19:58] <michaelni> lua and scripting is nice and good,yet this doesnt solve the problem that a generic filter that disables/enables a filter that is given as its argument is a huge mess ATM escaping wise. [20:00] <wm4> ah, I see, even when using av_opt, filters still get a flat argument string [20:00] <wm4> so maybe export the options directly [20:00] <saste> wm4, having a consistent av_opt interface for all filters is the first step in that direction [20:01] <wm4> definitely [20:05] <durandal_1707> saste: i already said my opinion on hacking [20:07] <durandal_1707> multiselect,multivolume,multidelogo,multiyadif,multicolor,multilife and so on [20:09] <durandal_1707> when integer dsputil stuff will be moved to lavu? [20:18] <durandal_1707> the earwax filter does different thing from sox effect [20:20] <durandal_1707> and above scalarproduct comment in earwax is bogus because there is extra shift... [20:21] <durandal_1707> actually function name is just wrong [20:23] <saste> durandal_1707, earwax is broken as well, i filed a ticket [20:23] <saste> patches are very welcome... [20:26] <durandal_1707> saste: i'm aware of both statements very well and I work on stuff i understand, I also avoid clusterfuck code [20:27] <durandal_1707> i just posted small cleanup patch for earwax [20:31] <durandal_1707> now only me and michaelni are left :( [20:32] <durandal_1707> lol, user on ml list posted long filter_complex example [20:33] <wm4> durandal_1707: that's the direction it'll take [20:33] <durandal_1707> wm4: what? [20:34] <wm4> users coming up with more and more complex scripts [20:35] <durandal_1707> ahh good, i was thinking you are leaving too... [20:38] <durandal_1707> having useful scripting in lavfi is critical for its success [20:41] <Compn> ehe you guys are piling up on saste :P [20:41] <Compn> gotta help him create the scripting stuff [20:41] <Compn> its a difficult thing to create an api by yourself [20:42] <durandal_1707> nope it should be simple stupid [20:47] <llogan> has any recieved any "ffmpeg-* has been disabled due to excessive bounces" messsages? [20:47] <llogan> someone is complaining in ffmpeg-user and is "sure" that it isn't his fault. [20:48] <llogan> i blamed him, but i wanted to double check [20:57] <durandal_1707> llogan: i think i did not get first mail in thread [20:58] <durandal_1707> well not that thread but some other ..... [20:59] <llogan> i was going to say if it is from "reindl" then you're just missing a troll [21:03] <Compn> llogan : any reply from ubuntu about ffmpeg package ? or is it settled now ? [21:03] <Compn> llogan : you saw siretart's explanation ? [21:09] <llogan> Compn: yes. i saw his explanation. as i was unaware of upstream activity at the time i dropped my previous plan. beastd is working it it right now, and i have another plan if that doesn't go through. [21:09] <llogan> *so i dropped... [22:21] <Compn> llogan : ok , good. [22:50] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:665f2d432ccd: hls: add missing checks for accessing avoption fields [22:50] <cone-16> ffmpeg.git 03Michael Niedermayer 07master:31cd1e20bbd5: eval: add function to access the current "wallclock" time. [23:54] <saste> michaelni: ping on eval docu patches [00:00] --- Thu Jan 24 2013

1 0

ffmpeg.log.20130123
by burek 24 Jan '13

24 Jan '13

[00:04] <dionys> thanks heaven - it works with the latest ffmpeg on my mac :-) [00:13] <dionys> in order to edit and cut some 1440x1080 MXF files in blender for youtube, I like to format them down to 1280x720. What codec can you recommend? mp4, or avi? [00:17] <teratorn> dionys: those aren't codecs :) [00:21] <dionys> sure, so mpeg4 or h264. My guess is to convert into something raw, then edit it in blender and render it to avi [00:26] <suzaru> i noticed joining crf encoded mp4 files results in vfr result. even when using "-r #" . any suggestions for maintaining cfr output? [00:41] <llogan> emc: 2004. absolutely ancient. [00:42] <llogan> just tell us what you're trying to do and we'll give you a ffmpeg command that isn't 9 years old [00:42] <emc> llogan :) & hehe, i've been playing around with different settings. [00:43] <emc> i managed to find something that worked. but thanks for your help. [00:43] <llogan> i recommend using a recent ffmpeg if you have more to encode [00:44] <emc> but maybe somebody of you knows that: i have a screencast that is recorded at 1280x960 & and i need to downsize it to 1024x760. what's the best way to do this to keep the text still sharp? [00:45] <emc> llogan: great, thanks for that! [00:45] <llogan> emc: you can use the scale filter -vf scale=1024:-1 [00:45] <llogan> and what output format are you wanting? [00:47] <emc> llogan, i want to play it on an ipad. [00:47] <emc> so probably h.264? [00:48] <emc> i will try the scale filter, thanks for the hint. much appreciated! [00:48] <llogan> damn, that's not what i wanted... [00:48] <llogan> emc: https://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide [00:49] <emc> cool, thanks llogan! [00:49] <emc> any recommendation what's the best format for a screencast? [00:49] <emc> to avoid compression artifacts etc? [00:49] <llogan> H.264 will be find for ipad. [00:49] <llogan> *fine [00:49] <llogan> use CRF as shown in guide [00:50] <llogan> i'm not sure what profile ipad can use [00:50] <emc> ok, i will try, thanks a lot! [00:51] <llogan> high profile apparently...you'll be fine [00:52] <llogan> wow the Gimps rotate is incredibly slow [01:03] <funyun> hi. does anyone know how to make ffmpeg out a log file while encoding? [01:03] <funyun> output* [01:07] <fatpony> even though i specify -r 23.976 when encoding x264, mediainfo says output mkv has a variable bitrate, is this normal? [01:07] <fatpony> s/bitrate/frame rate/ [01:11] <funyun> fatpony: -r is framerate. -b is bit rate [01:11] <fatpony> funyun: i know, 01:04 < fatpony> s/bitrate/frame rate/ [01:15] <llogan> fatpony: i trust ffmpeg more than mediainfo [01:16] <llogan> and you should use -r 24000/1001 instead of -r 23.976 [01:16] <fatpony> yeah i trust ffmpeg too but it seems strange still... i guess i'll force the bitrate in mkvmerge [01:28] <suzaru> i get vfr too with joining mp4s that are encoded as crf, used r parameter [01:29] <suzaru> avidemux has the same problem in 2.6.x. even though content is really crf. i don't think mediainfo is showing it wrong [01:30] <suzaru> i dont know how to force cbr in the container instead of vfr which ffmpeg introduces [01:30] <suzaru> cfr* [01:43] <emc> does anybody know what "illegal instruction: 4" could mean? http://pastebin.com/Mc3BWLCv [01:44] <sacarasc> What does `file ffmpeg` say? [01:44] <emc> sacarasc: ffmpeg: Mach-O 64-bit executable x86_64 [01:45] <emc> sacarasc: i guess it's the wrong binary? [01:45] <sacarasc> That would be my first guess... [01:46] <emc> sacarasc: great, thanks& will look for another one [01:49] <dunpeal> Hey. Anyone knows what this error message means? "decode_band_types: Input buffer exhausted before END element found" [02:52] <spillere> is there how to convert a rmvb into mkv? [02:53] <relaxed> did you try ffmpeg -i blah.rmvb -c copy output.mkv [03:04] <spillere> will try [03:04] <spillere> thanks [05:21] <Demon_Fox> What is a way to reduce quality without setting a bitrate? [05:22] <Demon_Fox> I know it starts with a q [05:22] <Demon_Fox> quantilizer? [05:22] <Demon_Fox> quant [05:27] <Demon_Fox> I know xvid had an option to do variable bitrate encoding more directly, but what was it? [06:31] <relaxed> Demon_Fox: maybe -q:v 2 (it works with ffmpeg's mpeg4 encoder) [06:38] <Mista_D> fresh centos install, on ffmpeg compile I get "/usr/bin/ld: cannot find -lc"... any ideas? [06:40] <Mista_D> my mistake.. "yum install glibc-static" worked. [07:18] <praveenmarkandu> hi is there anywwhere i can read up about the -threads command in ffmpeg [07:18] <praveenmarkandu> the man doesnt seem to say much [07:18] <praveenmarkandu> like what does thread 0 do? [07:19] <praveenmarkandu> or if i have 4 transcodes going at one time, how can i make each one go to its separate thread [07:21] <relaxed> praveenmarkandu: use -threads 1 for each [07:21] <praveenmarkandu> relaxed, it will then automatically assign a free thread to each one ? [07:21] <relaxed> 0 is auto when using libx264 (the default) [07:22] <praveenmarkandu> oh okay [08:51] <praveenmarkandu> can anyone give some basic system specs to handle 4 transcodes? [08:51] <praveenmarkandu> will an i7 sandy bridge suffice? [08:53] <spaam> it depends on what you are going to encode. [08:55] <praveenmarkandu> spaam, h264 [09:00] <praveenmarkandu> im actually down encoding like 10mbps files to 1.5mbps, 1.2mbps, 800kbps and 500kbps files [09:00] <praveenmarkandu> and i want to do it in one shot [09:00] <praveenmarkandu> looking into also xeon processors [09:01] <praveenmarkandu> company buying a hp blade server for dedicated transcoding [10:45] <plavcik> quit [11:51] <kode54> is it possible to do frame doubled yadif? [11:52] <kode54> since I thought yadif was capable of motion estimating two frames out of two fields [11:57] <untaken> with 2 pass encoding, what does the pass 1 do exactly? As it creates no file other than a log? I notice a lot of command say this rawvideo -y /dev/null... so where does the first pass actually do. I understand the second pass, as it creates the file [12:03] <JEEB> it goes through the clip to see its complexity in one way or another, also possibly decides on the positioning of IDR frames so it isn't done in the second pass to make it faster [12:04] <JEEB> bit rate based encoding pretty much needs this to be able to know how to evenly distribute bits for a leveled'ish "quality" while hitting a certain average bit rate goal [12:05] <kode54> JEEB: I take it you don't know the answer to my question [12:05] <untaken> JEEB: thanks for that, very interesting [12:05] <JEEB> kode54, yadif had a mode for one picture from one field if you look at the filter's documentation :P [12:05] <JEEB> it's not a really great bobber tho [12:06] <JEEB> untaken, do note that only modes that need to hit a certain average rate need something like this, as the rate control needs to be limited [12:06] <JEEB> stuff like x264's crf rate control mode just set a value to what you specify and encode with it through the clip [12:07] <JEEB> (unlimited rate control with rate factor set to X) [12:11] <JEEB> with x264 you can have a long as hell lookahead though, so with slower settings the true need for a second pass gets lower, but naturally 2pass > lookahead in most cases [12:30] <erle-> what can be the problem if atempo filter just does nothing? [12:30] <erle-> i use 2.0 btw, a speedup factor in range [12:35] <erle-> it just converts and does everything i tell it just tempo is ignored [12:35] <erle-> audio bitrate, sample rate etc is converted as i wanted it [12:41] <kode54> ah yeah [12:41] <kode54> seems he was using yadif=1 [12:41] <kode54> but the issue may be that his videos are not being detected as the correct field order [12:41] <kode54> which didn't seem to be an issue for two fields become one frame [12:42] <erle-> i was using [12:42] <erle-> -filter:a 'atempo=2.0' [12:42] <erle-> only mp3 audio [12:42] <erle-> no video [12:45] <erle-> i would like to speed up the mp3 [12:45] <erle-> it is a talk from a very slow professor [12:45] <erle-> i would like to keep the tone [12:45] <erle-> any suggestions what filters are available? [12:46] <erle-> speedup to 1.5x 1.7x and 2.0x [12:46] <durandal_1707> filter works fine here, you still did not show your ffmpeg command output [12:46] <durandal_1707> without it i cant help you [12:47] <durandal_1707> ffmpeg -filters shows all available filters for that version/build [13:01] <erle-> durandal_1707, that filter doesnt seem to be available [13:02] <erle-> now i made a audacity script [13:02] <erle-> thank you anyway [16:06] <Guest14279> can any one help me with ffprobe options -show_frames ? [16:06] <Guest14279> ffprobe -show_frames *.mp4 [16:06] <Guest14279> it shows unrecognized option [16:06] <Guest14279> ? [16:07] <Mavrik> just how old is your ffprobe? [16:07] <Guest14279> installed few min back [16:08] <Guest14279> currently using linux mint os [16:08] <klaxa> sounds like you installed avconv :V [16:08] <Guest14279> can u please tell me what it is ? avconv :v ? [16:08] <Mavrik> yep [16:09] <klaxa> avconv is an ffmpeg fork [16:10] <Guest14279> avprobe version 0.8.4-6:0.8.4-0ubuntu0.12.10.1, Copyright (c) 2007-2012 the Libav developers [16:11] <Guest14279> this is shown inthe sarting line of the out put [16:11] <klaxa> ask in #libav or get ffmpeg (static build or compile from source) [16:12] <klaxa> with ffprobe 1.0.1 i can confirm it to work [16:13] <Guest14279> thank u [16:37] <guest7> Hi everyone. I'm developing a multithreaded (pthreads) piece of software that uses ffmpeg (libav{format,codec} to be more precise). If avcodec_open2() has been called once, then used with avcodec_encode_video() for a while and then closed using avcodec_close() and I try to call avcodec_open2() again with the same encoder codec, I get a segfault. I lock around it as described in the documentation so there is no concurrent access. Any i [16:42] <durandal_1707> guest7: and? [16:46] <guest7> side question: avcodec_close() doesn't do anything to the codec associated with the context, does it? I'm using the same encoder codec in several different codec contexts concurrently. [16:48] <durandal_1707> avcodec_close free all buffers codec may allocated [16:48] <durandal_1707> so if you still use same context it will obviously crash [16:49] <guest7> not the same context, a different context with the same codec as the former. [16:51] <guest7> I mean, avcodec_find_encoder() just looks for the codec and returns a pointer to it so I figured it doesn't matter whether I call it seperately for every context or just use the same codec for all the contexts. [16:51] <guest7> durandal_1707: Is that correct? [16:56] <durandal_1707> guest7: there is documentation [16:56] <Umeaboy> Hi! [16:56] <Umeaboy> Would it be possible to download Premium-content with ffmpeg by somehow using the userinfo? I mean like this example: http://pastebin.com/U6q6pssd [16:57] <guest7> durandal_1707: Are you referring to anything specific regarding my question? [16:57] <durandal_1707> guest7: that particular function accpets codec is as arguments and return encoder struct with calls to init/close/encode.... [16:58] <durandal_1707> and it always return same pointer to struct if given same argument [16:58] <durandal_1707> so you do not need to call that function multiple times to get same enocder [16:58] <guest7> durandal_1707: Right, so basically what I've been saying is correct? There exists only one encoder codec object for every codec. [16:59] <guest7> durandal_1707: All right, so that should be fine then, thanks. [16:59] <durandal_1707> guest7: there is multiple encoders possible per codec [16:59] <durandal_1707> if you want specific one there is another function which searches by name [17:00] <durandal_1707> guest7: AVCodec *avcodec_find_encoder_by_name(const char *name); [17:00] <guest7> durandal_1707: Yeah I know that one, thanks for the hint though. [17:01] <guest7> durandal_1707: Anyway, my original question is the bigger problem, do you have any idea what could be happening in avcodec_open2() that causes a segfault? I'm pretty sure I clean up properly before opening the codec in a different context. [17:02] <durandal_1707> guest7: what version are you using? [17:02] <durandal_1707> that function is not thread safe, mentioned in documentation [17:03] <guest7> durandal_1707: The latest version available on Ubuntu, libavcodec53 [17:03] <guest7> durandal_1707: I'm locking around the function. [17:04] <durandal_1707> you probably should always call alloc_context [17:04] <guest7> durandal_1707: libavcodec 53.35.0 (major/minor/micro) [17:04] <guest7> durandal_1707: I'm doing that too. [17:04] <durandal_1707> that is very old [17:05] <durandal_1707> you call alloc_context before every call to open2? [17:07] <guest7> durandal_1707: No, I call avcodec_alloc_context3(), then avcodec_open2(), then I use avcodec_encode_video() for a while, then I call avcodec_close() and av_free() on the context again. This may repeat a number of times. [17:07] <guest7> The encoder codec stays the same across the different contexts though, it is opened only once using avcodec_find_encoder() at the beginning of the program. [17:08] <Umeaboy> I've perfomed this command to download from SVT Play before: ffmpeg -i "http://svtklipp-f.akamaihd.net/i/fritt/20130121/K-20130121-222819-4387/GULD…" -acodec copy -vcodec copy -absf aac_adtstoasc "Ett pucko i Phuket.mp4" [17:09] <durandal_1707> guest7: you must call alloc always before open [17:09] <Umeaboy> Would it be possible to download the content even thou it's for Premium-users? [17:10] <guest7> durandal_1707: That's what I'm doing already. [17:10] <durandal_1707> guest7: i'm lost, you said "No" first time [17:11] <quechec> hi; can I say something like `ffmpeg -i INPUT -c:a vorbis -c:b 256k -vn OUTPUT.ogg' and tell ffmpeg to set the output bitrate lower, respectively, depending on whether it makes sense regarding the input quality? [17:11] <guest7> I said: "I call avcodec_alloc_context3(), then avcodec_open2(), then I use avcodec_encode_video() for a while, then I call avcodec_close() and av_free() on the context again." [17:11] <guest7> durandal_1707: Right, that should have been a yes. Sorry to confuse you. [17:14] <durandal_1707> and it crashes on next open? [17:16] <guest7> durandal_1707: Yes. Although sometimes only the third or fourth time or so, which makes me think I could be using it wrong. But so far you would say everything look correct? [17:16] <durandal_1707> nope, i did not even looked at code .... [17:18] <durandal_1707> i mean every open2 shuld come after alloc context [17:19] <msmithng> I posted this a couple days ago, but didn't see any reply... I'll ask again and then shoot it to the list [17:19] <msmithng> is there anyway for me to cross reference a Lavf version to a git branch revision? [17:19] <guest7> durandal_1707: I'm talking about the general sequence of function calls and allocation of objects. [17:19] <guest7> durandal_1707: I do call alloc_context3 before every open2. [17:19] <durandal_1707> good [17:20] <durandal_1707> msmithng: can you elaborate? I do not know what you really mean. [17:21] <durandal_1707> guest7: well. problem may be buggy version you are using.... [17:22] <guest7> durandal_1707: valgrind tells me there's an invalid write of size 4 in avcodec_open2(). I'll try to debug that, thanks for your help and time, I appreciate it! [17:23] <msmithng> durandal_1707: sure... let's say I'm seeing Lavf56.blah.foo in my output... can I cross reference that to a particular ffmpeg version, N-41668-g564bb24 for example [17:25] <msmithng> I'm sorry if this doesn't make sense. I'm speaking directly to lavf53.6.0 [17:26] <durandal_1707> msmithng: lavf*.*.* is version number which does not map 1 to 1 with ffmpeg-version N-.... .... [17:26] <msmithng> ah, fair enough [17:26] <durandal_1707> there may be multiple versions N-.... with sampe lavfX.Y.Z number [17:26] <msmithng> that makes sense [17:27] <msmithng> I'll see if I can get the actual ffmpeg-version N-... from the client [17:27] <durandal_1707> i think last git commit is not exported in lib versions [17:27] <msmithng> anyway to confirm that? [17:28] <msmithng> if not, no worries [17:28] <durandal_1707> so you are only left with relase builds which have known git version.... [17:32] <antonello> hallo , I am doing an audio transcode , Can help me ? [17:34] <msmithng> Ah, ok. This is coming back to me. I've been removed from the community for a few months on other projects. Thanks for the reminder! [17:34] <durandal_1707> antonello: yes, can you explain what are you doing? [17:36] <guest7> durandal_1707: FYI: I just grabbed the debug symbols and found out that the segfault happens in avcodec_open2() -> encode_init() -> MPV_encode_init() -> ff_convert_matrix(). I just loaded the source files and looked at it but I don't know what's going on. Might try a more recent version first. [17:38] <antonello> i want create I audio transcoding , in this moment i want only obtain a raw file from aac and encoding in aac , after i want obtain a pcm.My actual problem is ....my output file is only noise [17:38] <antonello> Can i post my code ? [17:42] <durandal_1707> antonello: sure, use pastebin or similar [17:45] <antonello> ok this is my code http://pastebin.com/QuEw2ZrF [17:49] <durandal_1707> antonello: you dont set sample_fmt for decoder/encoder [17:50] <antonello> yes it is AV_SAMPLE_FMT_S16P [17:50] <durandal_1707> you set request_sample_fmt instead if decoder support that sample fmt it will use one, otherwise you need to check what sample format encoder/decoder is using [17:50] <durandal_1707> antonello: decoder/encoder set sample_fmt value [17:50] <durandal_1707> not you [17:51] <durandal_1707> also sample fmt tells how raw audio data you get from decoder or feed to encoder looks like [17:52] <durandal_1707> data you give to encoder must have same sample_fmt encoder expect [17:52] <durandal_1707> encoder/decoder do not make any conversion between them [17:53] <durandal_1707> so if decoder outputs SAMPLE_FMT_S16 but encoder wants SAMPLE_FMT_S16P and you give encoder exact data from decoder you will get mostly noise [17:54] <antonello> if i don't set in decoder the sample_fmt it use the AV_SAMPLE_FMT_FLTP [17:54] <durandal_1707> what decoder? [17:54] <antonello> but this format is unsupportet by encoder [17:55] <durandal_1707> swr_convert from libswresample converts between different SAMPLE_FMTs [17:56] <durandal_1707> antonello: sample_fmt should not be modified from user [18:01] <durandal_1707> does somebody know important reasons of current dsputil split surge? [18:01] <durandal_1707> lol. wrong ch [18:02] <antonello> ok , I use swr_convert for convert SAMPLE_FORMAT and after codific the new raw format , but the must I set the coding SAMPLE_FORMAT ? [18:02] <durandal_1707> you set input and output sample format as argument for swr_convert [18:03] <durandal_1707> there are examples in doc/example of source code [18:05] <antonello> I read that, but i found very difficulty combyne the encoding and decoding [18:05] <durandal_1707> what exactly is difficult? [18:06] <durandal_1707> you need extra buffer to resample input sample format to output sample format [18:06] <durandal_1707> and resampled buffer you give to encoder [18:07] <antonello> I tried but avcodec_encode_audio2 gave me an error, now I think it's for the incopatible raw format [18:10] <antonello> ora tu pensi che il mio codice possa funzionare? [18:10] <durandal_1707> antonello: paste error you get [18:10] <antonello> it's only RUN FAILED exit 1 [18:17] <Rajeeev> with the latest ffmpeg my thoughput goes down is there any known issues? [18:19] <Rajeeev> if i do 50 simultaneous trans-coding my CPU usage is low but Mbps per CPU percentage is going 30% higher? [19:11] <Holden> saste, Hello, are you there? [19:12] <saste> Holden, yes [19:12] <DelphiWorld> hey everyone [19:12] <DelphiWorld> burek: [19:12] <Holden> saste, hi... I tried to debug the problem we were talking about yesterday, I wanted to show you a pastebin, do you have 5 minutes? [19:12] <DelphiWorld> :P [19:13] <saste> Holden, show it [19:13] <DelphiWorld> saste: hey, how do i find the libx264 presets? [19:14] <Holden> saste, thanks, here it is: http://paste.ubuntu.com/1563787/ [19:14] <saste> DelphiWorld, checing in the docs ;-) [19:14] <saste> *checking [19:14] <DelphiWorld> saste: ;) [19:14] <DelphiWorld> saste: i checked it but didnt find them [19:15] <saste> DelphiWorld, sorry i have no time, check the wiki as well [19:15] <saste> -preset or something like that [19:16] <saste> Holden, no i don't know how to fix it at the moment [19:16] Action: DelphiWorld push saste ;-) [19:17] <saste> Holden, but again filing a ticket seems in order, so we don't discard it/you can track it [19:20] <Holden> saste, ok, I will file a ticket then with all the details, I was hoping I could make it work (seemed easy) but no luck :| [19:27] <DelphiWorld> saste, the issue is if i use a preset i get: File for preset 'ultrafast' not found [19:29] <DelphiWorld> saste: log: http://paste.debian.net/227914/ [19:30] <DelphiWorld> saste: configuration: http://paste.debian.net/227915/ [19:32] <saste> DelphiWorld, ffserver -> out of my league [19:32] <DelphiWorld> saste: same issue if i use ffmpeg... issue is the file not found [19:32] <saste> pb ffmpeg command and output [19:33] <DelphiWorld> saste: see... in /usr/local/share/ffmpeg/ i see libvpx-1080p50_60.ffpreset libvpx-720p50_60.ffpreset libx264-ipod640.ffpreset [19:34] <DelphiWorld> saste: what should i pb [19:34] <saste> DelphiWorld, can you read? [19:35] <DelphiWorld> saste: i can. but you just done pb, what i should pb for you [19:35] <saste> *ffmpeg* command [19:36] <DelphiWorld> saste: i'm using ffserver directly and ffserver is launching ffmpeg. ffmpeg -i http://localhost:8090/sast http://localhost:8070/sast.ffm [19:36] <DelphiWorld> first url is source, second is ffserbver to transcode [19:52] <DelphiWorld> saste: , ffmpeg log: http://paste.debian.net/227931/ [19:54] <DelphiWorld> Error while opening encoder for output stream #0:1 - maybe incorrect parameters [19:55] <saste> DelphiWorld, broken ffmpeg default settings detected [19:55] <DelphiWorld> saste: what setting, where? [19:55] <saste> you should use libx264 presets [19:55] <DelphiWorld> saste: is it used but inside of FFSERVER [19:55] <saste> the libx264 wiki encoding guide tells everything you want to know about that [19:56] <DelphiWorld> saste: so where did i get it from then ? [19:57] <saste> DelphiWorld, do your own research, i'm not an help desk [19:58] <saste> about ffserver i can't help [19:58] <DelphiWorld> thank saste [20:24] <DelphiWorld> saste: ok, you have ffmpeg compiled and runing? [20:26] <DelphiWorld> saste: ffprob http://46.165.201.28:8090/kashmir.flv [21:47] <brontosaurusrex> how would "422 hi-quality photojpeg" command line look like? [21:56] <llogan> brontosaurusrex: -pix_fmt yuvj422p -q:v 2 [21:56] <llogan> i guess [21:56] <brontosaurusrex> ok, whats that "2" ? [21:58] <llogan> qscale range for mpeg*: a linear scale of 1-31 where 31 is lowest quality [21:58] <brontosaurusrex> ok, but iam getting really low quality with this [21:59] <llogan> there was a time when qscale didn't work with jpeg, IIRC. [21:59] <brontosaurusrex> expected bitrate for 1920x1080 should be around 100 Mbits and up, getting 12 Mbits [21:59] <Mavrik_> wat [22:00] <llogan> lol [22:00] <brontosaurusrex> what? [22:00] <brontosaurusrex> using almost latest git [22:01] <llogan> you asked about a "photojpeg" which i assumed a normal still jpeg image. [22:01] <llogan> then you mention 100Mbits [22:02] <llogan> -flags +quality-enhance!!!!!!!!!!! [22:02] <brontosaurusrex> photojpeg in quicktime container [22:02] <llogan> i don't know what that is [22:02] <brontosaurusrex> me neither [22:02] <brontosaurusrex> certain internet service requests that .... [22:02] <brontosaurusrex> iam guessing [22:03] <llogan> show output of: "ffmpeg -i input.mov" if you already have one of these to see what the non-apple name is for it [22:03] <brontosaurusrex> ok [22:08] <brontosaurusrex> llogan: https://dl.dropbox.com/u/79532365/photosomething.txt [22:09] <llogan> -c:v mjpeg -pix_fmt yuvj422p -q:v 2 [22:09] <llogan> i don't think i've ever encoded with mjpeg, so i'm just guessing [22:10] <brontosaurusrex> thats exactly the cli i used, but i guess quality switch does not kick in [22:10] <llogan> does it actually look shitty? [22:10] <brontosaurusrex> yes [22:11] <llogan> maybe someone else has a better idea [22:11] <brontosaurusrex> another unrelated puzzle is how the hell did i get this 1906x1080 resolution .... [22:11] <brontosaurusrex> thanks llogan [22:11] <llogan> you could always declare an arbitrary -b:v, but i'd assume -q:v should have worked. [22:12] <brontosaurusrex> yeah, but quality based switch would be actually usefull [22:12] <llogan> yes [22:13] <brontosaurusrex> well, i guess its late, iam making one silly mistake after another, tommorow [22:18] <llogan> brontosaurusrex: seemed to work for me [22:19] <brontosaurusrex> -q:v 2? [22:19] <llogan> yes [22:19] <brontosaurusrex> hmmm [22:19] <brontosaurusrex> ok, will do more testing [22:20] <llogan> of course i didn't try the output in qiucktime [22:20] <llogan> maybe it's being retarded like usual [22:24] <brontosaurusrex> i see [22:24] <brontosaurusrex> later [00:00] --- Thu Jan 24 2013

1 0

ffmpeg-devel.log.20130122
by burek 23 Jan '13

23 Jan '13

[00:51] <cehoyos> Could a few peple with different x264 versions installed please test ticket 2165 ? [00:51] <cehoyos> The two necessary command lines and the input sample are in the ticket. [00:52] <cehoyos> (Works fine here on OpenSuse with an old x264 distributor build, fails with static Zeranoe binary on Windows.) [00:57] <cbsrobot> cehoyos: I get the same missmatch with x264 0.129.2245 bc13772 [00:57] <cbsrobot> on osx [00:57] <cbsrobot> I also get a crc of 0x92ee27a2 [00:58] <llogan> same. x264 0.129.2230 1cffe9f on Arch. [00:59] <llogan> *same as cbsrobot [01:02] <Daemon404> so im [01:02] <Daemon404> has anyone tried with x264cli [01:02] <Daemon404> this ight not even be our bug. [01:04] <llogan> there isn't anything to lazy copy n paste [01:05] <cbsrobot> llogan: format -fr / [01:09] <Compn> cehoyos : can you review the dxva2 patch on ffmpeg-devel ? [01:09] <Compn> or just commit it :) [01:15] <cbsrobot> using: x264 --qp 0 --preset fast --output-csp i422 --output lossless_h264_encoding.mkv lossless_h264_encoding.avi [01:16] <cbsrobot> and then: ffmpeg -i lossless_h264_encoding.mkv -pix_fmt uyvy422 -f framecrc - [01:16] <cbsrobot> I get the correct crc [01:18] <cbsrobot> when using x264 with "i444", then ffmpeg segfaults as mentionned in https://ffmpeg.org/trac/ffmpeg/ticket/2156 [01:30] <cehoyos> cbsrobot: Why are you using a different command line than I tested? [01:31] <cbsrobot> cehoyos: first I used the same [01:31] <cbsrobot> then I tested with x264cli [01:31] <Daemon404> um [01:32] <Daemon404> i thought qp 0 was not a valid way to do lossless [01:32] <Daemon404> because some qp scales go lower than 0 [01:32] <Daemon404> (e.g. 10 bit) [01:32] <J_Darnley> No, that is crf [01:33] <JEEBsv> yeah, qp 0 is valid, crf goes to negative [01:34] <Daemon404> oh [01:34] <Daemon404> woops [01:38] <cehoyos> So it fails for everybody with 129 ? [01:44] <cehoyos> Compn: I don't know anything about dxca2 (I even failed to fix the vdpau regression), nevcairiel offered to have a look when he has time again iiuc. [01:47] <wm4> the vdpau regression caused by the merge? [01:53] <cehoyos> Why do you think it was caused by the merge? [01:53] <wm4> see mplayer mailing list [01:54] <wm4> someone identified 41bdef1d39584727c9d as commit that breaks mpeg1/2 decoding with vdpau [01:54] <cehoyos> Iirc (and I do), the regression is caused by a commit that originated in avconv (ie it was unreviewed). [01:55] <cehoyos> It was actually aaf9d0e [01:56] <wm4> I wonder how that is possible, because the commit doesn't touch any of the existing decoders [02:07] Action: J_Darnley curses trac! He now sees why his ffmpeg did not like the "avi" file he downloaded. It starts with "<!DOCTYPE" [02:10] <michaelni> we dont have xml and html decoders ? ;( [02:16] <cehoyos> cbsrobot: While the lossless test succeeds with ffmpeg and 128 here, I am unable to get a correct result even for the first frame with x264 cli :-( [02:19] <Daemon404> have you actually compared how they differ [02:19] <Daemon404> the yuv [02:20] <llogan> J_Darnley: yeah, that is annoying. i'd rather have a normal link instead of "preview" [02:20] <Daemon404> or even read teh x264 docs to see what .128->1.29 signifies [02:21] <llogan> ...but i'm glad it isn't bugzilla. i'd blow my brains out. [02:39] <cone-805> ffmpeg.git 03Michael Niedermayer 07release/1.0:f7bf3becf679: Update for 1.0.3 [02:39] <Compn> michaelni : the text decoder should print it... maybe its set as binary file ? [02:40] <Compn> J_Darnley : you know you're supposed to be able to play the file in some other player before blaming ffmpeg :P [02:58] <cone-805> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n1.0.3': unknown revision or path not in the working tree. [02:58] <cone-805> Use '--' to separate paths from revisions [02:58] <cone-805> refs/tags/n1.0.3:HEAD: avf_showwaves: fix null pointer dereference [03:50] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:2338af0fb48e: doc: document aresample:first_pts [03:50] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:b3ab2810277d: avcodec/x86/cabac: workaround llvm 4.2.1 bug [05:41] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:0a84174f2d41: mov: prevent double free if the caller replaces the io context. [07:41] <cbsrobot> michaelni: thanks for fixing #2156 [10:57] <durandal_1707> so if you want apply filter to only part of stream, how do you do it? [11:10] <burek> michaelni, I've added the number of tests and also introduced a little timeout on mouse over, so that people would only see popup with the log content if they hover their mouse on the link for a second-two, to avoid all those popus to show all the time [12:07] <durandal_1707> michaelni: give me some introduction for implementing f_conditinal which does separate filtergraph bug 1062 [12:51] <wm4> lol wanting to support wav files without wav header... [13:25] <juanmabc> man, i must say that seeing the channel from time to time, the project seems to be improving quite at a good rate, keep it going, dudes [13:25] Action: juanmabc cheers up [13:30] <durandal_1707> you mean fflogger spam? [13:41] <juanmabc> durandal_1707: correction, colorful spam [13:41] <michaelni> durandal_1707, about f_cond, i think it only has to create a filter from its name & args and then pass all calls to it and [13:42] <michaelni> do some pass through for filter_frame / req_frame when its not active [13:43] <michaelni> burek, didnt look yet (id have to refind the link) but thanks! [13:44] <cone-731> ffmpeg.git 03Jindrich Makovicka 07master:570a4a018994: avidec: use sensible error codes instead of -1 [13:44] <cone-731> ffmpeg.git 03Michael Niedermayer 07master:b4a7f4d018aa: Merge commit '570a4a0189946c2c983da41d37fdd67fa13266e7' [13:53] <durandal_1707> michaelni: i more loking in something like acut filter which cut audio at certains points and apply separate filter to each pieace and resulting output is of same length, like linear editing [13:54] <durandal_1707> instead of spliting it manually and then applying filters to each componenet and cating it all together again [13:55] <cone-731> ffmpeg.git 03Diego Biurrun 07master:4af5310d2937: get_bits/put_bits: K&R formatting cosmetics [13:55] <cone-731> ffmpeg.git 03Michael Niedermayer 07master:766f05520124: Merge commit '4af5310d29379283553bcd9f541a3f6c317f706e' [14:17] <cone-731> ffmpeg.git 03Luca Barbato 07master:9146e872c643: Revert "fate: Use wmv2 IDCT for wmv2 tests" [14:17] <cone-731> ffmpeg.git 03Michael Niedermayer 07master:eb2f499e37ee: Merge commit '9146e872c643fb8f20f9043f1b93a3cddfad15c6' [14:47] <cone-731> ffmpeg.git 03Martin Storsjö 07master:ab587f39b220: rtpenc: Start the sequence numbers from a random offset [14:47] <cone-731> ffmpeg.git 03Diego Biurrun 07master:4f56e773fe8a: x86: ac3: Fix HAVE_MMXEXT condition to only refer to external assembly [14:47] <cone-731> ffmpeg.git 03Michael Niedermayer 07master:73b59cc1bad8: Merge commit '4f56e773fe8a554b8c2662650aaf799c2ece2721' [14:47] <cone-731> ffmpeg.git 03Michael Niedermayer 07master:9774251c45ea: rtpenc: do not use random values for seq when bitexact is requested. [15:47] <cone-731> ffmpeg.git 03Ronald S. Bultje 07master:4a73fbd9c5d5: vp3/5: move put_no_rnd_pixels_l2 from dsputil to VP3DSPContext. [15:47] <cone-731> ffmpeg.git 03Michael Smith 07master:0881cbf31498: proresdec: support mixed interlaced/non-interlaced content [15:47] <cone-731> ffmpeg.git 03Michael Niedermayer 07master:26345acb0e72: Merge remote-tracking branch 'qatar/master' [16:11] <durandal_1707> michaelni: so it is basically not possible to slice filtergraph input? [16:25] <michaelni> durandal_1707, iam not sure i understand what you mean by slice. But sure a filter can route different segments to different filters and the combine them [16:30] <durandal_1707> michaelni: i not interested in combining different parts yet [16:31] <durandal_1707> michaelni: i just want to split stream in several concated stream [16:32] <durandal_1707> somethink like reverse of concat... [16:40] <michaelni> durandal_1707, a filter could open a muxer like the segment muxer and write to several files, or a segment muxer could be at the end of the filter chain and a filter could indicate to it when to start new files, maybe by using a keyframe flag [16:51] <durandal_1707> michaelni: you dont understand me, i want to cut input audio stream to n pieces and aply each filter to each piece sequentially [16:52] <durandal_1707> i do not work with multiple files nor output to multiple files.... [16:52] <durandal_1707> basically i need asplit functinality except i need only part of stream not x stream clones [16:53] <durandal_1707> what is different between JPEG vs MPEG color range? [16:53] <durandal_1707> JPEG one use all bytes? [16:54] <michaelni> the difference is where the max / min points are in YCbCR [16:56] <michaelni> about the audio you can split things as you want and apply filters as you want iam pretty sure its supported but i dont understand what you mean exactly [16:56] <michaelni> you can split things temporaly or between audio channels [16:56] <michaelni> you can apply filters as wanted or pass things through [16:56] <michaelni> and you can pass things on to one or multiple output pads or store in files [16:57] <michaelni> you could also instanciate a whole filter graph or multiple ones in a filter and pass things throgh them [16:58] <durandal_1707> michaelni: so what is example for: fade in video - fade out - fade in - fade out? [16:59] <durandal_1707> michaelni: and what color range have max/min set to 0/255 ? [17:00] <michaelni> if you want multiple fades in a single video stream, simply using multiple fade filters should work [17:01] <michaelni> jpeg uses the full 0..255 range [17:05] <durandal_1707> michaelni: multiple fade filters do not work simply because after fade out everything is black [17:06] <jeje101> hi to all [17:06] <durandal_1707> adding options or new filter for this feature is stupid [17:06] <jeje101> I'm facing a problem using FFMPEG and swscale to decode and display video stream from cameras [17:07] <durandal_1707> jeje101: than ask it in #ffmpeg [17:07] <jeje101> depending the resolution of the stream, I can display correctly or not the video stream [17:08] <jeje101> in fact my problem are when I use 720*576 or 720*288 [17:08] <jeje101> the rendered is look like a stride image pb [17:09] <jeje101> the stream I want to decode is H264 [17:09] <jeje101> if someone can help me please ... [17:11] <jeje101> ok sorry It isn't the good room [17:11] <jeje101> bye [17:17] <michaelni> durandal_1707, if fade would fade back from black then multiple should be chainable [17:18] <durandal_1707> michaelni: and how would that be possible? [17:21] <michaelni> fade could be changed to support fade to and from black [17:22] <durandal_1707> fade from black is called fade in, fade to black is called fade out [17:22] <durandal_1707> adding extra option is silly [17:22] <durandal_1707> because you would need to add list parsing to each filter that does it [18:52] <durandal_1707> michaelni: can you share more about various fixes you commit? So I can be sure such fixes are not cosmetical and real bug is resolbed. [18:54] <michaelni> durandal_1707, which commits? [18:55] <durandal_1707> any commit that fix crashes/dos/CVE/invalid read/write and so on [18:55] <durandal_1707> there is FUD spread by quatar peers that your just apply bandaid [18:56] <michaelni> durandal_1707, where do they say that ? [18:56] <durandal_1707> here: http://blog.flameeyes.eu/2013/01/crashes-and-dos-what-is-it-with-them-anyway [18:58] <michaelni> if there are specific commits where someone makes a specific claim then id like to know about it so i can look into it [18:59] <michaelni> if theres a general "we dont understand ffmpeg code" so it must be wrong then well iam not sure what i could do [19:13] <burek> michaelni, it's http://ffmpeg.gusari.org/fate2/ [19:14] <burek> btw, I could send it for your review (or whoever is willing to take a look at the code) [19:14] <burek> so we can copy it to the fate server and test it [19:14] <burek> it doesn't touch/change any file that is already existing, so it's mostly read-only [19:15] <burek> it just creates some stats files, in order not to read/grep same data twice [19:15] <burek> so, I believe it's pretty much safe [19:17] <michaelni> burek, great, yes please post it, not sure i will review (its not so much my area) but iam happy to put it on the server alongside the existing fate if bcoudurier agrees and it passes review [19:18] <burek> ok, let me tar/gzip it [19:23] <michaelni> burek, you need tgz ? its so big ? [19:23] <michaelni> review is more likely if people can read it , tgz may be a problem here [19:24] <burek> well, how do you want me to post it [19:25] <burek> i was thinking like tar/gzip it and send it as attachment (it's not big) [19:25] <burek> so that anyone can just extract it to his/her webdir and test if he/she wants [19:25] <burek> but, let me know how to do it in a better way? [19:28] <durandal_1707> burek: how much it is big when not gzipped? [19:28] <burek> 72 kb (with images) [19:28] <burek> 42 kb without [19:28] <burek> the main index.php is 15 kb [19:30] <durandal_1707> michaelni: should that be compressed? [19:31] <Compn> its fine if its that small [19:31] <Compn> i think mailing list limits to 80K [19:32] <michaelni> it maybe once was 80k but while i dont remember its exact value its higher now [19:32] <durandal_1707> i think my patch was bigger than that..... [19:34] <durandal_1707> ...not it was more than 10 times smaller :) .... [19:37] <cone-731> ffmpeg.git 03James Almer 07master:e65d8509f065: libm: Add fallback definition for cbrt() using pow() [19:37] <Compn> yeah it should be higher now [19:37] <Compn> i think it was bumped to 140K and may even be 200K+ after the server was switched [19:38] <michaelni> yes 200-300 sounds similar to what i remember [19:39] <michaelni> at least for some list [19:39] <Compn> i dont remember that option being in the moderator panel of mailman so it must be in a config somewhere :P [19:40] <michaelni> your memory sucks then or your eyes ;) [19:40] <michaelni> i do remember it in the mailman web interface [19:40] <Compn> could be :) [19:41] <Compn> Maximum length in kilobytes (KB) of a message body. Use 0 for no limit [19:41] <Compn> ah yes there it is [19:41] <Compn> 400K for ffmpeg-devel [19:49] <mateo`> durandal_1707, michaelni: do you think the aiffenc/id3 patch is ok to be applied ? [19:57] <durandal_1707> i already gave my opinion multiple times [19:57] <llogan> yes, 400k for devel and 100k for user [19:58] <llogan> i increased them both since it was no longer 2004 [19:58] <Compn> haha [19:58] <Compn> llogan : 65K should be enough for anyone! :P [19:58] <durandal_1707> it is ok but i do not want to apply it because it is evil [19:59] <llogan> Compn: you would also get tired of approving >65k messages. [19:59] <llogan> (or rejecting too) [20:05] <llogan> http://pastebin.com/jwWArB4g looks like mplayerhq.cn is finally going to be a reality [20:05] <mateo`> durandal_1707: :( [20:05] <Compn> i just ignore those letters :D [20:05] <Compn> arent they basically blackmail ? [20:06] <llogan> i actually replied to one for the hell of it for a personal domain. it didn't seem blackmailesque but maybe that was the intent [20:06] <Compn> you didnt get a reply ? [20:07] <llogan> yes, two replies [20:08] <Compn> oh [20:16] <durandal_1707> michaelni: could you be so evil and apply mateo` aiff patch? [20:20] <cone-731> ffmpeg.git 03Carl Eugen Hoyos 07master:40297a8bdb69: Support solid gray frames when decoding Lagarith. [20:25] <michaelni> durandal_1707, sure [20:26] <cone-731> ffmpeg.git 03Stefano Sabatini 07master:7694533fda5d: doc/muxers: remove outdated comment about live flags for the segment muxer [20:26] <cone-731> ffmpeg.git 03Stefano Sabatini 07master:30a40d4a554c: lavf/segment: provide link to HTML M3U8 draft, also fix mentioned version in docs [20:27] <cone-731> ffmpeg.git 03Matthieu Bouron 07master:c114414a57b2: lavf/aiffenc: ID3 tags support [20:28] <cone-731> ffmpeg.git 03Carl Eugen Hoyos 07master:379d03b0c9fd: Fix compilation with --disable-everything --enable-demuxer=loas [20:36] <cone-731> ffmpeg.git 03Stefano Sabatini 07master:5b10c5e7e4d7: lavfi/showwaves: fix check on showwaves->outpicref [20:52] <cone-731> ffmpeg.git 03Paul B Mahol 07master:c4274d118b1a: lagarith: return more meaningful error codes [20:58] <burek> so, I should just attach all the web files (.js, .css, .php) without images (?) as a plain text attachment and send it to the mailing list, right? [21:01] <burek> here is a dir tree http://i47.tinypic.com/ok7tbo.png [21:05] <llogan> burek: a non-tarbomb archive with images might be useful for the lazy who might want screw with it locally [21:06] <llogan> ...as an additional attachment. [21:07] <burek> so, just to be sure, I should insert all those (text) files as inline attachments and then compress everything and add another (complete) attachment? [21:07] <burek> (btw, whole thing tar/gzipped is 30 KB) [21:08] <llogan> burek: sounds good to me [21:08] <burek> ok [21:15] <cehoyos> burek: Could you comment on ticket 1387? [21:15] <burek> sure, just let me finish this first [21:18] <llogan> burek: i'm surprised you didn't try -input_format for 1387 [21:20] <saste> burek, i recently added some logs to help in case of failure [21:45] <burek> ok, I've sent the mail to ML, let me see the ticket now [21:54] <burek> i started testing that h264 input thing only to figure out i'm not testing it with h264 web cam :) anyway in next few days i'll get the cam again and test it [21:56] <burek> saste, in case of h264 input? [21:56] <burek> llogan, i will :) thanks for the hint :) [22:00] <cone-731> ffmpeg.git 03Michael Niedermayer 07master:49b729d3af84: sanm: check image dimensions before using them [22:05] <saste> burek: in all cases [22:08] <burek> cool [22:08] <burek> in regular stderr output or some additional option is needed? [22:09] <burek> like -report or something [22:12] <saste> burek: -loglevel debug [22:12] <saste> i'd like to implement per-context logging level at some point [22:13] <durandal_1707> michaelni: how that can overflow? [22:13] <saste> like -loglevel muxer/v4l=debug [22:13] <burek> saste cool :) thanks :) [22:13] <burek> well, this is also a big improvement [22:13] <burek> since, usually, there is a lot of point of possible failure, when working with v4l... [22:14] <burek> points* [22:14] <michaelni> durandal_1707, the number of pixels and buffer sizes are calculated by w*h in various forms [22:17] <durandal_1707> michaelni: ahh, i was thinking about + [22:19] <llogan> burek: like the guy in #ffmpeg now [22:23] <burek> llogan, I see, nice :) [22:29] <durandal_1707> cehoyos: so you reply to ticket for fun? [22:30] <durandal_1707> you hardcoded aiff demuxer to use 2 and this gives correct output? [22:34] <cehoyos> I wanted to ask you the same question ;-) [22:34] <cehoyos> Iirc, I was unable to fix the error message, but if I knew how I would test if "2" works. [22:38] <durandal_1707> perhaps 16 bits per coded sample means it is using 16k bit rate [22:38] <durandal_1707> while in fact it is using 2 bits .... [22:39] <durandal_1707> terminal brain damage.... [22:44] <cehoyos> To a certain, degree that makes sense to me... [22:46] <durandal_1707> unfortunately files is so small that i hear nothing [22:49] <cehoyos> This is indeed a problem;-) [22:49] <cehoyos> The source of the sample is this page afaict: http://www-mmsp.ece.mcgill.ca/documents/AudioFormats/AIFF/Samples.html [22:51] <cehoyos> It contains another link: ftp://ftp-ccrma.stanford.edu/pub/Lisp/ I believe there is more information if you can find it... [22:51] <cone-731> ffmpeg.git 03Michael Niedermayer 07master:1bb05797ec27: sanm: remove "duplicate" for loop. [22:51] <cone-731> ffmpeg.git 03Michael Niedermayer 07master:4c9f35bb7c94: mvdec: use avpriv_set_pts_info() instead of directly setting tb. [22:55] <durandal_1707> michaelni: that sanm fix may be reason why I could never get other codecs to work properly .... [23:14] <burek> is there an input option to tell ffmpeg that input video is progressive, although inside the video file the flag has been set as if it was interlaced [23:14] <burek> something like ffmpeg -deinterlaced -i ... [23:15] <burek> "According to the production supervisor of the release, the actual video encoded is 1080P30, but since the spec doesn't support that format, it is flagged as 1080i (vc-1)." [23:15] <cehoyos> What difference should that flag make? [23:16] <burek> http://ffmpeg.gusari.org/viewtopic.php?f=12&t=766&p=1479#p1479 [23:16] <burek> wait [23:16] <burek> bad url [23:16] <burek> http://ffmpeg.gusari.org/viewtopic.php?f=16&t=801 [23:17] <cone-731> ffmpeg.git 03Michael Niedermayer 07master:e6cc3c869b92: mvdec: minor simplification, remove av_inv_q() [23:17] <cone-731> ffmpeg.git 03Michael Niedermayer 07master:b89815f5199f: mvdec: check channel count. [23:18] <cehoyos> Sorry, but the question as it is posed ("can ffmpeg play 1080p30") makes no sense at all - ffmpeg decodes 960p40 if you encode your input with 960 lines and 40fps and codec that FFmpeg supports. [23:18] <cehoyos> I suspect that the OP has a (interlaced) vc-1 stream that FFmpeg does not support. A patch is welcome, there is no simple work-around. [23:18] <cehoyos> (If he wants to provide a sample, it is welcome.) [23:19] <burek> ok, thanks :) [23:19] <cehoyos> (And yes, FFmpeg can better judge if the stream is interlaced than the production supervisor. Or in other words: VC-1 encoder typically sucks [23:20] <cehoyos> ... A VC-1 encoder typically sucks and encodes even progressive material in interlaced mode.) [23:20] <burek> great.. :) [23:21] <cehoyos> s/does not support/does not decode correctly [00:00] --- Wed Jan 23 2013

1 0

ffmpeg.log.20130122
by burek 23 Jan '13

23 Jan '13

[00:00] <MachinaeWolf> How would I use ffmpeg to just record sound that is on the system without using a mic? [00:02] <klaxa> MachinaeWolf: depends on your sound system how ffmpeg would handle that [00:03] <MachinaeWolf> I have alsa [00:03] <klaxa> i.e. if you have pulse, set the monitor of your soundcard as recording device for ffmpeg [00:03] <klaxa> hmm no idea on alsa :V [00:04] <MachinaeWolf> darn :/ [00:04] <llogan> MachinaeWolf: https://ffmpeg.org/trac/ffmpeg/wiki/Capturing%20audio%20with%20FFmpeg%20and… [00:05] <MachinaeWolf> ooh thanks llogan :) [00:05] <llogan> lots o' stuff on that wiki [00:25] <MachinaeWolf> hmm um I followed that guide and I got no sound llogan :/ [00:31] <llogan> MachinaeWolf: then you need to find out if your sound card has an audio loopback device (often called Stereo Mix/Wave out mix/Mono Mix/What U Hear) [00:31] <llogan> or use a software solution [00:32] <llogan> open alsamixer and look at what is available to record from [00:33] <MachinaeWolf> capture tab has mic boost, capture, digital, and internal mic boost [00:33] <MachinaeWolf> ^ on alsamixer [00:33] <llogan> maybe it's "capture". is this HDA Intel? [00:34] <MachinaeWolf> Yep card: HDA Intel PCH Chip: Intel CougarPoint HDMI [00:34] <MachinaeWolf> well capture was at almost 100%... [00:35] <llogan> maybe burek can help. i've actually never tried to capture system [00:35] <llogan> i lied. i did it with pulseaudio once i think [00:56] <jorge_> Hi, I'm trying to compile ffmpeg with libfaac -- I've used "--prefix=/my/dir" to compile libfaac, but now I get a "libfaac not found" when compiling ffmpeg. I've tried to setup cflags and ldfags to the faac-1.28/libfaac/without success. Any hints? [01:02] <msmithng> is there anyway for me to cross reference a Lavf version to a git branch revision? [01:03] <llogan> jorge_: not an answer, but fdk-aac is generally considered better than faac [02:21] <burek> MachinaeWolf did you solve your problem [02:41] <MachinaeWolf> nope burek [02:41] <MachinaeWolf> sorry I was doing some stuff for a while [02:49] <burek> can you type: arecord -L [02:49] <burek> and copy the output to a pastebin-like site (like www.pastebin.com) [02:50] <MachinaeWolf> sure hold on [02:51] <MachinaeWolf> http://pastebin.com/GHJcjffE [02:53] <burek> oj [02:53] <burek> ok [02:53] <burek> did you try typing alsamixer [02:53] <burek> and pressing <tab> [02:53] <burek> to switch to recording devices [02:53] <burek> also F6 to choose audio card (if you have multiple installed) [02:54] <MachinaeWolf> yeah it's capture devices [02:54] <_numbers> does ffmpeg use -preset and -vprofile or does it just pass those to x264? [02:55] <burek> can you printscreen it? [02:55] <MachinaeWolf> sure burek [02:55] <burek> _numbers, passes to libx264 [02:55] <burek> you can also use -x264opt [02:55] <MachinaeWolf> burek, http://db.tt/OKpzq9EU [02:56] <_numbers> is it better to specify things like -preset, -vprofile, -maxrate, -bufsize or can i just use --x264opt [02:56] <_numbers> with profile=high:preset=veryslow:tune=film:vbv-maxrate=364:vbv-bufsize:182 [02:56] <burek> MachinaeWolf, you have more devices on the left [02:56] <burek> _numbers it depends on what you want [02:57] <_numbers> i mean does ffmpeg use them or just pass 100% to x264 [02:57] <MachinaeWolf> No that's all of them on the capture tab burek [02:57] <burek> also, no need to touch -profile (unless you really know what you are doing), since it is automatically selected based on your image size [02:58] <burek> MachinaeWolf, you might be better of buying a cheap audio card [02:58] <burek> this one is like 5$ card [02:58] <burek> it doesn't even have standard controls [02:58] <MachinaeWolf> well sound works I just can't get ffmpeg to record it from it [02:58] <burek> well ok [02:59] <MachinaeWolf> Would pulse maybe work? Distro had alsa and pulse installed, though I prefer alsa [03:00] <burek> you can always test and see [03:00] <MachinaeWolf> k [03:04] <MachinaeWolf> Is the syntax ffmpeg -f pulse -i hw:0 filename.wav for pulse? [03:25] <_numbers> does this seem valid: ffmpeg -i hq.mp4 -ss 00:00:30.0 -t 00:00:10.0 -threads 4 -f mp4 -vcodec libx264 -acodec libfdk_aac --x264opts profile=high:preset=veryslow:tune=film:level=3.1:pass=1:crop=1,1,1,1:resize:width=679,height=288,csp=yv12,method=bicubic:fps=10:crf=18:vbv-maxrate=300:vbv-bufsize=3000 -ac 1 -ar 22050 -ab 32k v35.mp4 [03:25] <_numbers> i am passing all the video-related options i can directly thru --x264opts instead of using the ffmpeg pseudo-options [03:26] <MachinaeWolf> burek, Is the syntax ffmpeg -f pulse -i hw:0 filename.wav for pulse? [03:27] <burek> _numbers, test and see if it is [03:28] <burek> usual way to do that is: ffmpeg ... -profile:v ... -preset:v ... -tune:v ... -level ... etc. [03:28] <burek> MachinaeWolf yes [03:28] <MachinaeWolf> hmm well that didn't work ?_? [03:28] <burek> but not -i hw:0 [03:28] <burek> read the manual [03:28] <burek> hw:0 is alsa's thing [03:28] <MachinaeWolf> oh [03:29] <_numbers> [libx264 @ 0x2dd7b80] bad option 'profile': 'high' [03:29] <_numbers> i guess that is one that cannot be passed through x264opts [03:29] <burek> MachinaeWolf try ffmpeg -f alsa -i pulse [03:29] <burek> _numbers, one stupid question [03:29] <burek> why would you want to do that [03:29] <burek> if there is -profile ffmpeg option? [03:30] <_numbers> its just confusing to memorize two options by different names, when ffmpeg doesn't seem to do anything with it [03:30] <_numbers> one option by two different names* [03:30] <burek> well, x264 is usually used by experts (people that know what they are doing and why) [03:30] <burek> to set some advanced x264 options [03:31] <burek> most people need only -c:v libx264 -preset ... -crf ... output [03:31] <_numbers> ya i'm trying to stream the video tho so i need control over bitrate [03:31] <burek> what's wrong with -b:v and -maxrate [03:31] <MachinaeWolf> burek, that command worked but still no sound on the file :( [03:32] <burek> MachinaeWolf, sorry, I can't help anymore since I never use pulse [03:32] <_numbers> well i dont want CBR which is what -b:v is. i want VBR specifically CRF [03:32] <_numbers> so crf=18 is one option i must pass to x264 [03:33] <klaxa> crf 18 for streaming? [03:33] <_numbers> and then -maxrate is renamed x264 option vbv_maxrate [03:33] <klaxa> lan or...? [03:33] <_numbers> klaxa: its a good q. i havent gotten far enough to find out what crf is going to be acceptable. i guess that seems high? [03:33] <_numbers> its just over regular cable modem from aws ec2 [03:34] <klaxa> well depends on how compressed the video will get, i.e. how slow you can encode for it to still be realtime [03:34] <klaxa> if you encode with fast preset, bitrate will be high for crf 18 [03:34] <_numbers> https://sites.google.com/site/linuxencoding/x264-ffmpeg-mapping [03:35] <_numbers> i'm using veryslow. u can see my exact command above [03:35] <_numbers> i am gonig for 500k bitrate, 944x400 (non-standard) high resolution, and super-low 10fps framerate, and willing to compromise on RF [03:36] <klaxa> <_numbers> i guess that is one that cannot be passed through x264opts <-- pretty much this [03:36] <klaxa> that may work with Handbrake, but not with ffmpeg :P [03:36] <_numbers> the video is more like a 2 hour powerpoint presentation with a few <5min stock videos sprinkled inbetween [03:36] <_numbers> ya i am coming from handbrake gui [03:36] <klaxa> ffmpeg can do the resizing instead of libx264 [03:36] <_numbers> because it ignored my VBV settigs [03:36] <klaxa> same goes for cropping i think [03:37] <MachinaeWolf> ooh searched up a bit and I got it [03:37] <burek> _numbers use -crf [03:37] <klaxa> colorspace conversion will be forced automatically, fps can be set too by ffmpeg [03:37] <MachinaeWolf> records with the pulse in the command after I messed with pavcontrol [03:37] <MachinaeWolf> pavucontrol* [03:38] <burek> also [03:38] <_numbers> i'm sure almost all the options can be done by ffmpeg, but which will be better to ask for the task? ffmpeg or x264 [03:38] <burek> using -preset veryslow will not help in live stream [03:38] <burek> but, you'll figure that out sooner or later [03:38] <burek> when you start testing [03:38] <klaxa> lol MachinaeWolf: i think i said what to do a few hours ago :P [03:38] <MachinaeWolf> Hmm must've not been around then [03:38] <klaxa> oh wait [03:39] <klaxa> you said back then you wanted to use alsa [03:39] <MachinaeWolf> yeah [03:39] <klaxa> <klaxa> i.e. if you have pulse, set the monitor of your soundcard as recording device for ffmpeg [03:39] <MachinaeWolf> don't know why alsa doesn't work for ffmpeg to record it [03:39] <burek> <_numbers> i'm sure almost all the options can be done by ffmpeg, but which will be better to ask for the task? ffmpeg or x264 :) [03:39] <burek> really? [03:39] <klaxa> either specify monitor of your soundcard as default, or set it after the ffmpeg record sink has been instantiated [03:39] <_numbers> burek: what preset would you recommend for streaming? i thought it only had to do with compression/encoding effectiveness, and not really much to do with bitrate ? perhaps you mean it will raise client-side/player decoding cpu requirements? [03:40] <burek> didn't we just tell you that ffmpeg just passes those options directly to libx264? [03:40] <_numbers> burek: ok right so why do i care to give them to ffmpeg at all, except for the ones that i have to because --x264opts wont accept it there [03:41] <burek> _numbers, -preset will slow down your cpu and utilize more ram memory [03:41] <burek> if you don't have a very good cpu, then you'll have dropped frames [03:41] <_numbers> so in that case i want ultrafast probably [03:41] <burek> and bigger latency [03:41] <_numbers> or zerolatency? [03:41] <burek> also, if you use -maxrate, there is no point in using -crf [03:42] <_numbers> i've had ppl tell me crf is compatible with maxrate [03:42] <burek> there are 3 things you can optimize on, but usually 2 of them can go together at the same time: speed of encoding, image quality and output file size [03:43] <_numbers> basically 95% of my 2hr video is still frames, but a few (maybe 6) scenes (less than 5min each) are action scenes, so thats the only time i need bitrate to ramp up [03:43] <_numbers> but even then it shouldn't exceed a maximum cap of like 500kbps [03:43] <burek> hmh [03:43] <burek> how is a 2hr video a live streaming? [03:43] <_numbers> its got audio that is mostly talking [03:44] <_numbers> its educational material [03:44] <burek> if you have your material already prerecorded :) [03:44] <burek> that's not live streaming then [03:44] <_numbers> no its VOD [03:44] <_numbers> still streaming [03:44] <burek> too much confusion right here :) [03:44] <_numbers> i'm using RTMP and flash player in browser [03:44] <burek> just use 2-pass encoding then [03:44] <burek> and you'll be fine [03:44] <burek> live streaming is when you stream some live events [03:45] <burek> not a vob (or any other) file from your hdd [03:45] <_numbers> right i know. i never used 'live' word ;) [03:45] <burek> ok [03:46] <_numbers> when i say 'streaming' i mean rtmp. its NOT the pseudo-streaming mp4 with qt-faststart [03:46] <_numbers> so i can afford a veryslow profile for encoding then right? because its just one-time [03:46] <_numbers> as long as it doesn't negatively impact decoding, then its fine i think [03:48] <jbermudes> I'm trying to run: ffmpeg -y -f video4linux2 -i /dev/video0 -f jack -ac 1 -i ffmpeg_1 -f jack -ac 1 -i ffmpeg_2 -map 0 -map 1 -map 2 test.mp4 [03:48] <jbermudes> In order to end up with 2 audio tracks each with 1 channel, but I still end up with each audio track having 2 channels [03:49] <burek> _numbers, 2-pass will help you if you are targetting a specific bitrate [03:49] <_numbers> but is 2-pass valid with crf? [03:50] <burek> 1st pass will analyze the video and 2-pass will encode various parts of the video using different encoding params [03:50] <burek> no [03:50] <burek> crf will optimize the image quality [03:50] <burek> nothing to do with the bitrate [03:50] <burek> jbermudes, you need to use audio filter for that [03:50] <burek> I think it's amix or amerge [03:51] <burek> hmh [03:51] <burek> it seems that wiki has better SEO than a main website :/ [03:52] <burek> https://ffmpeg.org/ffmpeg-filters.html#amerge [03:52] <burek> jbermudes <- [03:59] <jbermudes> burek: Hmm, but if that's the case, then why do tutorials suggest -ac 1 for the simple case of converting a stereo mp3 to mono? I thought ac operaters on the input to mixdown [04:02] <burek> jbermudes which tutorials? [04:04] <jbermudes> burek: Well, like for instance: http://howto-pages.org/ffmpeg/ (Ctrl+F for 'to mono') [04:05] <jbermudes> does -ac only operate on the output file? [04:07] <burek> oh man.. [04:07] <burek> first of all [04:07] <burek> howto-pages.org is not ffmpeg.org [04:07] <burek> we've got our own tutorials on our wiki [04:08] <burek> next, you are not following that tutorial correctly [04:08] <burek> look where is your "-ac" and where is it in that tutorial [04:08] <burek> yours is before -i, making it input option [04:09] <burek> if you put it after -i, it will be considered as output option [04:09] <burek> thus, mixing it down to mono [04:10] <_numbers> is -an necessary ? [04:10] <burek> ffmpeg -y -f video4linux2 -i /dev/video0 -f jack -i ffmpeg_1 -f jack -i ffmpeg_2 -map 0 -map 1 -map 2 -ac 1 test.mp4 [04:10] <burek> try that [04:11] <burek> -an means "no audio"... it will mute your input/output [04:20] <_numbers> ffmpeg -i hq.mp4 -ss 00:00:30.0 -t 00:00:10.0 -threads 4 -f mp4 -vcodec libx264 -pass 1 -profile:v high -preset veryslow -tune film -level 3.1 -s 604x256 -r 10 -crf 18 -maxrate 300k -bufsize 3000k -acodec libfdk_aac -ac 1 -ar 22050 -ab 32k v35.mp4 [04:20] <_numbers> i tried to say audio codec should be mono, but its coming out stereo [04:21] <_numbers> also the audio freq and bitrate didnt apply either [04:23] <_numbers> do i need -vcodec copy -ac1 ? [04:24] <_numbers> hm no -ac 1 should work right [04:27] <_numbers> Stream #0:1(eng): Audio: aac ([64][0][0][0] / 0x0040), 22050 Hz, mono, s16, 32 kb/s [04:28] <_numbers> strange that vlc thinks its stereo 44hz [04:31] <_numbers> when i tell ffmpeg -ss 00:59:42.8 -t 00:00:55.3 it takes forever to reach that point. is that normal? [04:31] <_numbers> i see this for a long time: frame= 0 fps=0.0 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A [04:37] <burek> as usual [04:43] <_numbers> burek: http://www.hastebin.com/puwobihugo.sm [04:46] <_numbers> heres plain text version that doesnt get cut off http://www.hastebin.com/raw/puwobihugo [04:47] <Zeranoe1> _numbers: Audio: aac ([64][0][0][0] / 0x0040), 22050 Hz, mono, s16, 32 kb/s [04:47] <_numbers> correct thats what i pasted above [04:48] <_numbers> but vlc doesn't report that [04:48] <_numbers> is there a 2nd opinion i can get somehow? [04:48] <Zeranoe1> _numbers: Are you on Linux? [04:48] <_numbers> ya [04:48] <Zeranoe1> _numbers: mediainfo is an option, or ffprobe [04:48] <beware> can ffmpeg remux an avi and remove junk chunks? [04:49] <beware> i find that when i encode an avi with it, by default, it introduces junk chunks. and mencoder does, too [04:49] <Zeranoe1> beware: Junk chunks? [04:49] <beware> they are chunks, that are added to the begin of the file and make it bigger [04:49] <beware> in the avi standard, "junk" are chunks without meaning [04:50] <burek> why avi [04:50] <Zeranoe1> beware: When doing -c:a copy and -c:v copy are they also copied? [04:50] <beware> burek: compatibility with something, specific needs [04:52] <_numbers> Zeranoe1: ya ffprobe reports it correctly Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 22050 Hz, mono, fltp, 32 kb/s [04:52] <beware> let's see [04:52] <beware> yes they are [04:52] <beware> well, [04:52] <beware> ffmpeg omits the original junk chunks, but introduces it's own which are bigger still [04:52] <Zeranoe1> Zeranoe1: I would trust that [04:53] <beware> mencoder's are about 4 kb, ffmpeg's 6 kb [04:53] <_numbers> how can i tell ffmpeg to use the yuv420p color profile for video? [04:54] <Zeranoe1> _numbers: -pix_fmt right? [04:55] <Zeranoe1> _numbers: Available options is -pix_fmts [05:00] <_numbers> if i want to crop AND resize/scale is it important where i specify the -vf? e.g. before or after -s ? and am i cropping the input dimensions, or the output dimensions? [05:04] <llogan> why would you use -s and scale? [05:04] <llogan> one will be ignored [05:04] <_numbers> i'm only using -s [05:04] <_numbers> i assumed it meant -scale [05:05] <_numbers> the input file is a 944x400 resolution [05:05] <llogan> makes more sense to use scale if you're cropping to keep it all within a filterchain [05:05] <_numbers> and i specify size of -s 604x256 [05:05] <_numbers> ah ok . so i want to crop 1px off each side before scaling [05:06] <_numbers> http://ffmpeg.org/trac/ffmpeg/wiki/FilteringGuide [05:08] <llogan> -vf crop=iw-2:ih-2,scale.... [05:11] <_numbers> thx [05:11] <_numbers> [libx264 @ 0x245ff40] constant rate-factor is incompatible with 2pass. [05:11] <_numbers> argh [05:12] <llogan> what are you trying to do? [05:13] <_numbers> was seeing if CRF would get more efficient at staying within -maxrate [05:17] <_numbers> "CRF in streaming can be thought of as a more advanced "min QP" where the codec won't waste bits improving quality beyond a certain threshold. You'd then set the peak bitrate and VBV to whatever the max you want for your stream to be. Peak is really what matters in streaming for compatibility and support." [05:18] <_numbers> anyone understand which ffmpeg options he's referring to? [05:18] <llogan> who is "he"? [05:18] <llogan> maxrate and bufsize probably [05:20] <llogan> bufsize/maxrate = latency in seconds [05:20] <llogan> typical bufsizes are 1-5 seconds [05:21] <llogan> probably [05:21] <_numbers> he = benwaggoner from http://forum.doom9.org/archive/index.php/t-163968.html [05:21] <_numbers> i can't get this 55sec video clip to stream over rtmp without stuttering [05:23] <llogan> your maxrate should exceed your minimum connection speed [05:24] <_numbers> see here http://bit.ly/10KRXub (click in the "Filename" box then click away to trigger onblur which starts that video) [05:24] <llogan> ignore that [05:25] <llogan> your maxrate determines the minimum connection speed that you want to support [05:25] <_numbers> i have: -s 674x286 -r 8 -crf 16 -maxrate 300k -bufsize 150k -ac 1 -ar 22050 -ab 32k [05:26] <_numbers> ffprobe says the output file is Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 674x286 [SAR 703:702 DAR 236911:100386], 1327 kb/s, SAR 8437:8425 DAR 59:25, 8 fps, 8 tbr, 16384 tbn, 16 tbc [05:26] <_numbers> notice the 1.327mbps bitrate!? [05:27] <_numbers> sry disregard, i probed wrong file [05:27] <_numbers> its Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 674x286 [SAR 703:702 DAR 236911:100386], 182 kb/s, SAR 8437:8425 DAR 59:25, 8 fps, 8 tbr, 16384 tbn, 16 tbc [05:27] <_numbers> 182kbps is more realistic [05:27] <_numbers> but i cant figure out why it stutters during flash player playback [05:27] <_numbers> over rtmp [05:41] <llogan> _numbers: you can use crf in your first pass as a "test" to determine a certain quality. when it's done encoding it will tell you the average bitrate. if the average bitrate is lower than your desired limit, then you know you won't be wasting bits by declaring a certain bitrate with -b:v instead. [05:41] <llogan> if it's over, then just use -b:v [05:42] <llogan> i mean just use your max allowed bitrate with -b:v [05:42] Action: llogan is sucking at communicating tonight [05:42] <llogan> this scotch is from 1989 [05:46] <llogan> if your limit is 300k and -crf 24 is "good enough" and under 300k, then in your second pass, use whatever bitrate -crf 24 gives you instead of using -b:v 300k. [05:49] <_numbers> hm ok will try [05:51] <Fjorgynn> Good morning [06:33] <_numbers> what does that 'Format for web" option do? what is the cli equivalent? [06:33] <_numbers> sry meant for #handbrake [07:30] <_numbers> is there a way to tell ffmpeg to guess the ETA while its encoding? [07:43] <llogan> _numbers: you could watch "time=" and infer from there i guess [07:43] <_numbers> easier to watch frames since time varies by encoding settings [08:27] <kcm1700> CODEC_ID_RAWVIDEO <- what is the purpose of it? [08:28] <kcm1700> does it actually create a dummy codec? [10:41] <Abhijit> hi [10:42] <Abhijit> May I ask if I can hire someone with ffmpeg knowledge ? [11:13] <burek> Abhijit, did you check consulting page [11:13] <burek> https://ffmpeg.org/consulting.html [11:15] <Abhijit> nope. let me check. thanks for the link burek [11:31] <simonec77> Hi, it is possible to biuld FFMPEG libs (libavcodec, libavfilter and so on) in debug mode for MSVC? [11:32] <simonec77> I have already follewed instruction at:http://ffmpeg.org/platform.html#Microsoft-Visual-C_002b_002b [11:33] <simonec77> but when I put the dll, and the .lib in my MSVC project [11:33] <simonec77> and try to Degug [11:34] <simonec77> the Output Window say:"Binary was not built with debug information".. [14:36] <kode54> <DEATH> ffmpeg yadif outputs half the rate by default [14:36] <kode54> <DEATH> Only SOMEHOW MPC-HC would claim the files were still full rate when outputting Matroska [14:36] <kode54> <DEATH> But stepping frame by frame and counting revealed 25FPS [14:36] <kode54> <DEATH> And remux to MP4 made MPC-HC show correct rate [14:36] <kode54> <Case> ctrl-J should show stats that tell realtime values too [14:36] <kode54> <DEATH> So ffmpeg muxes Matroska with nonsense info when usign yadif in default mode [14:43] <relaxed> The default is one frame per frame, not per field. [14:43] <relaxed> What does mediainfo report? [14:45] <relaxed> or mkvinfo [14:47] <durandal_1707> mediainfo is banned [14:47] <relaxed> says who? [14:59] <suzaru> im trying to join 2 aac files with latest ffmpeg. the result from the cmd i use is just the first aac file [14:59] <suzaru> ffmpeg -i "concat:1_Output.m4a|2_Output.m4a" -c copy -absf aac_adtstoasc out.m4a [14:59] <suzaru> is there something else i should try? [15:00] <Mavrik> kode54: have you tried forcing fps with "-r 25" to ensure correct metadata? [15:00] <kode54> I'll tell him that [15:00] <kode54> although he would prefer that it perform frame doubling instead [15:00] <suzaru> i also tried: ffmpeg -i "concat:1_Output.m4a|2_Output.m4a" -acodec copy -absf aac_adtstoasc out.m4a [15:01] <suzaru> same result [15:02] <suzaru> any suggestions appreciated =} [15:08] <suzaru> i noticed joining crf encoded mp4 files it results in vfr result. even when using "-r #" [15:09] <suzaru> any ideas to preserve cfr frame rate mode? [15:10] <suzaru> both mp4 files had identical frame rate of 25 and constant frame rate mode. each frame was unique in each respective source [15:12] <relaxed> suzaru: demux aac from the mp4 and concat with mkvmerge [15:13] <suzaru> it's actually m4a [15:13] <suzaru> so demuxed already [15:13] <relaxed> m4a is mp4 [15:13] <relaxed> So, it's not demuxed [15:13] <suzaru> how do i demux it? [15:13] <suzaru> someone told me to try demuxing it before. he didn't have any idea how to [15:14] <suzaru> he only had mkvextract as a suggestion to demux but it doesn't work on m4a (just mkv i'm guessing) [15:14] <Mavrik> yeah, mp4 (m4a) really can't be concatenated easly [15:15] <relaxed> ffmpeg -i blah.m4a -c:a copy -bsf:a aac_adtstoasc demuxed.aac [15:15] <Mavrik> do "ffmpeg -i <your>.m4a -codec copy <your>.mkv" to remux it to mkv [15:15] <Mavrik> then concat it with mkvmerge [15:15] <Mavrik> :) [15:15] <Mavrik> you can remux it back to m4a again afterwards [15:18] <suzaru> neat. lol.. i was able to demux it to .aac. gonna install mkvmerge so i can concat [15:18] <suzaru> how to mux it back to m4a ? [15:18] <suzaru> remux* [15:18] <Mavrik> just wap parameters :) [15:18] <Mavrik> *swap [15:19] <Mavrik> ffmpeg -i demuxed.aac -c:a copy blah.m4a [15:19] <suzaru> thanks [15:23] <suzaru> out of curiousity is the mkvmerge step needed? i did concat on the .aac and i think it worked [15:24] <suzaru> run time/size looked right [15:25] <suzaru> yeah runtime is perfect [15:25] <suzaru> on the output [15:34] <suzaru> can i change Format profile : Apple audio with iTunes info to Format profile : Base Media / Version 2? [15:34] <suzaru> what it was before [15:34] <suzaru> for the new m4a [16:15] <microchip_> how can i set the quantizer used for b-frames? [17:12] <jeje632> hi to all [17:13] <jeje632> I'm facing a problem using FFMPEG avdecode_video2 and swscale to decode a H264 video stream from a camera [17:13] <jeje632> depending the resolution it work well or not [17:14] <jeje632> if I set the camera to 720*288 or 720*576 it doesn't dsiplay well [17:14] <jeje632> look like a stride problem [17:14] <jeje632> if someone can help me please [17:15] <durandal_1707> jeje840: pastebin code [17:16] <jeje840> I think the problem is just to using swscale to convert from YUV420 to RGB32 [17:16] <durandal_1707> swscale converts fine [17:17] <jeje840> after the decompression, I have my AVFrame* m_lpFrame [17:17] <jeje840> I try to set the linesize by this [17:17] Last message repeated 1 time(s). [17:17] <jeje840> m_lpFrame->linesize[0] = dwWidth+32; [17:17] <jeje840> m_lpFrame->linesize[1] = m_lpFrame->linesize[0]/2; [17:17] <jeje840> m_lpFrame->linesize[2] = m_lpFrame->linesize[0]/2; [17:18] <jeje840> where dwWidth is equal to 720 [17:18] <durandal_1707> hmm, you are supposed to modify linesize/ [17:20] <jeje840> I do this before passing to the function [17:20] <jeje840> img_convert( AVPicture* dst, PixelFormat dst_pix_fmt, int dstWidth, int dstHeight, AVPicture* src, PixelFormat src_pix_fmt, int srcWidth, int srcHeight, DWORD dwStride) [17:20] <jeje840> and in this one I do [17:20] <jeje840> if (dst->linesize[0] == 0) dst->linesize[0] = dwStride; [17:20] <durandal_1707> do you really need to set linesize, there are tools that do that for you.... [17:21] <jeje840> I don"t know what tools I need to use [17:21] <jeje840> are you speaking about ffmpeg function? [17:25] <durandal_1707> av_image_fill_linesizes [17:25] <jeje840> I use swscale to directly convert in a DirectDraw surface [17:26] <jeje840> I suppose I need to use this function instead of my manual linesize values set [17:34] <jeje840> I just have to do [17:34] <jeje840> av_image_fill_linesizes(m_lpFrame->linesize, PIX_FMT_YUV420P,dwWidth); [17:34] <jeje840> isn't it? [17:50] <jeje840> When I try the av_image_fill_linesizes with the real value of the width video entry, it doesn't work even for other resolution working before [17:50] <Zeeflo> This is when I run FFmpeg: http://pastebin.com/qvt7ZzPj [17:50] <jeje840> if I want it work like before, I'm always need to add 32 to the real width [17:50] <jeje840> and it still doesn't work with a width of 720 [17:51] <Zeeflo> And this is what happens: http://pastebin.com/etVjZWeP ? Whats going on? [17:53] <jeje840> I think there soething wrong when my video width isn't modulo 32 [17:55] <durandal_1707> jeje840: your buffers should be padded [17:55] <Mavrik> and aligned. [18:00] <jeje840> could you please check this part of code because I don't see where is the problem [18:00] <jeje840> http://pastebin.com/2cHu3fKk [18:01] <jeje840> thank you very much [18:03] <BtbN> Hi. When playing/decoding my h264 video, i get some "[h264 @ 0x7f3604c938c0] number of reference frames (0+5) exceeds max (4; probably corrupt input), discarding one" errors. [18:04] <BtbN> I'm trying to fix my encoder, but i can't realy find what i did wrong, does anyone know what might be causing this? [18:04] <BtbN> It started after i added more than one idr frame [18:04] <BtbN> before i just had exactly one idr frame in the beginning, what made the stream unseekable [18:05] <Mavrik> BtbN: that's a message from the decoder, not encoder [18:05] <Mavrik> check your input [18:05] <BtbN> Mavrik: sure it is, that what i'm ascing [18:05] <BtbN> i wrote the encoder [18:05] <BtbN> and i'd like to know what i'm doing wrong [18:06] <Mavrik> you're creating too many B-frames per GOP [18:06] <jeje840> when you say: buffer need to be padded (and aligned), you tell me this about the input buffer (before calling avdecode_video2) or the result of the decoding (the AVFrame)? [18:07] <BtbN> I don't create any B-Frames. Just I and P for the moment [18:07] <Mavrik> which level/profile are you targeting? [18:07] <BtbN> High, level 3.1 [18:10] <BtbN> http://btbn.de/files/testout.mp4 here's a sample output which has the problem. [18:11] <Mavrik> hmm [18:11] <Mavrik> it seems number of reference frames is written into the stream [18:11] <Mavrik> check h264_ps.c file in the decoder to see how it's set [18:13] <BtbN> the maximum number is written, yes. I set it to 4. Or is there something else? From my look at h264_refs.c it seems like i have to issue a "MMCO_RESET" or something similar. But no idea how [18:14] <Mavrik> why don't you trace through the code with a debugger? :) [18:16] <BtbN> cause i don't realy know what i'm looking for. I looked at the bitstream output from x264, it also uses a max_num_ref_frames value of 4, ans also has multiple idr frames. But i don't see what it does to avoid filling up the reference buffer [18:18] <jeje840> about your answer, are you telling me to add FF_INPUT_BUFFER_PADDING_SIZE to the packet data before call avcodec_decodevideo2? I try it but no change make [18:21] <yermandu> hi deers : > [19:01] <Fjorgynn> Vad är detta? [19:04] <juanmabc> :O [19:05] <Fjorgynn> Wrong channel [19:05] <Fjorgynn> it is [20:27] <blue_misfit> Hello folks - is there a way to determine the starting GOP timecode of an MPEG-2 video stored in MPEG-PS using ffmpeg? [20:27] <blue_misfit> I can get this info by demuxing and using a stream analyzer like Manzanita, but haven't been able to do it via simple ffprobe or ffmpeg -i [20:27] <saste> blue_misfit, ffmpeg -show_packets / -show_frames [20:27] <saste> ehm ffprobe [20:28] <blue_misfit> nice [20:28] <blue_misfit> hmm [20:29] <blue_misfit> show_frames seems to give me lots of useful information including pts/dts [20:29] <blue_misfit> but I'm looking specifically for the GOP timecodes [20:29] <blue_misfit> which are something different AFAIK [20:31] <saste> blue_misfit, -show_entries frame=pict_type,pts_time,pts [20:31] <saste> i don't know what you mean by "GOP timecodes" [20:34] <blue_misfit> apparently it's a "thing" involving storing standard HH:MM:SS:FF timecodes in the GOP headers, but I don't know more than that [20:35] <saste> blue_misfit, we have support for timecode display, but can't say much more than that [20:40] <JEEB> sounds like SMPTE time codes [20:40] <JEEB> in theory with the frame count and PTS you could create the smpte timecode of a frame [20:41] <blue_misfit> JEEB: sure that makes sense, but wouldn't you have to assume what the starting timecode of a given stream would be then? [20:41] <JEEB> aye [20:41] <blue_misfit> that's the problem :) [20:41] <JEEB> if you have a nonzero start point, then you'll have fun [20:41] <blue_misfit> I've got a bunch of files, some start at hour 00 [20:41] <blue_misfit> some start at hour 01 [20:41] <blue_misfit> welcome to broadcast <herpderp> [20:41] <JEEB> I guess kierank or someone who is close to broadcast should be able to answer some questions :S [20:42] <blue_misfit> Yep. I'm whining in #ffmbc [20:42] <JEEB> but I'd guess ffmpeg/ffprobe out-of-the-box aren't capable of showing those timecodes [20:42] <blue_misfit> indeed. ffmbc can do this for MOV and MXF [20:42] <blue_misfit> I'm hoping there's some super secret way to do this for MPEG-PS [20:42] <JEEB> most probably not :P [20:43] <JEEB> which means om nom nom parsing code needed [20:43] <blue_misfit> dreadful [20:44] <blue_misfit> yeah I can do this with Elecard / Manzanita, but blech [22:15] <RobertNagy> is there any difference quality wise between deinterlacing RGB or YUV, i.e. is YUV->Deinterlace->RGB == YUV->RGB->Deinterlace?e [22:16] <Holden> Hello guys, can anyone tell me why I can't seem to open my tv card with ffplay? http://paste.ubuntu.com/1560780/ [22:24] <burek> Holden, did you try: vlc v4l2:///dev/video0 [22:25] <Holden> burek, hmm, no, I do not have vlc installed atm, but I know for sure the tv card it's working (at least it works with mplayer and my own program)... I am noticing now that ffplay works with the webcam, but not with the tv card... [22:26] <burek> hm, i havent used any tv cards with v4l so far.. [22:29] <Holden> this is strange... I'm fairly sure it use to work... it can list available formats okay, but it just hangs without an error or an hint on what it might be wrong [22:31] <saste> Holden, latest version? [22:31] <Holden> saste, yep, compiled two hours ago [22:33] <saste> Holden, can you try this patch and see if it improves the feedback? [22:33] <saste> http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/157567/focus=157961 [22:33] <saste> even if i suspect the problem is unrelated [22:35] <llogan> saste: what does enabling libv4l2/v4l-utils provide that native doesn't? [22:35] <Holden> saste, sure, I can test it... [22:35] <saste> llogan, i don't know, i never understood that... [22:36] <llogan> at least you're not alone [22:36] <Holden> saste, also running ffmpeg like this http://paste.ubuntu.com/1560841/ I get a 100% cpu usage, and after pressing ctrl-c (line 33) I get all that [22:36] <saste> llogan, it's useful if you don't have libswscale I suppose [22:37] <saste> or for testing libv4l2 issues/comparing it with libswscale [22:37] <saste> but i might be totally wrong [22:38] <saste> the rfps spam is annoying... [22:38] <saste> Holden, that's the fun of v4l2, it's hard to debug/reproduce [22:41] <Holden> saste, I hear you... ok, trying your patch now... just one thing I noticed: the doc says "-channel <int> .D.... set TV channel, used only by frame grabber (from 0 to INT_MAX)" but I think 'channel' is misleading, it should be 'input', because looking at the code what it does is setting the input: input.index = s->channel; v4l2_ioctl(s->fd, VIDIOC_S_INPUT, &input.index); [22:43] <saste> Holden, "channel" (given it's a TV grab) sounds better than "input", too generic, input index would be fine too [22:49] <Holden> saste, yes, input index would be better imo, channels makes me think of a...well tv channel. ok, patch applied, nothing really changed: http://paste.ubuntu.com/1560869/ :( [22:56] <burek> llogan, v4l-utils are used at cmd line to set some "defaults" for the device or to change some parameters while capturing the video, like brightness, zoom, contrast, etc [22:56] <burek> i can only imagine that libv4l2/v4l-utils is a wrapper for that [22:59] <burek> Holden, can you type: v4l-ctl --list-formats-ext [22:59] <burek> and copy the output to a pastebin-like site (like www.pastebin.com) [23:00] <saste> burek, well libv4l is a lib, v4l-utils is an utility controlling the v4l2 device settings (which are persistent across device usage sessions) [23:00] <saste> in theory we could extend v4l2 to make it set the other options [23:00] <burek> not all of them [23:00] <Holden> burek, it does not seem to find that command http://paste.ubuntu.com/1560902/ [23:00] <saste> but since different driver may support different options, this may be difficult [23:01] <burek> Holden, apt-get install v4l-utils [23:01] <saste> Holden, and yes setting a specific input may help [23:01] <burek> saste, v4l-ctl -L gives the list of user-settable options [23:01] <saste> we recently added a few options which may affect the operation of your device [23:01] <burek> and it's in a standard format i guess [23:01] <burek> but imho it's pointless to make a wrapper around something that already works on cmd line [23:02] <burek> unless ffmpeg resets some parameters on device open [23:02] <saste> burek, what if you use a program? [23:02] <Holden> burek, I'm on a old (10.04) ubuntu install, it says it cannot find the v4l-utils package [23:02] <saste> ffmpeg is a library, amongst the various tools [23:03] <saste> Holden, you sure your kernel supports v4l2? [23:03] <burek> it should have v4l-utils [23:03] <Holden> saste, of course, as I said the tv card works ok with other programs [23:03] <saste> ok [23:05] <burek> http://packages.ubuntu.com/search?keywords=v4l-utils&searchon=names&suite=q… [23:06] <burek> does that mean v4l-utils [23:06] <burek> is not available in previous ubuntu versions? [23:06] <burek> if so, ubuntu sucks big time [23:06] <Holden> lol [23:06] <burek> anyway, try apt-cache search video4linux | grep utils [23:06] <burek> maybe it was renamed [23:07] <Holden> burek, it returns me nothing. can you show me the typical output of that command? maybe I have something equivalent [23:07] <burek> you might also try luvcview to test all your uvc compliant v4l devices [23:08] <burek> err.. try v4l2-ctl :D [23:10] <Holden> :O sudo apt-get install ivtv-utils wants to install 25 packages and 21mb :D [23:11] <burek> o.O [23:11] <burek> output should look like this: http://pastebin.com/DTevc8TM [23:14] <Holden> burek, ok, then I can show you something similar from my program: http://paste.ubuntu.com/1560916/ [23:15] <burek> set your input to one of the numbers 1-3 [23:16] <Holden> burek, that's what I did... it does not work [23:18] <burek> try this: ffmpeg -loglevel debug -f v4l2 -i /dev/video0 test.mpg [23:18] <burek> and copy the output to a pastebin-like site (like www.pastebin.com) [23:21] <Holden> burek, just a second, I have changed the code and recompiled, maybe I have found the problem [23:21] <burek> ok [23:23] <Holden> great, I think I know what it does not work now... saste, you still around? [23:24] <saste> Holden, yes [23:26] <Holden> saste, the problem seems to be that v4l2_ioctl(s->fd, VIDIOC_G_PARM, &streamparm); returns a tpf of 0/0, after applying this patch http://paste.ubuntu.com/1560962/ it seem to work http://paste.ubuntu.com/1560966/ [23:27] <saste> Holden, cool [23:27] <Holden> of course that's not a fix... it just serves to show where the problem is... I had noticed a while back that this tv card returned 0/0, while the webcams return a correct value, but I thought it was related to the tv standard or something [23:27] <saste> so seems a missing validation problem [23:28] <saste> Holden, would you mind opening a ticket on trac, so we don't discard it? [23:28] <Holden> saste, also if I try -framerate 25, it does not accept it [23:29] <Holden> saste, ok, I will do it, but can't promise you I do it now... do I have to register to open a ticket? [23:29] <saste> Holden, yes, there is a register button on the top-right side of the trac page [23:30] <dionys> hi - I try to do avconv -i aaa.MXF aaa.avi the formats are shown and all starts well from the output, the avi file gets created but then it stops at the first frame without error. The avi file is 38k and the MXF 100MB. Why is it not running through the whole file? [23:31] <saste> Holden, https://ffmpeg.org/trac/ffmpeg [23:32] <saste> dionys, missing pastebin but keeps in mind: ffmpeg != avconv [23:32] <saste> *keep [23:34] <Holden> saste, ok, I'm collecting all the info, I'll do it in the following days, I still want to look at the doc on linuxtv.org on why it does return 0/0... [23:37] <dionys> pastebin.com/hfvm1k9n [23:41] <saste> Holden, thanks [23:43] <Holden> saste, ok, I also see we call v4l2_ioctl(s->fd, VIDIOC_S_PARM, &streamparm) without checking for (streamparm.parm.capture.capability & V4L2_CAP_TIMEPERFRAME) first, and that's why -framerate 25 fails... hmm, ok, I'll call it a day, I'll come back in the following days either with a ticket or a patch [23:45] <dionys> I can convert a mp4 into a different size though - but it seems the MXF is just stopping. What do I do wrong there? [23:49] <emc> hi, does anybody know how/if it's possible to show the ffmpeg command executed with a specific ffmpegX configuration? [23:49] <emc> e.g. the output shows: http://pastebin.com/hGshysb2 but it doesn't contain the command executed [00:00] --- Wed Jan 23 2013

1 0

ffmpeg-devel.log.20130121
by burek 22 Jan '13

22 Jan '13

[00:08] <cone-223> ffmpeg.git 03Xi Wang 07master:eb553096e598: lavfi: fix null checking in avfilter_config_links() [00:57] <cone-223> ffmpeg.git 03Michael Niedermayer 07master:00cae86754ef: swr: support first_pts [02:55] <cone-223> ffmpeg.git 03Micah Galizia 07master:c8ea5ccd5db1: fix memory leak on string reassign [03:07] <cone-223> ffmpeg.git 03Michael Niedermayer 07master:c74d4658dcf6: oggdec: #if 0 disabled code [04:24] <cehoyos> nevcairiel: Could you comment on ticket 2153 ? [07:36] <highgod> Hi, I want ask a question, does the community count the download of ffmpeg per month? [11:32] <cone-805> ffmpeg.git 03Paul B Mahol 07master:098d3891be08: EVRC decoder [12:48] <saste> fef906c77c09940a2fdad155b2adc05080e17eda breaks building here [12:48] <saste> http://paste.org/60478 [12:48] <saste> building with no optims [12:52] Action: durandal_1707 gonna push afade [13:45] <cone-805> ffmpeg.git 03Luca Barbato 07master:4603ec85ed62: get_bits: change the failure condition in init_get_bits [13:45] <cone-805> ffmpeg.git 03Mans Rullgard 07master:0b711ca3f3e5: dsputil: drop non-compliant "fast" qpel mc functions [13:45] <cone-805> ffmpeg.git 03Janne Grunau 07master:68f18f03519a: videodsp_armv5te: remove #if HAVE_ARMV5TE_EXTERNAL [13:45] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:a85311ef841b: Merge commit '68f18f03519ae550e25cf12661172641e9f0eaca' [13:51] <cone-805> ffmpeg.git 03Martin Storsjö 07master:d160a2fb4caa: ppc: Include string.h for memset [13:51] <cone-805> ffmpeg.git 03Martin Storsjö 07master:21f5c24b80ca: rtpdec_mpeg4: Return one AAC AU per AVPacket [13:51] <cone-805> ffmpeg.git 03Martin Storsjö 07master:a717f9904227: mpegts: Share the cleanup code between the demuxer and lavf-internal parser functions [13:51] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:c3f3c2256325: Merge commit 'a717f9904227d7979473bad40c50eb40af41d01d' [13:57] <cone-805> ffmpeg.git 03Martin Storsjö 07master:d5bb8cc2dd91: rtpdec: Reorder payload handler registration alphabetically [13:57] <cone-805> ffmpeg.git 03Martin Storsjö 07master:2326558d5277: rtpdec: Split mpegts parsing to a normal depacketizer [13:57] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:950482bf5878: Merge commit '2326558d5277ec87ba6d607a01ec6acfc51c694c' [14:06] <cone-805> ffmpeg.git 03Martin Storsjö 07master:a9c847c1baff: rtpdec: Split handling of mpeg12 audio/video to a separate depacketizer [14:06] <cone-805> ffmpeg.git 03Martin Storsjö 07master:c6f1dc8e4cd9: rtpdec: Move setting the parsing flags to the actual depacketizers [14:06] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:0a5da9cc148b: Merge commit 'c6f1dc8e4cd967ae056698eafb891a08003c211c' [14:15] <cone-805> ffmpeg.git 03Martin Storsjö 07master:a23abaf30257: avconv: Increase the SDP buffer size to fit xiph SDPs [14:15] <cone-805> ffmpeg.git 03Diego Biurrun 07master:d9bf71694504: ppc: vorbisdsp: Drop some unnecessary #includes [14:15] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:144529e09493: Merge commit 'd9bf716945046333c24580d538f450b516f15710' [14:17] <durandal_1707> how should i name this curve: http://fooplot.com/#W3sidHlwZSI6MCwiZXEiOiIxLWNvcyhwaS80KCgyeC0xKV4zKzEpKSI… ? [14:24] <saste> durandal_1707, exponentially increasing frequency cosine graph? [14:48] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:fbdb0313f53f: ac3dec: fix non-optimal dithering of zero bit mantissas [15:00] <cone-805> ffmpeg.git 03Mathias Rauen 07master:04ea5491a83e: ac3dec: fix non-optimal dithering of zero bit mantissas [15:00] <cone-805> ffmpeg.git 03Martin Storsjö 07master:a76bc3bc44e3: rtpdec: Check the return value from av_new_packet [15:00] <cone-805> ffmpeg.git 03Martin Storsjö 07master:f53490cc0c80: rtpdec/srtp: Handle CSRC fields being present [15:00] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:acc0c0190bd6: Merge commit 'f53490cc0c809975f8238d5a9edbd26f83bd2f84' [15:06] <cone-805> ffmpeg.git 03Martin Storsjö 07master:e1d0b3d875a2: srtp: Add support for a few DTLS-SRTP related crypto suites [15:06] <cone-805> ffmpeg.git 03Martin Storsjö 07master:a2a991b2ddf9: srtp: Improve the minimum encryption buffer size check [15:07] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:bc639dbd9b83: Merge commit 'a2a991b2ddf951454ffceb7bcedc9db93e26c610' [15:10] <durandal_1707> saste: now i only need crossfade example... [15:17] <cone-805> ffmpeg.git 03Martin Storsjö 07master:b4bb1d493c44: srtp: Don't require more input data than what actually is needed [15:17] <cone-805> ffmpeg.git 03Martin Storsjö 07master:3ef6d22e1ba5: srtp: cosmetics: Use fewer lines for the test vectors [15:17] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:248140f8a2a6: Merge commit '3ef6d22e1ba544ab37c73e8fc61382f13aac250f' [15:23] <saste> durandal_1707, what do you mean? [15:23] <saste> currently unsupported, we need caching for that [15:24] <saste> btw i can push my fade patchset somewhere if you're still interested into it [15:25] <durandal_1707> what fade patches? [15:26] <cone-805> ffmpeg.git 03Martin Storsjö 07master:ae01e8d29540: srtp: Add tests for the crypto suite with 32/80 bit HMAC [15:26] <cone-805> ffmpeg.git 03Martin Storsjö 07master:c9311f3e4664: srtp: Move a variable to a local scope [15:26] <cone-805> ffmpeg.git 03Ronald S. Bultje 07master:8a4f26206d79: dsputil: remove butterflies_float_interleave. [15:26] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:2cf9ab655549: Merge commit '8a4f26206d7914eaf2903954ce97cb7686933382' [15:28] <durandal_1707> for crossfade amix could be used... [16:13] <cone-805> ffmpeg.git 03Ronald S. Bultje 07master:e6bc38fd49c9: wmv2: move IDCT to its own DSP context. [16:13] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:db23d5de2718: Merge commit 'e6bc38fd49c94726b45d5d5cc2b756ad8ec49ee0' [16:18] <burek> sorry to bug you guys, but can some of you developers give a little help on this: http://ffmpeg.gusari.org/viewtopic.php?f=16&t=800 [16:19] <burek> i would answer the guy if i would knew what he was asking :S [16:21] <cone-805> ffmpeg.git 03Ronald S. Bultje 07master:93af2734ee60: lavc: put FF_IDCT_{VP3,H264,CAVS,BINK,EA,WMV2} under FF_API_IDCT. [16:21] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:5f61e09a8f28: Merge commit '93af2734ee609133eaeca28329d7f13241dd411f' [16:26] <michaelni> burek, @return the size in bytes required for src, a negative error code in case of failure [16:27] <cone-805> ffmpeg.git 03Paul B Mahol 07master:6ea8a830e8f1: afade filter [16:28] <burek> thanks michaelni [16:34] <cone-805> ffmpeg.git 03Ronald S. Bultje 07master:1768e43cebd6: vorbisdsp: change block_size type from int to intptr_t. [16:34] <cone-805> ffmpeg.git 03Ronald S. Bultje 07master:ce378f0dd0c4: fate: Use wmv2 IDCT for wmv2 tests [16:34] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:cf4515ecd985: Merge commit 'ce378f0dd0c4e5350b3280e6b3e8d6b46fe4b0a3' [16:40] <cone-805> ffmpeg.git 03Brad Smith 07master:c7df1532e5d6: libgsm: detect libgsm header path [16:40] <cone-805> ffmpeg.git 03Daniel Kang 07master:9f00b1cbabab: dsputilenc: x86: Convert pixel inline asm to yasm [16:40] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:f6a80d6e6318: Merge remote-tracking branch 'qatar/master' [16:46] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:ed37a566c966: oggdec: remove unused variables [17:18] <durandal_1707> michaelni: where is rendered nut.texi? [17:29] <durandal_1707> saste: ping [17:31] <michaelni> durandal_1707, https://ffmpeg.org/nut.html [17:32] <saste> durandal_1707, ping on what? afade? [17:33] <durandal_1707> saste: your abandoned patches [17:36] <saste> ok [17:38] <saste> durandal_1707, http://gitorious.org/~saste/ffmpeg/sastes-ffmpeg/commits/extend-fade-201301… [17:38] <saste> warning: it is very old code [17:39] <saste> i wanted to extend parse_time format, the code applies no more since av_parse_time() was changed in many ways [17:39] <durandal_1707> michaelni: there is no (u)int64 pcm for some obvious reason ? [17:42] <saste> also i can't remember exactly why i wanted to move pixinfo to lavu, but that patch was (and rightly) rejected [17:42] <saste> the code is complex mostly because of parsing stuff (the idea was to be able to specify multiple fading intervals) [18:02] <durandal_1707> saste: so there is no way to mix 2 or more audio streams at random position? [18:03] <saste> durandal_1707, amix/amerge? [18:03] <saste> no i never do that [18:03] <durandal_1707> michaelni: you did not updated ac3 ref? [18:04] <saste> so can't say if it is achievable with what we have [18:06] <durandal_1707> i do not think so, both of them are very dumb [18:09] <michaelni> durandal_1707, there are new ac3 ref [18:09] <durandal_1707> michaelni: fate failures with ac3 [18:12] <michaelni> where exactly ? [18:12] <michaelni> note, some boxes probably didnt rsync the files yet [18:12] <michaelni> that is they should have a "file not found" issue for a few hours [18:21] <durandal_1707> michaelni: have you looked at -random fate machine failures? [18:21] <durandal_1707> something to do with snow and ff_dsputil_init_dwt [18:46] <durandal_1707> perhaps next useful to implement would be reverb, delay and chorus filter [18:50] <Compn> did you do karaoke filter already? :P [18:50] <Compn> jk [18:52] <durandal_1707> isn't that simple equalizer? [18:55] <wm4> durandal_1707: port scaletempo from mplayer [18:55] <wm4> unless atempo is that [18:56] <durandal_1707> afaik atempo is not based on mplayer [18:59] <Compn> scaletempo is a great audio filter [18:59] <Compn> a lot of users use it to play video at 1.5x speed with normal audio pitch... [18:59] Action: Compn wonders what all these users are doing watching hours and hours of video at fast pitch :D [19:00] <Compn> i guess if you have to watch 6 hours video a day, you can make it less hours by speeding it up [19:03] <durandal_1707> Compn: and atempo can be used? [19:04] <cbsrobot> durandal_1707: I use it from time to time [19:04] <cbsrobot> only problem is that sometimes it is not accurate [19:05] <cbsrobot> if I have a few pcm files with the same length after using atempo the lenght can be different [19:08] <durandal_1707> i dont see it in limpcodecs so do not need to port it :/ [19:09] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:5b95d39ee157: vp3: fix compilation with a disabled theora [19:11] <michaelni> when i added libmpcodecs to ffmpeg there was no audio filter support [19:11] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:aeef76b38371: dsputil: CONFIG_DWT doesnt exist anymore [19:29] <cone-805> ffmpeg.git 03Stefano Sabatini 07master:4630dfd1ebb6: lavf/hlsenc: append last incomplete segment when closing the output [19:29] <cone-805> ffmpeg.git 03Stefano Sabatini 07master:0f2d4eddd813: doc/muxers: extend and clarify documentation for the segment muxer time options [19:32] <wm4> for lavfi filters which take both audio and video, is it defined how these have to be synchronized? (I don't know if there are any such filters) [19:41] <saste> wm4: there are no such filters [19:49] <llogan> michaelni: can you give me WIKI_RENAME permission? only if it is trivial to do and isn't a pain in the ass. [19:52] <michaelni> llogan, done [19:53] <llogan> thanks michael [19:54] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:05340be97bc3: img2dec: support seeking in infinitely looped stream [20:09] <durandal_1707> Compn: if there is bug in atempo filter you should open bug with all information needed for its reproduction [21:05] <durandal_1707> michaelni: it could be nice if eval could support various attractors for use in aevalsrc [21:09] <durandal_1707> there is bunch of audio filters to add that are implemented in one or other way in sox/ladspa/nquist/vst/... [22:28] <durandal_1707> i get crash with this: [22:28] <durandal_1707> ./ffmpeg -f lavfi -i "amovie=eva_2.22_6.1_16bit.wv,asplit=3[out1][a][b]; [a]showwaves=s=340x240,pad=iw:ih*2[waves]; [b]showspectrum=s=340x240[spectrum]; [waves][spectrum] overlay=0:h [out0]" [22:43] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:2dd07518afb3: docs/swr: better used_channel_count docs [22:43] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:83440f40ba37: doc/swr: Mention that the internal sample format is choosen automatically if not set [22:43] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:4606183262e0: doc/swr: try to fix non english "inclusively included between" [22:43] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:e0f208a637c0: doc/swr: docu lfe mix level unit & range [22:43] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:ace74b2122c9: doc/swr: typo fix [22:43] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:c5f9cc96366f: doc/swr: elaborate on what the LFE mix level is. [22:53] <mateo`> michaelni: durandal_1707: i should update the aiff/id3 patch to add a padding byte if the size of id3 data is odd. [22:54] <durandal_1707> you did not do it already? [22:54] <mateo`> no i don't [23:06] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:03a82d588b74: doc/swr: More verbose description of the force resample flag. [23:06] <cone-805> ffmpeg.git 03Michael Niedermayer 07master:e4eebc2da9da: avf_showwaves: fix null pointer dereference [23:26] <cone-805> ffmpeg.git 03Anton Khirnov 07release/1.0:5ee483ae6266: h264: fix ff_generate_sliding_window_mmcos() prototype. [23:26] <cone-805> ffmpeg.git 03Michael Niedermayer 07release/1.0:78526c279fdd: avf_showwaves: fix null pointer dereference [23:26] <cone-805> ffmpeg.git 03Michael Niedermayer 07release/1.0:bec97981badc: h264/ff_generate_sliding_window_mmcos: fix use of uninitialized variable [23:26] <cone-805> ffmpeg.git 03Michael Niedermayer 07release/1.0:75aa373bec58: h264: print mismatching opcode details in check_opcodes() [23:26] <cone-805> ffmpeg.git 03Michael Niedermayer 07release/1.0:7ca4b918989c: h264: fix use of uninitialized variable [00:00] --- Tue Jan 22 2013

1 0

ffmpeg.log.20130121
by burek 22 Jan '13

22 Jan '13

[00:07] <KeshlWare> D: [00:10] <mudlord> oh dear [00:10] <mudlord> thats not supposed to happen! [00:24] <mudlord> hm [02:08] <elkng> "James Cameron stated his intention to film the two Avatar sequels at a higher frame rate than the industry standard 24 frames per second, in order to add a heightened sense of reality", but we cant see more than 24 frames anyway, how it will [02:08] <elkng> increase sense of reality ? [02:17] <KeshlWare> I can see faster than it. [02:17] <KeshlWare> I can see the difference between 60 and 120 FPS. [02:17] <KeshlWare> So there. [02:17] <KeshlWare> <.< [02:17] <KeshlWare> 24 FPS drives me nuts. [02:17] <mudlord> no you cant [02:17] <mudlord> elkng said so [02:17] <KeshlWare> Yes I can. D: [02:17] <KeshlWare> ... Also, seriously? [02:17] <KeshlWare> that's a planned trilogy? Oh boy.. <.< [02:18] <elkng> KeshlWare: quadrology [02:19] <KeshlWare> Whus? .É. [02:59] <_numbers> anyone have tips for eliminating stuttering? i have a video is that is 1hr 52min and i'm trying to stream it over rtmp. it works fine except for a few action scenes. [03:00] <_numbers> i'm using h.264 mp4 and rtmp. i've tried over 40 kinds of bitrates, framerates, and quality settings for both video and audio. each stream a little differently but this one scene where a person is mountain biking always stutters [03:00] <_numbers> i've tried different rtmp servers (aws cloudfront, nginx rtmp), and different rtmp streaming players (in-browser flash, ffplay) [03:01] <_numbers> i've looked at bandwidth monitors and cpu monitors. my cpu is not peaking, neither is my bandwidth being challenged. i don't fully understand why a few scenes would stutter [03:01] <_numbers> i've tried only compiling that scene to its own 15min video, and it streams fine. it seems like a combination of the scene and the length of the video that is streaming [03:02] <_numbers> or , specifically to ffplay, what is the 'A-V: 0.00XX' metric that shows in the stdout console while it is playing rtmp? i noticed it goes from -0.0021 to like -1.500 which is the only unusual thing i could find about that scene [03:03] <iive> my guess audio time minus video time. [03:03] <_numbers> it kind of looks like a buffer because it only goes between 0.000 and some number like -2.000 [03:03] <iive> that is the position of audio samples played in seconds, minus the timestamp of the video frame. [03:03] <_numbers> and it is always bouncing up and down [03:04] <iive> so if I understand correctly, you have the thing already encoded, so you are not re-encoding during streaming. [03:05] <_numbers> 70.15 A-V: -0.001 fd= 104 aq= 102KB vq= 22KB sq = 0B f=0/0 f=0/0 [03:05] <_numbers> right i've tried various encoders too from Sony Vegas Pro, TX264, Handbrake, and ffmpeg in console [03:05] <iive> yes it bounces because audio and video frames are not always exact multiples and something process scheduler is not quite exact. [03:05] <_numbers> but when i stream its a VOD .mp4 file [03:06] <iive> so when you play the file, everything is normal? [03:06] <_numbers> it plays fine except on scenes where there are a lot of pixels changing [03:06] <iive> when playing the file from the hdd ? [03:06] <_numbers> when i play locally its flawless [03:07] <_numbers> i also tried setting up an rtmp server locally with nginx and it played flawlessly [03:07] <_numbers> which must mean that bandwidth plays some role [03:07] <iive> do you have a tool to capture the received stream? [03:07] <_numbers> or the server cpu but its a aws ec2 large instance and i'm the only client streaming it so i doubt that. i have watched in htop and its got 2 cores that never go above 7% [03:08] <_numbers> i have been using streaming video players to test the stream. what do you mean by "capture"? [03:08] <iive> to save it to a file. [03:08] <iive> so you can compare if e.g. frames are missing. [03:08] <_numbers> i am aware of rtmpdump but how would that help [03:08] <_numbers> hmm [03:09] <_numbers> i can try to see if i can get rtmpdump to save it [03:09] <iive> i'm total newb about rtmp protocol. I know nothing of it. [03:10] <iive> Is there any way to specify bandwidth limit for the steaming servers? [03:10] <_numbers> i don't think its rtmp because i've tried two different servers and got the same result [03:10] <_numbers> i think i need to tweak the bitrate or framerate or resolution or some codec setting of h.264 [03:11] <iive> well... if you have limitation of max bitrate, then the server may simply discard any frame that doesn't fit into it. [03:11] <_numbers> i have one version of the encoding that will play the scene without stuttering but its 300kbps bitrate and the quality is unacceptable. very blocky [03:11] <_numbers> ah that could be part of it; server discarding frames [03:12] <iive> usually stuff like vbv is used. [03:12] <iive> but... again... this should be known to the encoder before encoding and you should know what the buffer size is on the client. [03:13] <iive> not that big buffers are problem on PC software. [03:13] <_numbers> the two streaming players i've tried client-side are: MediaElement.js, and the ffplay binary [03:16] <_numbers> if you want i can link you to the test page i made for the video [03:16] <iive> btw, I know of only one good h264 encoder - x264. handbrake and ffmpeg does use it. Sony may be using another encoder... [03:17] <iive> to watch it on my browser? [03:17] <_numbers> the video bitrate i am targeting is 768kbps. i chose that because i am comparing against a disney movie trailer for pirates that plays extremely well at that same bitrate for me [03:17] <_numbers> ya [03:17] <_numbers> the disney trailer is nothing but action scene after action scene for like 12 minutes and it plays great [03:17] <_numbers> through my rtmp server [03:18] <_numbers> whereas my ~2hr video, which is more like a powerpoint slideshow with a few very sparse and short action scenes, plays fine except on the action scenes which stutter horribly [03:25] <_numbers> i just don't understand how disney's 16mb action-packed trailer streams flawlessly but my 669MB with the (as close as i could tell) same encoding settings stutters badly on any action scene of mine [03:35] <iive> so, the issue 12mbps in the relevant part. rc=abr and 2 pass encoding. [03:40] <iive> cu [03:43] <_numbers> found http://mewiki.project357.com/wiki/X264_Encoding_Suggestions#VBV_Encoding [03:43] <_numbers> seems like i would want --vbv-bufsize=768kb --vbv-maxrate=uncapped/maximum --vv-init=minimum [05:21] <Zeranoe> Hi, how can I encode with subtitles using FFmpeg? I'm using the command ffmpeg.exe -i "input.wmv" -vf subtitles="subtitles.srt" -y -b 384000 -r 24 -s 640x480 -codec:v rv20 -vsync 1 -async 1 "output.rm" but I get the error: Error initializing filter 'subtitles' with args 'subtitles.srt' [05:57] <_numbers> how do i pass the string of x264 options to ffmpeg without using a file? [05:58] <_numbers> like: ffmpeg -vpre cabac=1:ref+2:deblock=1:0:0... [06:00] <relaxed> use a preset [06:00] <_numbers> i want to customize the preset [06:01] <relaxed> specify a preset and then add the options you want changed. [06:02] <relaxed> run `ffmpeg -h full | less` and scroll down to the "libx264 AVOptions:" section [08:02] <Nick-S> why is #java invite only ?:( [08:44] <Nick-S> dear fellas, i find it very hard to work with h264 presets [09:01] <jbermudes> I have jack setup with an audio interface that provides 4 channels. Can I use ffmpeg to record a webcam and get the audio from those 4 channels? [09:47] <arshavin> can i watch a video while converting or dumping it from a website simultaneously [09:51] <arshavin> like ffmpeg -i rtsp some link out.mp4 then pipe it to ffplay [09:51] <buhman> ffmpeg is so awesome [09:52] <arshavin> buhman: it sure is,but how can i do that? [09:52] <buhman> oh [09:53] <buhman> arshavin: I wasn't talking about you; I just impressed a group of women with my ffmpeg skills (though they had no idea that I used ffmpeg to do it, doesn't matter :P) [09:53] <buhman> arshavin: if ffmpeg/lavf can decode it, surely ffplay can too [09:54] <arshavin> buhman: but how? [09:54] <buhman> arshavin: well, I'm wondering why you're wanting to "convert or dump" a video and play it at the same time, while you could just play the original [09:54] <buhman> arshavin: is it your ultimate goal to transcode this video, or to play it? [09:55] <arshavin> buhman: is it possible withot writing a shell script? [09:55] <buhman> 02:54 < buhman> arshavin: is it your ultimate goal to transcode this video, or to play it? [09:56] <arshavin> buhman: cos i am on a limited bandwidth,if dump isn't quality i would stop and retry [09:56] <arshavin> buhman: obviously i could do tht separately [09:57] <buhman> ok; so how is ffmpeg involved? [09:57] <buhman> arshavin: in any case, depending on the container, you could just play the output file as it's being written [09:59] <arshavin> buhman: yes,i could ,but I have been wondering if only one line of ffmpeg command could do that [10:00] <arshavin> buhman: the output file of ffmpeg could be piped to ffplay or mplayer [10:00] <buhman> that would not satisfy your first requirement [10:00] <buhman> you are contradicting yourself [10:01] <buhman> arshavin: but yes, you could tell ffmpeg to use stdout, then tee the output to some video player, as well as a file [10:01] <buhman> depending on your requirements (which are still unclear) [10:02] <buhman> why you want this decode -> encode -> decode pipeline is beyond my comprehension [10:02] <arshavin> buhman: yes stdout and output to video player [10:04] <arshavin> buhman: i have tv tuner card on my old desktop,if i could do this i would be able to watch tv at the same time while i m recording [10:04] <klaxa> why not source | tee decode | decode --> encode ? [10:06] <arshavin> buhman:tv tuner device /dev/video wont work with two programs simultaneously so one can either record or wath tv [10:06] <jbermudes> If I have one device that presents itself as 4 jack capture ports, can I use them as separate input streams for a video? [10:08] <arshavin> jbermudes: not sure about that [10:08] <arshavin> jbermudes: if that device creates four separate /dev/device files only then [10:10] <arshavin> jbermudes:The main reason ,i want this to work is because I can't watch tv while I am recording with ffmpeg or mplayer [10:19] <rg_> hi..I just installed ffmpeg using instructions from https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide , on ubuntu 12.04, but am getting error: ./a.out: error while loading shared libraries: libavcodec.so.54: cannot open shared object file: No such file or directory when I try to run my opencv code [11:34] <raket> did anyone figure out how to setup ffserver the same m.youtube.com does? :-) [12:08] <buhman> raket: they use ffserver? [12:15] <raket> regardless if they do or not, i'm interested if ffserver can behave the same way as that does with rtsp protocol. for streaming :-) [12:19] <raket> is it possible for ffserver to stream h.264 (MPEG Part 10/AVC), Baseline profile, QVGA (240x340), 30 fps, 196kbit? [12:19] <raket> 240x320 :-) [12:21] <durandal_1707> it can stream everthing libav* supports [12:23] <raket> Video: h263, yuv420p, 176x144 [SAR 12:11 DAR 4:3], 29.97 tbr, 90k tbn, 29.97 tbc Audio: amr_nb, 8000 Hz, 1 channels, flt. what would be a suitable ffserver.conf or ffmpeg cmdline for exactly that :-)? [12:43] <raket> omg thank you guys. i got it to work, and even though the phone specs says only 192kbit is supported it supported 1mbit stream anyway [12:57] <scrambles> hey guys, i'm trying to compile ffmpeg with h264 to use in an iphone project and i'm getting the following error on make: http://pastebin.com/LDQB964q [12:58] <scrambles> can anyone give me a little help? thks [12:59] <Mavrik> hmm, looks like broken cross-compilation settings :\ [13:01] <scrambles> What do you suggest Mavrik? Do you want me to show you my configure command? [13:40] <raket> okay. Format rtp works in ffserver. but my device sends "PLAY test.3gp/streamid=0 RTP/TCP" while ffplay sends "PLAY test.3gp/streamid=0 RTP/UDP" ? :-) [14:11] <xetius> Im trying to convert some videos into one of the following formats for my daughter http://pastebin.com/Hpfai3mp. But I don't have a clue as to what I should set for the parameters. Could anyone help? [16:21] <burek> xetius, what exactly should the video and audio codecs (and params) be [16:21] <burek> since i dont quite recognize your pastebin [16:21] <burek> as an output of any familiar software [16:22] <burek> raket [16:22] <burek> you can force ffplay to tcp mode [16:23] <burek> take a look here http://ffmpeg.org/ffmpeg-protocols.html#rtp [16:23] <burek> what you need is -rtsp_transport tcp [16:24] <burek> so that ffplay would skip trying udp first [16:24] <Mavrik> burek: it seems either standard MPEG2 or xvid versions of it [16:24] <Mavrik> with PCM or MP3 audio [16:24] <xetius> sorry, the pastebin is from the TV manual, which specified the video files playable on the TV [16:25] <xetius> Apologies, I have found a solution to this now, using avconv [16:25] <xetius> avconv -i yourvideo -c:a libmp3lame -b:a 128k -c:v mpeg2video -qscale 2 -y output.mpg [16:25] <xetius> these work [16:31] <burek> Thanks Mavrik [16:31] <burek> xetius, well great :) [16:31] <burek> btw, you are not using ffmpeg (avconv is not ffmpeg) [16:31] <burek> it's just a fork of ffmpeg [16:35] <xetius> Yes, I have both (Source of both), and was just looking for something that worked. Have used FFMpeg over the years, but the guys at libav were a little quicker with their response [16:36] <xetius> I know about the history of ffmpeg and avconv [16:44] <durandal_1707> xetius: same paramters you found works with real ffmpeg too.... [16:55] <Zeeflo> Can someone remind me of the track options when I want to encode i.e track 0:1 and track 0:2 from a mkv? [16:57] <JEEB> map [16:57] <Zeeflo> ah yea [16:57] <JEEB> the first zero is the number of input file [16:57] <JEEB> so if you have more than one -i [16:57] <JEEB> the first one is zero, second one is 1 etc. [16:57] <JEEB> and then the second number is the track [16:58] <Zeeflo> so it would be map 0:2 then ? [16:58] <Zeeflo> 0 for video and 2 for the audio ? [16:58] <JEEB> no [16:58] <JEEB> first number is input file number [16:58] <JEEB> second is track number [16:58] <JEEB> so if you want to have tracks 0 and 2 of the first input [16:58] <JEEB> -map 0:0 -map 0:2 [16:59] <Zeeflo> AH YEA like that [16:59] <Zeeflo> so a MAP for every stream you want in the output file [16:59] <Zeeflo> got it [17:00] <Zeeflo> thanks [17:02] <Zeeflo> Can it be true that the encoding is faster when encoding from a 720p MKV then it is from a 720p AVI with the same settings? [17:03] <klaxa> encoding should not be faster if you have the same settings and the same machine [17:03] <JEEB> is the encoding faster from a red box than from a white box :P [17:03] <klaxa> decoding speed could be slightly faster [17:03] <JEEB> you mention containers [17:03] <klaxa> that too [17:04] <klaxa> i'mma assume h264 and xvid :P [17:04] <Zeeflo> yea, thats right [17:04] <JEEB> s/xvid/mpeg-4 part 2/ [17:04] <Zeeflo> h264 and xvid [17:04] <Zeeflo> it somehow seems encoding from h264 (mkv) to h264 (mp4) is faster then from xvid to h264 (mp4). [17:05] <Zeeflo> Even though I also would think it would be the same.. [17:05] <JEEB> well, the container probably doesn't matter jack [17:05] <Zeeflo> a setting is a setting.. [17:05] <JEEB> but the video format can [17:05] <Zeeflo> yea [17:05] <JEEB> in theory mpeg-4 part 2 should decode faster, but I wouldn't be surprised if no-one actually bothered to optimize it as much [17:06] <JEEB> because H.264 is much more needed nowadays [17:06] <Zeeflo> yea [17:06] <Zeeflo> I use it for streaming [17:06] <Zeeflo> i would use webm [17:06] <Zeeflo> but its just toooo god d*** slow! [17:06] <Zeeflo> (encoding that is) [17:07] <JEEB> also it could be the default for that format regarding decoding multithreading [17:07] <JEEB> I dunno [17:07] <JEEB> I have no idea :P [17:08] <Zeeflo> no matter what ive tried, including setting threads etc, webm encodes 3-4 times slower then h264 [17:08] <Zeeflo> and im talking on a 32 threaded dual cpu xeon with 32 gb ram [17:09] <Zeeflo> and raid0 ssd's [17:09] <Zeeflo> im looking forward to webm9 [17:14] <klaxa> webm9? you mean vp9 [17:14] <klaxa> ? [17:15] <klaxa> also vp8 != h264 [17:15] <klaxa> afaik [17:16] <JEEB> he meant libvpx and vp8 output [17:17] <JEEB> and yeah, he isn't saying that vp8 equals H.264 [17:17] <JEEB> and yeah, he probably meant vp9 [17:18] <Zeeflo> yea vp9 [17:18] <JEEB> BBB is hell-bent on kicking x264's ass, so all power to them. The more people are trying to compete, the better [17:18] <JEEB> that said, I really wonder what the result will be like [17:20] <Zeeflo> I just hope that I dont have to re-encode 300 titles from x264 to vp9 any time soon :P [17:22] <Zeeflo> but i do suspect vp9 will be far superior on crossplatforms/browsers.. [17:22] <Zeeflo> So they day _will_ come! [17:23] <Zeeflo> you try viewing a mp4 streamed to Firefox.. Its not really a pleasent experience. [17:23] <JEEB> the media foundation backend seems to work relatively well [17:23] <JEEB> in the nightlies [17:25] <Zeeflo> on IE8+ it works good! Except for true fullscreen [17:25] <Fjorgynn> i6 is the best still [17:25] <Fjorgynn> Zeeflo: stop lkying [17:25] <Zeeflo> im not lying. [17:25] <JEEB> on win7+ you can try the nightlies and do this http://browserfame.com/1033/firefox-h264-codec-windows [17:25] <Zeeflo> any of my mp4s streamed to IE9 wont play in true fullscreen [17:26] <Zeeflo> no matter what player I use! [17:26] <Zeeflo> the only browser that handles mp4 really well is safari (obviously) and chrome. [17:27] <Zeeflo> on second place comes IE, but the windowed full screen is f'ing annoying. FireFox and other browsers dont handle mp4 at all! [17:28] <Zeeflo> hmm [17:29] <durandal_1707> browser is not multimedia reproducer [17:29] <Zeeflo> Updating firefox.. I hope it helps! [17:30] <Zeeflo> Uhm.. [17:30] <Zeeflo> Why did FF only update to 18 ? [17:31] <Zeeflo> isnt nightly 20? [17:31] <Zeeflo> Still plays like crap in FF [17:50] <Zeeflo> JEEB, did you download and install the SDK kit to get better playback? [18:25] <fatpony> when i encode some m2ts with ffmpeg and x264 and mux them to mkv, sometimes the framerate is shown as variable, should i force it while encoding or while muxing? [18:26] <Zeeflo> variable framerate? [18:26] <microchip_> if i use the -ar option, will it automatically resample to the Hz specified? [18:27] <Zeeflo> VBR in the sound is ok.. [18:27] <Zeeflo> if thats what you mean [18:28] <Zeeflo> but if you dont want to use vbr, just use cbr [18:30] <Mavrik> fatpony: hmm... you shouldn't force fps to something that's not encoded [18:31] <Mavrik> fatpony: so... if you want to force fps then do it while encoding [18:31] <Mavrik> so you'll have as much frames as your container will claim you have [18:31] <Mavrik> microchip_: yes. [18:31] <fatpony> Mavrik: so i should add the framerate to ffmpeg? not mkvmerge? [18:31] <Mavrik> yep [18:31] <Mavrik> or... just ignore the flag if video looks ok :) [18:31] <Zeeflo> ahh you have raw video [18:32] <Zeeflo> probe it [18:32] <Zeeflo> define in ffmpeg when ffprobe tells you the video is [18:32] <microchip_> Mavrik: thanks [18:33] <fatpony> Zeeflo: yeah, i was encoding from a vc1 stream actually [18:35] <fatpony> will ffmpeg convert 23.976 to 24000/1001? [18:35] <Mavrik> yea [18:36] <Mavrik> you can also pass 24000/1001 to "-r" IIRC [18:36] <fatpony> well, as long as it ends up being the correct rate... [18:36] <Zeeflo> it will [18:47] <xetius> If I just supply multiple -i sources, will it just join the videos? or do I have to do something extra? [18:55] <llogan> xetius: http://ffmpeg.org/faq.html#How-can-I-join-video-files_003f [18:55] <llogan> so no, it won't just join the videos [18:56] <xetius> llogan, thanks. So I seem to need to convert the avi videos to mpg, cat them and then convert them back? [18:58] <llogan> that is one method as shown in the FAQ, but there is also the concat protcol, filter, and demuxer [19:08] <Zeeflo> exit [20:13] <d0x0b> hello i am trying to convert sequence of images jpg into movie avi or mov but only one image is shown through the whole video period i am using this command: ffmpeg -f image2 -r 24 -i image%03d.jpg -vcodec mpeg4 -b 800k video.avi [20:16] <d0x0b> llogan: sure one sec [20:16] <d0x0b> http://pastie.org/5785418 [20:18] <llogan> are your images sequentially numbered, starting with image001.jpg? [20:19] <d0x0b> yes [20:20] <d0x0b> image000.jpg is first one [20:20] <d0x0b> from image000.jpg upto image100.jpg [20:22] <llogan> can you provide the output video? [20:22] <Zeeflo> http://pastebin.ca/2305455 any ideas? [20:23] <d0x0b> llogan: yes but private pls [20:23] <llogan> d0x0b: you can PM the link if you prefer [20:24] <d0x0b> yes tnx [20:24] <d0x0b> one sec [20:24] <llogan> Zeeflo: that's not much context [20:25] <Zeeflo> llogan, its the error message [20:26] <durandal_1707> what error message? [20:26] <Zeeflo> http://pastebin.ca/2305455 any ideas? [20:26] <Zeeflo> that one [20:26] <Zeeflo> i get that error message and ffmpeg shuts down [20:26] <durandal_1707> i see no error message [20:27] <Zeeflo> what do you see then [20:27] <durandal_1707> you should use image%3d.jpg [20:27] <Zeeflo> i should what? [20:28] <d0x0b> durandal_1707: even if i change to image3d.jpg still one image in video only [20:29] <durandal_1707> are you sure that all images you have are really different? [20:30] <d0x0b> fuck wrong dir [20:30] <d0x0b> lol my mistake [20:30] <d0x0b> all images are same [20:30] <d0x0b> auch auch [20:30] <llogan> heh. i was wondering about the images too...sicne the duration was about right [20:30] <d0x0b> yes llogan hehe my mistake [20:30] <d0x0b> sorry for this [21:45] <Fjorgynn> :D [22:03] <apo> Hi [22:04] <apo> just checked out git, trying to compile it with the geq filter enabled, but even if I run configure with --enable-filter=geq it doesn't show up as enabled. Help? [22:07] <llogan> apo: that filter requires --enable-gpl [22:07] <llogan> you won't ever need to --enable-filter=foo [22:07] <apo> llogan: Thanks, that worked. [22:13] <apo> llogan: Do you know what color space exactly I'm working in when I'm fiddling with the values in geq? [22:13] <durandal_1707> yuv ones of 8 bit depth [22:14] <apo> YUV? Not YCbCr? [22:15] <apo> oh, it's different between videos [22:15] <durandal_1707> it is YCbCr [22:22] <apo> That... looks wrong [22:23] <apo> but at least it's doing something :P [22:45] <Snaggle> I'm getting a build error at libavcodec/h264_refs.c with ffmpeg-1.0.2 on OS X. http://paste.lisp.org/display/134819 This failure doesn't happen with either ffmpeg 1.0.1, 1.1* or ffmpeg-git head. I see a some chatter in the git logs for this file over the last couple days, but it doesn't seem to address this particular failure. configure and failed compile command at above paste url. [22:49] <durandal_1707> michaelni: ^ [23:18] <michaelni> Snaggle, will be fixed in the next 1.0 release [23:22] <Snaggle> michaelni: awesome. thank you for the fast response [23:26] <michaelni> Snaggle, you can check the release/1.0 branch, it should work [23:33] <Snaggle> michaelni: seems good now (took a sec to figure out how to checkout a release branch) [23:33] <Snaggle> thanks again [00:00] --- Tue Jan 22 2013

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