May 2015 - Ffmpeg-devel-irc - lists.ffmpeg.org

ffmpeg-devel.log.20150510
by burek 11 May '15

11 May '15

[01:18:04 CEST] <cone-425> ffmpeg 03Michael Niedermayer 07master:f349703e1dcc: avcodec/snowenc: change log level for iterative ME information to debug level [01:18:05 CEST] <cone-425> ffmpeg 03Michael Niedermayer 07master:db6d0a5bd244: avcodec/snowenc: Fix used reference MV in fullpel iterative ME [02:48:46 CEST] <cone-425> ffmpeg 03Michael Niedermayer 07master:58201c6e9eec: avcodec/snowenc: correct fullpel search pattern [08:48:50 CEST] <cone-751> ffmpeg 03Reimar Döffinger 07master:5af621278d45: avformat_alloc_context: initialize raw_packet_buffer_remaining_size. [14:30:17 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:729108515dc1: avcodec/snowenc: set dia_size for the first pass of motion estimation too [14:46:39 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:8b42b2222209: avformat/mxf: Simplify PRINT_KEY() [14:46:40 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:3d65570f2560: avformat/mxf: Always evaluate PRINT_KEY() arguments at compile time [15:21:49 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:a649f36f75f6: avformat/http: Mark lower_proto as const in http_listen() [15:21:50 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:54e64eaf68e0: swscale/x86/swscale: Fix warning about loosing significant bits in cast [16:22:22 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:4950bd4ebedb: libavutil/mem: use size_t for the length in av_strdup() [16:22:23 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:4c128ea16291: avutil/dict: Use size_t for appending strings [17:08:46 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:d66b623ad077: avformat/nutenc: Fix writing strings larger than 2gb [17:08:47 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:07cb6bf985e2: postproc/postprocess: Use size_t to hold strlen()s value [20:12:17 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:d0b294286c1c: avcodec/ppc/h264qpel_template: Fix alignment assert [20:12:18 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:ecb4ae7879fd: avformat/wavenc: fix string length variable types [20:12:20 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:004564c930ef: avformat/wavenc: Check umid length [21:59:35 CEST] <cone-583> ffmpeg 03Andreas Cadhalpun 07master:295e05a76233: ape: Support _0000 files with nblock smaller than 64 [21:59:36 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:918c5ea4d370: Merge commit '295e05a762332c5edcc84c325e94457815a51b5c' [22:08:33 CEST] <cone-583> ffmpeg 03Andreas Cadhalpun 07master:243e8443cd9e: alac: Reject rice_limit 0 if compression is used [22:08:34 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:ce1c0e99e9f6: Merge commit '243e8443cd9e83c887e3f5edf09a169e7783d14e' [22:17:11 CEST] <cone-583> ffmpeg 03Andreas Cadhalpun 07master:b34257eefd98: nut: Check chapter creation in decode_info_header [22:17:12 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:8b0ceb96da0e: Merge commit 'b34257eefd98cb768ea2f53a390b8684d51e689a' [22:27:14 CEST] <cone-583> ffmpeg 03Andreas Cadhalpun 07master:4d0ee4962be7: avi: Validate sample_size [22:27:15 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:73b8155b31ec: Merge commit '4d0ee4962be7e07cdc038a78008ef2e4e47e5f81' [22:38:46 CEST] <cone-583> ffmpeg 03Luca Barbato 07master:254f3daba427: nut: Make sure to clean up on read_header failure [22:38:47 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:55231323b0fd: Merge commit '254f3daba4271c1918d9a7ad155b1442ef93ed29' [22:49:57 CEST] <cone-583> ffmpeg 03wm4 07master:b9f7a6770836: png: Set the color range as full range [22:49:58 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:f230451b7a69: Merge commit 'b9f7a677083647d85e583d6d7384938766a293d7' [22:59:40 CEST] <cone-583> ffmpeg 03wm4 07master:379e5603f413: id3v2: Do not export empty APIC description [22:59:41 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:1cf81a2c509e: Merge commit '379e5603f4131e677ecf5ba79973dcd5603f28ea' [23:16:01 CEST] <cone-583> ffmpeg 03Luca Barbato 07master:2fb02ecfd115: fate: Reduce the number of frames used to test video filters [23:16:02 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:5e19651c9da1: Merge commit '2fb02ecfd115a627f22df21720aa0bd4f6fcee8b' [23:49:46 CEST] <cone-583> ffmpeg 03Michael Niedermayer 07master:8d916ef5f74d: fate: Use more than 5 frames for testing select and idet filters [00:00:00 CEST] --- Mon May 11 2015

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ffmpeg.log.20150510
by burek 11 May '15

11 May '15

[00:09:09 CEST] <RobotsOnDrugs> Cobra_Fast: q is not the same as crf [00:10:28 CEST] <RobotsOnDrugs> and also, why are you using maxrate/bufsize with crf? [00:11:34 CEST] <Cobra_Fast> because i'm realtime streaming [00:12:08 CEST] <RobotsOnDrugs> then why use crf? [00:12:48 CEST] <Cobra_Fast> my idea was to use only as much bandwidth as i need but cap it for the hbo logo [00:13:44 CEST] <RobotsOnDrugs> i suppose [00:14:45 CEST] <RobotsOnDrugs> anyway, you probably don't need to pay attention to q at all, but rather the bitrate and the quality you get for it [00:15:01 CEST] <Cobra_Fast> alright, so i just need to have faith in it doing the right thing [00:15:22 CEST] <RobotsOnDrugs> kind of [00:15:32 CEST] <RobotsOnDrugs> you should look at the quality you get [00:15:50 CEST] <RobotsOnDrugs> but q is not the same as crf, so don't expect them to match [00:16:04 CEST] <Cobra_Fast> okay, thanks [00:17:01 CEST] <RobotsOnDrugs> q is a more "objective" measurement of quality, whereas crf allows psychovisual stuff that will degrade quality to save bits in a way that you aren't likely to notice [04:24:15 CEST] <casanova> How does an IRC channel work? [04:24:41 CEST] <c_14> Magic electrons [04:24:57 CEST] <casanova> Good enough for me. [04:26:58 CEST] <klaxa> lol? [04:29:30 CEST] <c_14> For all your important FFmpeg related questions. [04:51:58 CEST] <trodis_> oh guys i need some help [04:52:04 CEST] <trodis_> going almost crazy with ffserver [04:52:51 CEST] <trodis_> i want to stream audio with as low delay as possible like this: ffmpeg <-> ffserver <-> mplayer / mpc / mpd e.t.c [04:53:21 CEST] <trodis_> streaming does work, wether it is mp3, ac3 whatever but the delay is to much [04:54:09 CEST] <trodis_> i dont know how i can reduce the delay, the buffering on the client side is turned off of course. I even tried libopus, but libopus has the biggest delay [04:54:22 CEST] <trodis_> does someone has experience with this kind of problems? [05:48:55 CEST] <casanova> Anyone good at compiling ffmpeg? [05:50:11 CEST] <BtbN> gcc does a solid job at it [05:50:20 CEST] <casanova> What is gcc? [05:50:28 CEST] <BtbN> A C compiler. [05:51:14 CEST] <casanova> I'm having issues with getting libfdk-acc and x264 together. [05:54:48 CEST] <casanova> I had everything running good with the Jon Severson PPA and then I updated i t like a fool [05:56:05 CEST] <casanova> brb [06:03:47 CEST] <casanova> What triggers the "past duration x too large" messages in the newest version of FFMPEG? [09:12:24 CEST] Action: messup hello mates.... [10:09:57 CEST] <messup> hello guys.... [10:10:28 CEST] <messup> i got stuck to concatenate videos in ffmpeg.. [10:13:39 CEST] <messup> i have a bunch a video with different codecs,container,size,sar and everything, but i wanna make them same codecs and parameter to use concat demuxer? what command i use to achieve this? and its allright if have to add some black screen if aspect ratio not same. [10:14:45 CEST] <messup> i cant find a really...really good command to make them exactly same? how to do that? thanks. [11:40:05 CEST] <seasc> messup, have a look at: http://ffmpeg.org/ffmpeg-all.html#concat-2 [11:46:45 CEST] <messup> thanks seasc, but i think its concat protocol didnt work on mp4 files. but anyway i had managed to make a same codecs and its parameters as well but when i use concat demuxer the file is scretching when its play. why?? thanks [11:51:38 CEST] <seasc> messup, yes, the example shows to use mpeg files, so you're best of converting all the related files to mpeg first [11:52:12 CEST] <seasc> messup, uh no idea, never used conact but once, and i did what i just suggest to you [11:53:26 CEST] <messup> thanks seasc. ok i try. [11:54:43 CEST] <messup> do you actively make videos??? [11:56:27 CEST] <messup> i just wanna make some youtube videos with less effort, so i think it would be ffmpeg. [12:01:00 CEST] <seasc> not atm, but planning to do so... but me wrote a linux script to handle most ffmpeg tasks i came across (ironicly its easier for me to write a script that handles 'complex' tasks, than doing them manualy) [12:01:40 CEST] <seasc> It is a great choice, though, at times i cannot upload videos, hangs at youtube at somewhat around 95% :( not sure if its me or the video. [12:04:49 CEST] <messup> oh you wrote a script, im planning to do so, but im new to ffmpeg so... need to dig around little bit. to make it most mostly less effort job as possible.hhhahhahha [12:08:55 CEST] <messup> so what software you use to make a videos for youtube? i found openshot, but its damn slow when its come to rendering? so i m still searching the alternative.... [13:40:55 CEST] <trodis> i want to stream audio with as low delay as possible like this: ffmpeg <-> ffserver <-> mplayer / mpc / mpd e.t.c [13:41:05 CEST] <trodis> streaming does work, wether it is mp3, ac3 whatever but the delay is to much [13:41:12 CEST] <trodis> i dont know how i can reduce the delay, the buffering on the client side is turned off of course. I even tried libopus, but libopus has the biggest delay [13:41:16 CEST] <trodis> does someone has experience with this kind of problems? [17:04:06 CEST] <Zerowalker> What metdata date times can be stored via ffmpeg? [17:04:26 CEST] <Zerowalker> and what does it translates to (creation_time == Encoded Date) i think for example [19:25:00 CEST] <jrun> how do i change the colour depth of the output to say 256 (8-bit)? [19:25:21 CEST] <jrun> sample_frmt? -vf format=something_here? [19:25:35 CEST] <jrun> i have discovered format=gray so far but what are the other options? [19:25:50 CEST] <jrun> how do i get a list of available formats? [19:52:49 CEST] <techtopia> is there a way to pause an ffmpeg encoding job while it's running [19:54:36 CEST] <klaxa> you can probably press ctrl+z in your shell if it supports that [19:55:22 CEST] <c_14> techtopia: or just send SIGSTOP [20:00:05 CEST] <techtopia> how do you send sigstop? [20:00:32 CEST] <c_14> kill -SIGSTOP pid [20:01:32 CEST] <techtopia> im on windows running it from command line [20:07:17 CEST] <c_14> I have no idea how job control works on windows. [20:08:01 CEST] <techtopia> yeah i don't think it's possible [20:08:05 CEST] <techtopia> just wanted to double check [20:27:40 CEST] <gr1sha> How can I fix this? [20:27:40 CEST] <gr1sha> undefined reference to `av_frame_get_best_effort_timestamp' [20:27:49 CEST] <gr1sha> I'm trying to comiple dranger's tut07 [20:28:54 CEST] <gr1sha> eran@eran-vm:~/Desktop$ gcc -o tutorial07 tutorial07.c -lavutil -lavformat -lavcodec -lswscale -lz -lm `sdl-config --cflags --libs` [20:28:54 CEST] <gr1sha> tutorial07.c:(.text+0x15c1): undefined reference to `av_frame_get_best_effort_timestamp' [20:28:54 CEST] <gr1sha> tutorial07.c:(.text+0x15f9): undefined reference to `av_frame_get_best_effort_timestamp' [20:28:54 CEST] <gr1sha> collect2: error: ld returned 1 exit status [20:29:29 CEST] <BtbN> What ffmpeg version are you using? [20:30:41 CEST] <gr1sha> ffmpeg version 0.8.13-6:0.8.13-0ubuntu0.13.10.1, [20:30:44 CEST] <gr1sha> O-o [20:31:04 CEST] <BtbN> Well, that's simply too old. [20:31:09 CEST] <gr1sha> yeah I can see now [20:31:16 CEST] <gr1sha> how come 2.6.2 isn't available through apt-get? [20:31:17 CEST] <BtbN> And propably not even ffmpeg, as it's ubuntu. [20:31:29 CEST] <BtbN> Because you are using a very old ubuntu version. [20:31:42 CEST] <BtbN> And because ubuntu has libav instead of ffmpeg. [20:32:59 CEST] <gr1sha> jeez [20:34:35 CEST] <gr1sha> what would you suggest? [20:40:13 CEST] <gr1sha> BtbN, would it work on 15.04? [20:40:28 CEST] <BtbN> No idea [20:42:15 CEST] <gr1sha> I'll have to try, Thaknks a lot! :) [21:47:25 CEST] <troy_s> gr1sha: Get a real FFMPEG is reasonable advice. [21:48:02 CEST] <troy_s> gr1sha: Also, the tutorial is dead old, and much of the API has shifted quite dramatically. [21:52:15 CEST] <gr1sha> troy_s, it was updated pretty recently and I'm using it because it's a requirment in a course I'm taking in the University [22:00:49 CEST] <troy_s> gr1sha: Regardless, that error is telling you that the API is missing that function call. So you should update your local FFMPEG and see if the errors persist [22:01:04 CEST] <gr1sha> I will, thanks! [22:47:35 CEST] <koz_> What command would I need to use to convert a video encoded in AVC and contained in an MKV into a VP9 contained in webm? The sound is already encoded in a decent enough way, so I don't wanna change that. [23:04:35 CEST] <JEEBsv> koz_: ffmpeg -i input.mkv -c:a copy -c:v libvpx-vp9 <any other libvpx vp9 options> output.webm [23:36:22 CEST] <koz_> JEEBsv: What do you mean by the 'any other' part of that command? [23:36:29 CEST] <koz_> Should I just literally put 'any other' there? [23:37:03 CEST] <c_14> If you have special options that you want to give to libvpx you can put them there, but you don't have to. [23:38:01 CEST] <koz_> Ah, ok. Thanks. [23:40:06 CEST] <JEEBsv> like rate control, coding settings etc [23:40:27 CEST] <JEEBsv> no idea how insane the defaults are for libvpx [23:41:54 CEST] <DragonsLord> Hello! [23:41:55 CEST] <DragonsLord> Does anybody know the best way to report FFmpeg a bug? [23:42:24 CEST] <c_14> https://ffmpeg.org/bugreports.html [23:42:40 CEST] <DragonsLord> Thank you c_14, pleased to read from you [23:42:43 CEST] <DragonsLord> :) [00:00:00 CEST] --- Mon May 11 2015

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ffmpeg-devel.log.20150509
by burek 10 May '15

10 May '15

[00:35:21 CEST] <cone-786> ffmpeg 03Michael Niedermayer 07master:d1d8ee5e9191: avdevice/v4l2: Try to fix build on netbsd/openbsd [01:25:39 CEST] <cone-786> ffmpeg 03Michael Niedermayer 07master:9025bc588ad4: avcodec/h263dec: use PIX_FMT_GRAY8 when the user specified CODEC_FLAG_GRAY [01:54:42 CEST] <cone-786> ffmpeg 03Michael Niedermayer 07master:82a6e18bbb6e: avcodec/mpegvideo: Fix -flags +gray with xvid_vlc_trac7411.h263 [03:49:02 CEST] <cone-786> ffmpeg 03Michael Niedermayer 07master:5c8809b45b78: avcodec/error_resilience: support grayscale [04:19:29 CEST] <jolson88___> Hey all. Not sure if the right channel is this or #ffmpeg. I'm trying to compile ffmpeg with Visual Studio 2015 for Universal Windows apps (following the guide: https://trac.ffmpeg.org/wiki/CompilationGuide/WinRT) But when I get to Win10 x64 and run the configure command, I get "cl is unable to create an executable file". Any thoughts? [04:23:42 CEST] <rcombs> jolson88___: post config.log [04:27:17 CEST] <jolson88___> Didn't know about config.log (I'm a newbie here :P). https://gist.github.com/jolson88/74e75a749d624aa25612. Looks like stdlib.h couldn't be found [04:27:50 CEST] <rcombs> (don't stick punctuation immediately after links) [04:28:24 CEST] <jolson88___> Oops. Apologies. https://gist.github.com/jolson88/74e75a749d624aa25612 [04:28:36 CEST] <rcombs> you're probably missing some env vars [04:28:47 CEST] <rcombs> visual studio comes with a script to set them [04:30:06 CEST] <jolson88___> Ah, that rings a bell. I thought they were automatically set via the tools shells that the wiki guide has it run. But I think you're right. I'll peek into that [04:31:16 CEST] <rcombs> they should be :\ [04:31:36 CEST] <rcombs> so if they're not there's something wrong, but that's where my experience in the area ends [04:31:39 CEST] <jolson88___> Looks like it does. The VS command prompt you run before launching MSYS2 is a shortcut to "%comspec% /k ""C:\Program Files (x86)\Microsoft Visual Studio 14.0\VC\vcvarsall.bat"" x86_amd64" [04:31:46 CEST] <jolson88___> :) [04:31:51 CEST] <rcombs> also, #ffmpeg is the place for user support [04:32:07 CEST] <rcombs> (but most people don't care much) [04:32:12 CEST] <jolson88___> great! Thanks. I'll dig a little more and then move the discussion there. [04:32:19 CEST] <rcombs> good luck! [04:32:21 CEST] <jolson88___> Would the lists or anything be better for traction? [04:33:00 CEST] <rcombs> if you can't work it out from here, try #ffmpeg and then if you don't get anything, yeah, try the user list [04:34:53 CEST] <Compn> theres also a ffmpeg forum [04:34:57 CEST] <Compn> if you like forums better [04:35:14 CEST] <rcombs> is there? [05:30:25 CEST] <jamrial> afaik, ffmpeg doesn't yet fully compiles with VS2015. bigpet was working on fixing the couple issues some weeks ago [05:30:41 CEST] <bigpet> the libs work fine [05:30:56 CEST] <bigpet> the command line client just has that one little problem [05:31:02 CEST] <bigpet> that's really really annoying to fix [05:31:22 CEST] <bigpet> because unless you want to change the semantics you need to introduce another thread for the windows version [05:33:18 CEST] <bigpet> rcombs: but if you just want to make it compile and work for most cases just delete these lines: https://github.com/FFmpeg/FFmpeg/blob/master/ffmpeg.c#L394-L397 [05:33:45 CEST] <rcombs> jolson88___ is the one who cares [05:33:59 CEST] <bigpet> whoops sorry [05:35:03 CEST] <bigpet> well but reading it, it doesn't seem like he wants the command line client anyway [05:35:16 CEST] <bigpet> so it should be fine as is [13:19:33 CEST] <cone-474> ffmpeg 03Michael Niedermayer 07master:9ba5dd37a96c: avcodec/mpegvideo: Fix indention [13:19:34 CEST] <cone-474> ffmpeg 03Michael Niedermayer 07master:107e4da47644: ffmpeg_opt: Fix -timestamp parsing [13:19:35 CEST] <cone-474> ffmpeg 03Michael Niedermayer 07master:876c89a88f64: cmdline: Check options array for not working flag combinations [17:30:45 CEST] <cone-319> ffmpeg 03Michael Niedermayer 07master:33529104e56a: avcodec/mpeg12dec: Fix "-flags +gray" [17:30:46 CEST] <cone-319> ffmpeg 03Michael Niedermayer 07master:cf31e2df08e3: avcodec/mpegvideo: Fix null pointer dereference with grayscale decoding with mpeg2_field_encoding.ts [20:44:31 CEST] <wm4> holy shit I'm being a dick to cehoyos [20:48:31 CEST] <ubitux> for a change [20:48:33 CEST] <ubitux> ;) [20:48:45 CEST] <ubitux> well just suggest something constructive maybe [20:49:29 CEST] <wm4> there's no way an application can do something reasonable with ffmpeg log output if user interaction is required [21:02:35 CEST] <kierank> othisproblemagain [21:02:36 CEST] <kierank> lol [22:17:32 CEST] <Compn> wm4 : but why trolling with nicolas and carl ? [22:17:48 CEST] <Compn> why not call it a difference of opinion, ignore any trolling, stick to the code and patches and ignore the rest. [22:18:33 CEST] <Compn> tl;dr reply with patches or stfu ? [22:18:38 CEST] <Compn> just trying to halp. [22:18:39 CEST] Action: Compn out [22:19:47 CEST] <wm4> well you didn't [22:45:16 CEST] <Timothy_Gu> michaelni: power cable with Chinese plug to FFmpeg huh? [22:45:38 CEST] <Timothy_Gu> michaelni: oh nvm [22:45:49 CEST] <Timothy_Gu> English parser crashed [23:53:31 CEST] <cone-425> ffmpeg 03Michael Niedermayer 07master:b62297805b5f: avcodec/snowenc: add intra_penalty AVOption [00:00:00 CEST] --- Sun May 10 2015

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ffmpeg.log.20150509
by burek 10 May '15

10 May '15

[04:41:28 CEST] <baffled> Good evening folks. Could someone please tell me what package I need to install on ubuntu for ffmpeg. Aptitude search ffmpeg doesn't come up with anything obvious. [04:42:46 CEST] <c_14> What version? (of ubuntu) [04:46:09 CEST] <baffled> Um 14.04.2 so it seems c_14 [04:46:57 CEST] <BtbN> I don't think 14.04 already has ffmpeg. [04:47:00 CEST] <c_14> You'll have to get a static build or from a ppa [04:47:07 CEST] <BtbN> the ppa is gone iirc [04:47:29 CEST] <c_14> There's a ppa that provides static builds [04:47:34 CEST] <c_14> The link on the dl page was updated [04:48:06 CEST] <baffled> Ah, I'll check thanks. [05:06:02 CEST] <onthemic> hi I am getting an av_interleaved_write_frame(): Broken pipe error in ffmpeg [05:06:13 CEST] <onthemic> http://pastebin.com/tGbjUXbJ [05:29:52 CEST] <baffled> c_14: et al, that seems to have worked. Thank you very much! [12:33:02 CEST] <JackRstream> hi all [12:36:03 CEST] <JackRstream> got an issue with the following CL ffmpeg.exe -i http://localsource:8000/live -map 0 -c copy -id3v2_version 3 -c:a libmp3lame -b:a 128k -write_id3v1 1 icecast://source:password@localhost:8000/128k.mp3 [12:36:20 CEST] <JackRstream> the metadata are not copied to the 128k.mp3 audio steam [12:36:23 CEST] <JackRstream> any idea ? [13:28:05 CEST] <Guest6078> Hi, i need to add a time stamp overlay to a video that has milliseconds this works ffmpeg -f dshow -i video="Microsoft LifeCam Cinema" -vf "drawtext=expansion=strftime:fontfile=arial.ttf:fontcolor=red:fontsize=23:x=545:y=7:text='%%H\:%%M\:%%S'" -crf 18 -preset ultrafast .\camera_out.mkv but %%L which as i understand it is for milliseconds doesn't and localtime doesn't seem to work at all on windows. [13:34:57 CEST] <Guest6078> I should have been more clear it still records when i add %%L or %%3 to show milliseconds but the time stamp no longer shows and i get a wall of real-time buffer [Microsfot LifeCam Cinema] [video input] too full or near too full <1001% of size 3041280 [rtbufsize parameter]>! frame dropped [14:18:25 CEST] <AlexRussia> Hey [14:18:45 CEST] <AlexRussia> Can I somehow convert flac to mp3 with saving tags?(I mean, song name, author name etc) into metadata of file? [14:23:02 CEST] <AlexRussia> oh, sorry [14:23:05 CEST] <AlexRussia> My mistake :D [15:40:06 CEST] <dijax> Hi! I have folder with video files how can i convert all to audio files? [17:15:19 CEST] <abique> Hi [17:21:33 CEST] <abique> I'm doing a little artistic project where I use cairo to draw into an image surface and then I'd like to encode each frame into a video using ffmpeg. This is my current video encoder: https://github.com/abique/kaleielak/blob/master/src/video-encoder.cc [17:21:58 CEST] <abique> My issue is that I don't know how to choose the codec and I'm having issue with the color space as well [17:22:30 CEST] <abique> baptiste, Hello is it you from SmartJog? :) [18:20:51 CEST] <abique> I got it working :) [19:19:10 CEST] <trodis_> im always getting Segmentation fault (core dumped) when i to play the ffserver stream with ffplay. on the same machine playing the stream with mplayer works without problem [23:37:40 CEST] <Cobra_Fast> Hi! I'm encoding a h264 video with -crf and -maxrate/-bufsize, but while encoding q will not go better than 23 although i've set crf to 18. what may i be doing wrong? [00:00:00 CEST] --- Sun May 10 2015

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ffmpeg-devel.log.20150508
by burek 09 May '15

09 May '15

[00:11:41 CEST] <cone-940> ffmpeg 03Michael Niedermayer 07master:3b1233539509: swscale/x86/hscale_fast_bilinear_simd: Remove ancient out-commented debug code [01:05:59 CEST] <cone-940> ffmpeg 03wm4 07master:72e7553125e6: pngdec: set correct range [01:19:19 CEST] <cone-940> ffmpeg 03wm4 07master:b697b297b77f: id3v2: do not export APIC description if empty [02:25:59 CEST] <yvear> does anyone know why libdcadec won't work on individual streams? does this bug have a ticket? http://pastebin.com/aw0H0vuB [02:27:54 CEST] <yvear> only works by -c:a libdcadec -i ... which is undesirable b/c it decodes all streams as dts [02:54:36 CEST] <jamrial> yvear: if with -c:a:0 libdcadec it doesn't use libdcadec at all during decoding then yeah, that's probably a bug [02:54:39 CEST] <jamrial> open a ticket for it [02:55:27 CEST] <jamrial> nonetheless, you could recompile with --disable-decoder=dca so libdcadec is the only decoder available [02:56:40 CEST] <rcombs> perhaps libdcadec should be positioned before dca in allcodecs.c [02:56:58 CEST] <rcombs> (IIUC that would make it be used by default, which seems sane enough to me) [02:57:10 CEST] <rcombs> (if you built ffmpeg with it, it figures you probably want it used) [03:01:11 CEST] <yvear> rcombs, that would be great yes [03:12:07 CEST] <BBB> wow kostya is really angry dude [03:12:15 CEST] <BBB> Im reading some of his more recent blog posts [03:12:21 CEST] <BBB> hes like the devil [03:18:46 CEST] <rcombs> https://gist.github.com/16146a3c86bac5ff20fc anybody wanna check my channel mappings before I send this [03:23:43 CEST] <atomnuker> uuh that function looks bad [03:23:57 CEST] <atomnuker> what if a file has 7 channels [03:24:45 CEST] <atomnuker> then s->channel_configuration would get set to 0 [03:25:10 CEST] <rcombs> atomnuker: yeah, and then we'd refuse to encode [03:25:34 CEST] <atomnuker> int channel_configuration; looks redundant [03:25:37 CEST] <rcombs> atomnuker: though in theory we should be able to proceed and write a PCE indicating the channel layout in that case [03:26:27 CEST] <rcombs> and channel_configuration should be based on channel layout, not on channel count [03:27:23 CEST] <atomnuker> yep [03:27:53 CEST] <rcombs> but I think that change belongs with adding PCE support [03:29:35 CEST] <atomnuker> though since s->channel_elements only gets used when writing to the bitstream it'll probably be better to move everything related to put_audio_specific_config() [03:30:27 CEST] <atomnuker> wait, no, nevermind, better to have a seperate variable [03:32:56 CEST] <rcombs> I figured I'd make channel_configuration its own member to keep track of what we're actually writing to channel_configuration [03:33:17 CEST] <rcombs> and if it's 0 then, as per the spec, we should write a PCE with the actual channel layout [03:33:31 CEST] <rcombs> (but since we don't support that yet we bail) [03:36:41 CEST] <atomnuker> I might get around to doing that in the summer, so yeah, that variable might get used in the future [03:43:11 CEST] <atomnuker> keeping the function but putting an assert when channels == 7 might be the best for now [03:47:14 CEST] <rcombs> atomnuker: eh, channels == 7 doesn't indicate a logic error, it indicates the user passed in something bad [03:47:25 CEST] Action: rcombs -> out for a bit [03:50:46 CEST] <atomnuker> yeah, a check and an error message is what I meant [03:51:20 CEST] <yvear> @jamrial, what bug priority should this be? it should be a quick fix [04:38:06 CEST] <jamrial> yvear: normal, unless it's a regression, in which case important [05:05:10 CEST] <rcombs> can anyone with a 7.1-channel setup confirm my channel mappings are correct, though? [05:06:32 CEST] <rcombs> http://puu.sh/hFu7K/a1cc82806f.mkv [12:01:15 CEST] <kierank> Daemon404: how's it going [12:13:02 CEST] <wm4> is today smpte troll day [13:44:27 CEST] <j-b> 'morning [14:25:20 CEST] <cone-786> ffmpeg 03hSÇ 07master:b6d8afd8208a: configure: replace arch loongson with arch extra list loongson [14:34:37 CEST] <cone-786> ffmpeg 03hSÇ 07master:ba1f56ae9bac: configure: remove loongson check inline asm and mips dependent [15:39:38 CEST] <cone-786> ffmpeg 03Ronald S. Bultje 07master:c97d30f02f34: vp9: de-duplicate some functions that are identical between 10/12 bpp. [16:06:12 CEST] <haasn> https://www.ffmpeg.org/doxygen/2.5/pixfmt_8h.html#ad4791ea14975f098b649db7f… [16:06:21 CEST] <haasn> why does ffmpeg specify both AVCOL_TRC_BT2020_10 and AVCOL_TRC_BT2020_12? [16:06:39 CEST] <haasn> There's just one BT.2020 gamma curve, the specification just suggests rounded values with good enough precision for 10 and 12 bit applications [16:06:44 CEST] <nevcairiel> probably because some video standard made a difference there [16:06:55 CEST] <haasn> There was an update to BT.2020 about a month ago clarifying this [16:07:23 CEST] <haasn> https://0x0.st/jz.png [16:09:15 CEST] <nevcairiel> this enum is supposed to match the values used in h264/h265, and they define two entries for BT.2020, with index 14 and 15 [16:10:27 CEST] <nevcairiel> which according to the h265 standard is really just the same as 709 or 601 [16:10:53 CEST] <haasn> Then the H.264/H.265 specifications don't really make sense after the update [16:11:15 CEST] <haasn> The unique combination of gamma curve + bit depth already uniquely identifies the exact constants [16:11:51 CEST] <nevcairiel> fact remains, our enum wants to mirror the exact values used in h264/265, so it has to include both [16:20:17 CEST] <nevcairiel> who the f' uses gray decoding [16:31:26 CEST] <haasn> nevcairiel: Interesting, the update of the H.265 standard eliminates this ambiguity [16:32:46 CEST] <haasn> It now defines 1, 6, 14 and 15 as all being aliases for the one true curve, and uses the exact numbers in its definition [16:33:52 CEST] <wm4> nice, more values that mean the same [16:34:11 CEST] <haasn> exactly [16:34:58 CEST] <haasn> more aliases which we all ignore in mpv because it's irrelevant information either way :p [18:10:04 CEST] <cone-786> ffmpeg 03Giorgio Vazzana 07master:23e6cf832ff6: lavd/v4l2: fix typo [20:01:16 CEST] <jamrial> ^ maybe rf64 should be set to "auto" as default [20:31:28 CEST] <BBB> haasn: I dont think theres a single user that uses these values for display purposes [20:31:36 CEST] <BBB> haasn: or if there is, I would love to see what they do with it [20:31:48 CEST] <BBB> (and no, printing that text to a label is not a display purpose :) ) [20:32:01 CEST] <BBB> </ate> [20:32:03 CEST] <BBB> </late> [20:33:48 CEST] <haasn> BBB: We use them for display purposes in a big cascading switch block that has case AVCOL_TRC_BT709: case AVCOL_TRC_BT2020_10: case AVCOL_TRC_BT2020_10: /*etc*/ return MP_TRC_BT1886; [20:34:29 CEST] <BBB> do you use MP_TRC_BT1886 to setup the monitor/window? [20:34:30 CEST] <haasn> Because every standard that uses those for OETF (BT.601, BT.709, BT.2020) specifies or is covered by BT.1886's output transfer [20:34:51 CEST] <haasn> BBB: we use it to set up the mapping from source (YUV) to display [20:35:37 CEST] <BBB> Im positively shocked [20:35:41 CEST] <BBB> (thats a compliment :) ) [20:35:53 CEST] <BBB> I always assumed clients just ignored that stuff [20:35:57 CEST] <BBB> (like chrome does...) [20:36:02 CEST] <haasn> No, we use it if it's there [20:36:10 CEST] <haasn> Most of the time it isn't, though, so we have to have a very good guessing algorithm :/ [20:38:19 CEST] <BBB> default: return MP_TRC_BT1886 [20:42:10 CEST] <cone-786> ffmpeg 03Michael Niedermayer 07master:982e7bbfa6b5: avcodec/vc1: Skip chroma operations if CODEC_FLAG_GRAY is set [23:10:43 CEST] <cone-786> ffmpeg 03Giorgio Vazzana 07master:28f20d2ff487: lavd/v4l2: produce a 0 byte packet when a dequeued buffer is flagged with V4L2_BUF_FLAG_ERROR [00:00:00 CEST] --- Sat May 9 2015

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ffmpeg.log.20150508
by burek 09 May '15

09 May '15

[00:17:03 CEST] <quantum_sm> hello, I had tried to use this command drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h-line_h:x=-50*t" but the text starts in the left. how I can change in order text to start from the right ? [00:28:01 CEST] <llogan> quantum_sm: w-tw-(50*t) or w-(50*t) depending on what you want [00:36:53 CEST] <quantum_sm> llogan: thank you that worked. Also it is a possibility to start the text after 20 seconds for example ? [00:45:40 CEST] <llogan> quantum_sm: http://ffmpeg.org/ffmpeg-filters.html#Timeline-editing [00:47:47 CEST] <quantum_sm> llogan: what i'm missing here Error initializing filter 'drawtext' with args 'enable=between(t,12,3*60):fontsize=25 [00:55:39 CEST] <quantum_sm> fflogger: here is the output http://pastebin.com/PvEGcxXA [00:56:36 CEST] <llogan> ffmpeg version 0.10.7 [00:56:39 CEST] <llogan> ancient and unsupported [00:57:03 CEST] <llogan> get a static build or compile [00:57:04 CEST] <llogan> http://johnvansickle.com/ffmpeg/ [00:57:10 CEST] <llogan> https://trac.ffmpeg.org/wiki/CompilationGuide/Centos [01:01:12 CEST] <quantum_sm> so that enable=between paramater in only in newer versions ? [01:01:19 CEST] <llogan> yes [01:01:43 CEST] <llogan> use a build from current git master [01:10:29 CEST] <quantum_sm> ok thank you [01:32:51 CEST] <yvear> can I set decoder for ffprobe? [07:53:44 CEST] <Dark-Bot> hello is anyone here? [07:54:32 CEST] <Dark-Bot> here is my error message http://pastebin.com/tGbjUXbJ [07:54:58 CEST] <Dark-Bot> am seeking advice on troubleshooting [07:55:50 CEST] <Dark-Bot> av_interleaved_write_frame(): Broken pipe [07:57:12 CEST] <Dark-Bot> i will try again tomorrow, thank you [09:22:23 CEST] <gr1sha> I can't get dranger's tutorial1.c to compile [09:22:29 CEST] <gr1sha> on Ubuntu 13.10 [09:44:09 CEST] <ffmpeguser> Hi [09:46:16 CEST] <ffmpeguser> Why sometimes video and audio duration difference when trim with -ss and -t option? [09:50:23 CEST] <Mavrik> ffmpeguser, if you're not transcoding, it's cut on GOP/audio packet boundaries [09:50:38 CEST] <Mavrik> which usually don't correspond [09:53:19 CEST] <ffmpeguser> I encoded to libx264 and ac3 (mkv to mp4) [09:54:15 CEST] <ffmpeguser> I understand duration difference when use vcodec copy but it was not [10:24:15 CEST] <termos> I get a floating point exception on this line https://github.com/FFmpeg/FFmpeg/blob/9d4fdfe24c731d1880797dee65365154b41c1… I'm guessing because it's trying to divide by 0. But I see that AVStream::pts is marked as deprecated, but here it is being used [11:47:08 CEST] <sybariten> Anoia: i'm partly in *nix (Red Hat) and partly Win7. More often linux. [11:51:02 CEST] <Anoia> ffmpeg $(cat ffmpegoptions) [11:51:03 CEST] <Anoia> ? [11:51:31 CEST] <Anoia> not sure how to make it mor inilike without passing it through a text preprocessor [12:00:40 CEST] <hta> Q: Does anyone want to hear about issues with the motion-jpeg en/decoder in ffmpeg? [12:12:40 CEST] <k_sze> ffmpeg fails so hard at basic consistency it's not funny. [12:39:40 CEST] <durandal_1707> k_sze: tell more [12:40:42 CEST] <k_sze> raw video data of same pixel format, same frame rate, same resolution, piped into ffmpeg for encoding as h.264 and muxing into a .nut file. [12:41:08 CEST] <k_sze> the timestamps are sometimes offset by two frames, sometimes not. [12:42:38 CEST] <k_sze> literally, at 30 fps, sometimes the first encoded frame gets a timestamp of 0.06666666 second. [12:42:42 CEST] <k_sze> sometimes 0.0 second [13:37:46 CEST] <seasc> just asking as i dont know, regarding last question, doesnt the FPS only 'apply properly' while recording. I mean, once a video has a given set of images per second, can ffmpeg create interpolated images for between the existing ones? [13:38:22 CEST] <seasc> i dont think so, correct me if im wrong [13:41:55 CEST] <sybariten> Anoia: hm. thats one way i suppose. Quite homemade. The optimal for me, i'm talking purely wishing here, would be if it could take an option -ini or something and then a "webPublish.in" could contain rows that defined the encoding ... like bitrate=3200 , etc [13:42:16 CEST] <sybariten> and then you could easily comment rows with a # [13:43:15 CEST] <seasc> sybariten, linux? [13:43:55 CEST] <seasc> sybariten, https://github.com/sri-arjuna/vhs [13:44:39 CEST] <sybariten> seasc: yeah i'm on both linux and Win, but more often linux ... but i do run bash on win too, so i should manage [13:45:33 CEST] <seasc> sybariten, vhs -Q fhd input.... using an external file wich contains a table for most resolutions, and their bitrates used for audio/video within the script [13:46:21 CEST] <seasc> sybariten, looks like: https://raw.githubusercontent.com/sri-arjuna/vhs/master/screenshots/example… [13:46:49 CEST] <sybariten> seasc: interesting...! [13:46:58 CEST] <sybariten> although i dont really understand the first example (for instance :) [13:47:08 CEST] <sybariten> i mean, wheres the output filename defined, for instance [13:48:12 CEST] <sybariten> ah, looked at the image now... is that curses?` [13:48:14 CEST] <seasc> say you have a 50GB FHD video, but want it smaller in size. -> vhs -Q fhd inputfile1 input2 <-- will take passed inputfiles one by one, and encode them the the 'fhd' settings [13:48:24 CEST] <seasc> its TUI, my own thing [13:48:39 CEST] <seasc> pulled in when using vhs the first time [13:49:51 CEST] <sybariten> FHD? TUI? im guessing this is vhs thing is something one has to try out a bit to get the grip of [13:50:06 CEST] <seasc> fhd = full hd [13:50:08 CEST] <sybariten> ok [13:50:58 CEST] <sybariten> so is "fhd" in this case an internal saved setting name, or an actual settings _file_? [13:51:47 CEST] <seasc> its the identifier in an external file in ~/.config/tui/presets, telling vhs wich resolution and bitrates to use [13:51:49 CEST] <seasc> http://forums.fedoraforum.org/showthread.php?t=299182 [13:52:06 CEST] <seasc> erm... ~/.config/vhs/presets :p [13:52:26 CEST] <seasc> so you can change to your own prefered bitrates and such [13:52:56 CEST] <seasc> there's also a file containing the container codecs and their file extension [13:53:24 CEST] <seasc> So i can pass "-e mp5" which will make mp4 files using the x265 coded (if ffmpeg is built with that) [13:53:35 CEST] <seasc> just as example [13:55:47 CEST] <seasc> sybariten, TUI is just like ncurses, to prove an interface on for the console [13:56:57 CEST] <sybariten> aha [13:57:06 CEST] <sybariten> and you built that too? [13:57:48 CEST] <seasc> yah, me needed soemthing to display the scripts i wanted to write :D [14:01:15 CEST] <sybariten> heh ok.... [14:01:30 CEST] <sybariten> soooo.... how does Da System see all this.... how is vhs installed? [14:02:24 CEST] <seasc> Are you on a redhat system? [14:02:55 CEST] <seasc> then i'd have a repositriy, otherwise either download the zipfile or use git to clone the code. [14:03:11 CEST] <seasc> then run the install.sh [14:03:21 CEST] <sybariten> ok, and what does it rely on... is it executables? or script? [14:03:30 CEST] <sybariten> in other words, on win, am i shit unluck? [14:03:49 CEST] <seasc> i have no idea if it works in win-bash [14:04:19 CEST] <sybariten> but its theoretically just bash reliant? And wants ffmpeg ? [14:04:38 CEST] <seasc> it needs also awk, grep and sed at times [14:04:55 CEST] <sybariten> oh .. well i have those too, in bash... [14:05:01 CEST] <seasc> well, tui does.... but.. as far i can think of... yes [14:05:03 CEST] <sybariten> i'll have to try it out [14:05:24 CEST] <sybariten> sounds intresting... but, is this like a general concept, have you done other "helper" things? [14:05:53 CEST] <seasc> what do you mean by that? [14:06:25 CEST] <sybariten> is this TUI thing something you use for other systems, like a wrapper thing for , i dunno, imagemagick or whatever [14:06:48 CEST] <sybariten> imagemagick is kinda similar to ffmpeg in some ideas, i suppose. [14:07:10 CEST] <sybariten> coffeee brb [14:07:17 CEST] <seasc> TUI is a set of... 44'ish commands to ease scripting and make the visual output cool looking. One could write tons of other helpers [14:07:39 CEST] <seasc> i'm stuck with complex things such as ffmpeg and developing environments [14:07:48 CEST] <seasc> stuck as in fully occoupied :p [14:08:11 CEST] <seasc> but yes, one could write a handler for imagemagick [14:08:19 CEST] <seasc> but/so [15:03:39 CEST] <sybariten> seasc: ok! [15:20:57 CEST] <WereCatf> How can I transcode the video and audio streams in an MPEG-TS - file captured from DVB-C to another MPEG-TS - file while keeping EPG and program names and numbers intact and other supplemental streams intact? [15:25:59 CEST] <seasc> WereCatf, sounds like to copy the file would be less hazzle? [15:26:29 CEST] <seasc> WereCatf, or are you streaming from DVB-C? [15:26:47 CEST] <WereCatf> I wouldn't be asking about transcoding if I just wanted to copy the file. [15:28:11 CEST] <seasc> its just that i know (understand) transcode from (as) one container to another. In the result of your requirements, you simply copy all the streams into the same container came codecs same file extension [15:28:52 CEST] <WereCatf> No, I would be transcoding the audio and video streams to use another codec [15:29:13 CEST] <seasc> you want it to remain MPEG-TS ? [15:29:16 CEST] <WereCatf> Yes [15:29:25 CEST] <seasc> Its the same codec, or use another container [15:29:49 CEST] <seasc> AFAIK* [15:29:54 CEST] <WereCatf> No, you can use e.g. H.264 in an MPEG-TS. I would know, I have such files captured from DVB. [15:30:56 CEST] <seasc> then it'll be something like: ffmpeg -i file_in -map 0 -cv x264 -f mpeg-ts file_out [15:31:19 CEST] <WereCatf> But that strips out all the EPG, changes program names and numbers and such. [15:31:44 CEST] <seasc> What is the EPG? [15:32:08 CEST] <WereCatf> http://en.wikipedia.org/wiki/Electronic_program_guide [15:33:53 CEST] <seasc> Could you provide please (on pastebin): ffmpeg -psrn -i file [15:34:34 CEST] <seasc> me not familiar with rpg yet, so i need a clue :) [15:34:45 CEST] <seasc> epg* [15:35:32 CEST] <WereCatf> That command doesn't list anything regarding EPG anyways [15:35:45 CEST] <seasc> ok [15:36:20 CEST] <seasc> then me sorry, good luck [16:47:19 CEST] <hendry> anyone managed to get CineForm decoding working on Linux? [17:04:39 CEST] <lordkrondor> l [17:14:24 CEST] <raytiley_> when do a transcode with ffmpeg is the "time" property in the output the time in the file, or how long the transcode has been going for? [17:16:02 CEST] <c_14> time in the file [17:16:24 CEST] <raytiley_> thanks [17:48:48 CEST] <miau> hello, how doi install ffmpeg on linux mint 17? i tried some older guides but this didnt work, because it doesnt find the repo [18:14:33 CEST] <dericed> hendry: no cineform decoder yet afaik, there's an open ticket for it [23:52:26 CEST] <tp_> anyone successfully used ffmpeg to transcode a video while its being watched in html5 with chrome or firefox? [23:59:08 CEST] <tp_> (h.264) [00:00:00 CEST] --- Sat May 9 2015

1 0

ffmpeg-devel.log.20150507
by burek 08 May '15

08 May '15

[00:30:19 CEST] <kierank> do I dare suggest dropping hscale_fast_bilinear_simd [01:06:27 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:0c015aa95c2d: avcodec/tableprint_vlc: Do not define functions to nothing [01:06:28 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:374cf93548d1: avformat/asfdec: do not define print_guid() to nothing [01:06:29 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:3e14ed05f1de: avformat/internal: always check arguments of hex_dump_debug() [01:13:31 CEST] <BBB> kierank: but its runtime generated simd!!112 [01:13:38 CEST] <BBB> kierank: how often have you seen that :-p [01:20:06 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:223a85985393: swresample/dither_template: Do not define macro functions to nothing [01:50:22 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:bc6f84fff741: avdevice/dshow_capture: avoid #ifdef, use ff_dlog() for dshowdebug() [02:57:45 CEST] <cone-461> ffmpeg 03Carl Eugen Hoyos 07master:b453e7684279: lavc/qdrw: Also support real-world qdraw images. [02:57:46 CEST] <cone-461> ffmpeg 03Carl Eugen Hoyos 07master:2279de6eca12: lavf/img2dec: Autodetect qdraw images. [02:57:47 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:5f2b4a2f6a13: Merge remote-tracking branch 'cehoyos/master' [04:12:12 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:21b30947f2ca: swscale/x86/hscale_fast_bilinear_simd: Fix variable names in comments [05:18:05 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:9253cc496a50: avcodec/diracdec: Replace default_bsep[] by multiplication [09:00:04 CEST] <yvear> does we support dts hd ma decoding now? [09:00:12 CEST] <yvear> *do [10:09:14 CEST] <yvear> if so, it's not working for me. it's not detecting any channel > 6, being decoded as 5.1 is all it will do [10:09:22 CEST] <yvear> can anyone confirm? [10:10:15 CEST] <Gopu> hello can i send the patch as a attachement(git format-patch )? [10:34:46 CEST] <wm4> Gopu: yes [10:35:10 CEST] <nevcairiel> we prefer git send-email though [10:46:04 CEST] <Gopu> OK Thanks [12:02:40 CEST] <BBB> michaelni: do you want new patchset on ML or is on github only OK for you? [13:07:50 CEST] <michaelni> BBB, the new vp9 fate tests fail on qemu mips [13:08:05 CEST] <BBB> Im shocked :-p [13:08:23 CEST] <BBB> I assure you I didnt write mips code [13:09:06 CEST] <BBB> do you have any furhter info? Like a backtrace or something? [13:09:14 CEST] <michaelni> wait, let me retest, i maybe did something silly [13:09:40 CEST] <michaelni> but it was wrong results no crash [13:09:49 CEST] <BBB> hm... [13:10:12 CEST] <michaelni> no, still are failing [13:10:21 CEST] <BBB> confused [13:10:56 CEST] <BBB> I dont have qemu [13:11:04 CEST] <BBB> do you know which patch in the set causes it to fail? [13:11:30 CEST] <BBB> or wait, you said new ones [13:11:35 CEST] <BBB> so the existing ones are fine? [13:11:41 CEST] <BBB> its because its BE [13:11:44 CEST] <BBB> the test results are LE [13:11:45 CEST] <michaelni> these are failung: fate-vp9p2-20-10bit-yuv420, fate-vp9p2-20-12bit-yuv420 fate-vp9p3-20-10bit-yuv422 fate-vp9p3-20-10bit-yuv444 fate-vp9p3-20-12bit-yuv422 fate-vp9p3-20-10bit-yuv440 fate-vp9p3-20-12bit-yuv444 fate-vp9p3-20-12bit-yuv440 [13:11:48 CEST] <BBB> and Im not converting it [13:11:54 CEST] <BBB> let me see how h264/hevc do that [13:12:31 CEST] <michaelni> yes BE/LE is likely, ist what its most of the time when mips fails [13:14:31 CEST] <BBB> testing a fix... [13:20:07 CEST] <BBB> michaelni: fix pushed to github [13:20:18 CEST] <BBB> (basically force -pix_fmt, similar to what h264 tests do) [13:34:27 CEST] <cone-277> ffmpeg 03Ronald S. Bultje 07master:98f7f0f7e83f: lavu: add yuv440p10/12 pixfmts. [13:34:28 CEST] <cone-277> ffmpeg 03Ronald S. Bultje 07master:57f970a7046f: lavc: add yuv440p10/12 formats to aligned pixfmt list. [13:34:29 CEST] <cone-277> ffmpeg 03Ronald S. Bultje 07master:711d8812adc1: swscale: add yuv440p10/12 pixfmts. [13:34:30 CEST] <cone-277> ffmpeg 03Ronald S. Bultje 07master:5c600d74aa4d: fate: add/update reference files for 440 addition. [13:34:31 CEST] <cone-277> ffmpeg 03Ronald S. Bultje 07master:01e59d48ed1a: vp9: add profile 2/3 to exported profiles. [13:34:32 CEST] <cone-277> ffmpeg 03Ronald S. Bultje 07master:346ce5da197b: vp9: parse profile 2/3 bitdepth in frame header. [13:34:33 CEST] <cone-277> ffmpeg 03Ronald S. Bultje 07master:b224b165cbbb: vp9: add keyframe profile 2/3 support. [13:34:34 CEST] <cone-277> ffmpeg 03Ronald S. Bultje 07master:23ba4538f9d7: vp9: add inter-frame profile 2/3 suport. [13:34:35 CEST] <cone-277> ffmpeg 03Ronald S. Bultje 07master:b8077d7a3b7b: vp9: add profile 2/3 fate tests. [13:34:36 CEST] <cone-277> ffmpeg 03Ronald S. Bultje 07master:2293ec6ab300: libvpxdec: add 440 pixfmts. [13:34:37 CEST] <cone-277> ffmpeg 03Michael Niedermayer 07master:b28d5c49f7e5: Merge remote-tracking branch 'rbultje/vp9-profile23-wip' [13:48:13 CEST] <BBB> \o/ [14:01:57 CEST] <Compn> another pixfmt ! :P [14:01:57 CEST] <Compn> ehe [14:12:42 CEST] <kierank> Compn: https://github.com/bcoudurier/FFmbc [14:21:50 CEST] Action: Compn does not know the significance of ffmbc on github [14:27:56 CEST] <kierank> there is git now [14:27:59 CEST] <kierank> there was not in the past [14:29:13 CEST] <nevcairiel> now that we actually have a clear changelog .. how does that LGPL -> GPL rule go .. I can just fork LGPL and make it GPL, but does that then truely prohibit any backporting to the LGPL project? because if it does, someone should really teach the fsf that they are idiots [14:32:10 CEST] <michaelni> Daemon404, is "0507 11:17 Gopu Govindaswa ( 10K) [FFmpeg-devel] [PATCH] avcodec/libx265: use x265 Multi-library Interface to query the API" ok to apply ? [14:32:20 CEST] <thardin> nevcairiel: ask a lawyer [14:33:25 CEST] <thardin> but of course the author of any patches to a project are free to choose whichever license they want, as long as it doesn't clash with the license of the original code [14:34:02 CEST] <nevcairiel> but the "original" code was LGPL, someone just invoked this one rule that you can also license LGPL code as GPL [14:34:21 CEST] <thardin> so take the code from the original lgpl source [14:34:42 CEST] <nevcairiel> the point is about backporting patches inside this GPL project [14:35:50 CEST] <thardin> hm.. is there line/character grained licensing? [14:38:40 CEST] <thardin> this also reminds me of how libav tends to take patches from ffmpeg, rewrite them and claim ownership (like some of my mxfdec patches) [14:38:59 CEST] <thardin> or rather, git changes their ownership (I think) [14:39:09 CEST] <nevcairiel> its easy to blame git [14:39:12 CEST] <BBB> nevcairiel: yes you can do that [14:39:18 CEST] <BBB> nevcairiel: and yes its obviously bad faith to do so [14:39:25 CEST] <nevcairiel> but its even easier to keep ownership intact when porting patches =p [14:39:32 CEST] <BBB> but I mean, were talking about bcoudurier here [14:39:33 CEST] <BBB> so... [14:40:23 CEST] <BBB> and the reimplementing of ideas between ffmpeg and libav is just a massive waste of manpower [14:40:31 CEST] <BBB> rgardless of the legal implications (or lack thereof) [14:42:35 CEST] <kierank> but.but.but code cleanliness [14:42:50 CEST] <thardin> muh indentation [14:43:21 CEST] <thardin> all I know is dual review processes are a burden [14:54:50 CEST] <cone-277> ffmpeg 03Michael Niedermayer 07master:a6153977df6d: avcodec/vp9dsp: Replace assert by av_assert0() [14:54:51 CEST] <cone-277> ffmpeg 03Michael Niedermayer 07master:cc77bb09e430: avcodec/x86/vp9dsp_init: Fix mix of declaration and statement [15:12:47 CEST] <BBB> kierank: so supposedly, code cleanliness leads to better end user experinece, right? I mean, eventually [15:13:00 CEST] <BBB> like, its faster, less bugs, less security issues, more features, easier development [15:13:01 CEST] <BBB> irght? [15:13:06 CEST] <BBB> so, if thats true [15:13:16 CEST] <BBB> then how come libavs vp9 decoder is a piece of shit compared to ffmpeg's? [15:13:43 CEST] <BBB> because clearly, libavs is much better indented than ffmpegs [15:58:42 CEST] <nevcairiel> that whole premise only works when you actually have manpower left to do all these things after the indenting =p [15:59:54 CEST] <BBB> admittedly, its a lot of reindenting [16:00:19 CEST] <BBB> fortunately, indenting adds massive value to your ohloh linecount stats [16:37:15 CEST] <cone-277> ffmpeg 03James Almer 07master:aa70801aaf40: ripemd: move ripemd{256, 320} into separate functions [16:53:39 CEST] <cone-277> ffmpeg 03Shivraj Patil 07master:7174df44fe7b: avcodec/mips: MSA (MIPS-SIMD-Arch) optimizations for HEVC uni copy, uni horizontal and uni vertical mc functions [16:54:55 CEST] <nevcairiel> i'm somewhat fascinated that someone really wants to decode hevc on a mips device [16:55:06 CEST] <nevcairiel> arent they all butt-slow mini SoCs [16:55:32 CEST] <wm4> yeah, that seems strange [16:55:45 CEST] <wm4> maybe some bad interacting with the marketing dept.? [17:07:01 CEST] <yvear> can anyone confirm 7.1 dts hd ma decoding works in the current build? only reading as 5.1 for me [17:08:05 CEST] <nevcairiel> you need to pass an option to enable it, ie "-disable_xll 0" or something like that, i forgot the exact wording, but apart from that its also not quite perfect [17:08:16 CEST] <nevcairiel> if you want a more reliable decoding, build ffmpeg with libdcadec instead [17:08:41 CEST] <wm4> also, if you want to use the libdcadec ffmpeg wrapper, you need to explicitly open that decoder, since the builtin one is preferred [17:08:58 CEST] <nevcairiel> or do what i do, and disable it at built-time =P [17:10:10 CEST] <yvear> wm4, +nevcairiel, my windows build says --enable-libdcadec [17:10:55 CEST] <nevcairiel> then you need to pass something like -c:a libdcadec to make use of it [17:11:03 CEST] <yvear> even ffprobe reads as 5.1, doesn't recognize 7.1 [17:11:07 CEST] <yvear> ok [17:13:29 CEST] <wm4> yvear: libavformat might still use the builtin decoder, but you can simply ignore the info it exports, and use what you get in the AVFrame instead [17:13:55 CEST] <nevcairiel> that reminds me that I was meaning to send my patch for the dts parser [17:23:14 CEST] <yvear> +nevcairiel, wm4, ah, ok it still defaults to old decoder. must do ffmpeg -c:a libdcadec -i file.dts outfile when will it default to new decoder? [17:24:06 CEST] <wm4> probably never, because ffmpeg always prefers internal decoders over external ones [17:24:41 CEST] <j-b> Do we have test cases for the failures? [17:24:50 CEST] <j-b> of the internal one? [17:25:38 CEST] <nevcairiel> i gave some to libav, and i stopped caring shortly after since i have a free, open, bitperfect and complete decoder now =p [17:27:13 CEST] <yvear> aww :( then will our internal decoder support full DTS-HD MA decoding soon? since we have the code and see how it's done [17:27:32 CEST] <nevcairiel> doubtful [17:31:31 CEST] <yvear> I mean, if there is code that works better and is open source, and we already use it as an optional decoder.. why do we not default to it? it couldn't be a license thing right? [17:31:56 CEST] <yvear> or is it? [17:49:05 CEST] <JEEBsv> yvear: so far it seems OK license-wise, and we do already have some kind of X > Y things regarding decoders/encoders, IIRC [17:57:16 CEST] <cone-277> ffmpeg 03Andreas Cadhalpun 07master:584cc1ade10a: aacsbr: break infinite loop in sbr_hf_calc_npatches [18:41:07 CEST] <yvear> hmm, is it possible to set decoder per stream? [18:46:44 CEST] <nevcairiel> yes [18:54:11 CEST] <yvear> +nevcairiel, with a .mkv containing h264, 7.1 dts-hd ma, and ac3, doing ffmpeg -c:a:0 libdcadec -i file.mkv only shows 5.1 dts. it doesn't apply decoder [18:54:41 CEST] <nevcairiel> ignore what it says, just check what it actually decodes [18:55:42 CEST] <nevcairiel> also check that the stream index is actually the correct one [19:04:52 CEST] <yvear> +nevcairiel, yes it's not decoding 7.1 with libdcadec just normal 5.1 [19:06:33 CEST] <yvear> +nevcairiel, it works when not in a container though [19:09:34 CEST] <cone-277> ffmpeg 03Tom Butterworth 07master:873d7e0e6326: avcodec/s3tc: fix decoding when dimensions are not a multiple of 4 [19:14:49 CEST] <Daemon404> michaelni, one change, and it needs to be fixed so it isnt malformed [19:15:03 CEST] <Daemon404> [16:47] <@muggs> Daemon404: looking at his last patch, you can remove the !api check after x265_api_get(0); when you request 0 it always returns an API structure pointer. [19:15:06 CEST] <Daemon404> [16:53] <@muggs> X265_BUILD 57 has the api request forwarding feature in it. [19:15:09 CEST] <Daemon404> ^ [19:20:16 CEST] <cone-277> ffmpeg 03Michael Niedermayer 07master:daea3209693f: avcodec/txd: Fix input size checks for dxt1/3 for dimensions % 4 != 0 [19:21:44 CEST] <michaelni> Daemon404, ok, please tell the patch author not me, or should i forward it to the ML ? [19:24:53 CEST] <Daemon404> michaelni, i told him during my first revieew... he ignored it [19:25:07 CEST] <Daemon404> ill reply after dinner today [19:25:10 CEST] <Daemon404> (was on a plane to berlin) [19:25:26 CEST] <Daemon404> actually, ill just fix it by hand later. [19:26:39 CEST] <michaelni> ok, thanks [22:54:48 CEST] <rcombs> oh hey, an fflogger bug [22:59:40 CEST] <wm4> rcombs: easiest way to check... write a patch, check if it works, if it does, look at the spec, if there's nothing suspicious, send a patch [23:02:19 CEST] <rcombs> wm4: I may or may not do that [23:03:54 CEST] <cone-940> ffmpeg 03Michael Niedermayer 07master:a7dd933b811b: avcodec: Add av_packet_side_data_name() [23:03:54 CEST] <cone-940> ffmpeg 03Michael Niedermayer 07master:5f0ebe865c42: ffprobe: Use av_packet_side_data_name() to find the side data name [23:03:54 CEST] <cone-940> ffmpeg 03Michael Niedermayer 07master:43e94d5af48b: avcodec/g2meet: Use init_get_bits8() [23:03:54 CEST] <cone-940> ffmpeg 03Michael Niedermayer 07master:b1b0baa3d6a3: avcodec/g2meet: Check init_get_bits8() return value [23:25:04 CEST] <cone-940> ffmpeg 03Tom Butterworth 07master:288dc5b4a118: avcodec/s3tc: fix alpha decoding when dimensions are not a multiple of 4 [23:36:08 CEST] <cone-940> ffmpeg 03James Almer 07master:e0a403e1c284: vp9: add missing changelog and APIchanges entries for new VP9 profiles [23:53:59 CEST] <rcombs> >Dependent substream decoding is not implemented [23:54:08 CEST] <rcombs> do we still need a sample of that^ [23:54:13 CEST] <rcombs> because I apparently have one [23:58:01 CEST] <cone-940> ffmpeg 03James Almer 07master:28eaf46da904: doc/APIchanges: fill missing versions and hashes [00:00:00 CEST] --- Fri May 8 2015

1 0

ffmpeg.log.20150507
by burek 08 May '15

08 May '15

[01:27:50 CEST] <Zeranoe> Has anyone had success building FFmpeg with libmfx using MinGW-w64 [05:12:46 CEST] <chankit> Hi, may I know who wrote the tiny_ssim.c in ffmpeg and how can I contact him/her? [05:17:04 CEST] <c_14> https://git.videolan.org/?p=ffmpeg.git;a=commit;h=064146480b22d541c1c2e0c3e… [05:17:18 CEST] <c_14> There's an email there, though whether or not they'll appreciate private email is another issue. [05:17:41 CEST] <c_14> If there's a bug, open an issue on trac, if you have a patch send it to the devel mailing list. [05:43:22 CEST] <almostworking> join #pydio [05:43:26 CEST] <almostworking> ooops, [08:13:33 CEST] <chankit> c_14: thanks. Will act appropriately [08:39:07 CEST] <mobad> i've got a dump of an interleaved rtsp stream that i'm trying to convert to something playable. i made a little parser to parse out the h264 and g726 streams in to separate files but then i lose the sync between vid/aud [08:39:50 CEST] <mobad> any idea on how i can combine them with the rtsp timestamps? (or just convert an interleaved dump to something else) [08:42:22 CEST] <LexSfX> can ffmpeg be used to do hardware encoding on ios? [08:42:33 CEST] <LexSfX> on, for example, an iphone [08:44:49 CEST] <Mavrik> LexSfX, check if videotoolbox support is in yet [08:45:10 CEST] <LexSfX> in ffmpeg? okay [08:45:15 CEST] <LexSfX> Mavrik: thanks [09:03:47 CEST] <yvear> does ffmpeg support dts hd ma decoding now? [09:28:15 CEST] <yvear> if so, it's not working for me. it's not detecting any channel > 6, being decoded as 5.1 is all it will do [09:30:30 CEST] <yvear> can anyone confirm? I'm on windows btw [10:02:38 CEST] <Gopu> hello can i send the patch as a attachement(git format-patch )? [10:19:35 CEST] <spaam> yes [10:19:57 CEST] <spaam> send it to the devel ml [10:22:11 CEST] <spaam> Gopu: http://ffmpeg.org/developer.html#Submitting-patches related [10:23:55 CEST] <Gopu> Spaam: thanks .., then i will attach my patch and send to ML [10:33:33 CEST] <kath> hello. i am trying to convert an mp4 file to mp3. using ffmpeg -i name.mp4 nam.mp3 .. However i get http://paste.debian.net/172453/ [10:37:17 CEST] <spaam> kath: you need to enable libmp3lame to do it [10:41:59 CEST] <kath> spaam: how do i do that [10:42:00 CEST] <kath> ? [10:49:15 CEST] <spaam> kath: how did you install ffmpeg? [11:01:25 CEST] <sybariten> mornin [11:59:54 CEST] <DrSlony> Hello, I'm using ffmpeg to make a video from a JPEG sequence. All the JPEGs fit in a 900x900 box but some are portrait, most are landscape. How do I tell ffmpeg to not stretch the portrait ones sideways, just to center them? [11:59:56 CEST] <DrSlony> ffmpeg -f image2 -start_number 001 -i %d.jpg -an -c:v libx264 -preset slower -crf 23 vid.mp4 [12:50:53 CEST] <sybariten> i wish ffmpeg could be run with a simple-to-read ini file as only argument [13:45:33 CEST] <Anoia> sybariten: what OS/environment? [16:45:12 CEST] <boxgamex> anyone know what the maximum -rtbufsize is? [16:45:56 CEST] <sopparus> hello [17:13:14 CEST] <sopparus> ive set up a live555server [17:13:21 CEST] <sopparus> am i supposed to be able to output to that? [17:13:56 CEST] <sopparus> like -f rstp rstp://ip:port/mystream.sdp [17:14:00 CEST] <sopparus> i get 405 errors [17:15:39 CEST] <sopparus> http://pastebin.com/ccpCXcVL [18:37:36 CEST] <jookiyaya> can you guys actually tell the difference between vc1 bluray vs h264 bluray [18:41:02 CEST] <sopparus> on a 100 inch 4000 euro projector you see all kind of stuff [18:41:10 CEST] <sopparus> but on a "normal" tv, no way [18:41:38 CEST] <sopparus> that doesnt mean I would see a difference [18:41:47 CEST] <sopparus> but I saw a few movies on a procejtor like that [18:42:09 CEST] <sopparus> and you could easily tell between a remux 30gb and a transcode to 20gb for example [18:48:25 CEST] <jookiyaya> i see [18:48:39 CEST] <jookiyaya> so is one better than the other even if you see the difference [18:48:47 CEST] <jookiyaya> between vc1 bluray vs h264 bluray [20:33:39 CEST] <yvear> how do I set decoder for an individual stream of input file? [20:50:50 CEST] <llogan> yvear: probably with -codec with a stream specifier http://ffmpeg.org/ffmpeg.html#Stream-specifiers [20:51:49 CEST] <yvear> llogan, before -i right? [21:05:32 CEST] <llogan> yvear: yes. as an input option [21:06:35 CEST] <yvear> llogan, with a .mkv containing h264, 7.1 dts-hd ma, and ac3 in that order, doing ffmpeg -c:a:0 libdcadec -i file.mkv only shows 5.1 dts. it doesn't apply decoder [21:07:31 CEST] <yvear> llogan, it works with -c:a libdcadec but then it treats the ac3 as dts... [21:08:17 CEST] <llogan> it doesn't decode it properly by default? [21:13:01 CEST] <DrSlony> Hello, I'm using ffmpeg to make a video from a JPEG sequence. All the JPEGs fit in a 900x900 box but some are portrait, most are landscape. How do I tell ffmpeg to not stretch the portrait ones sideways, just to center them? [21:13:03 CEST] <DrSlony> ffmpeg -f image2 -start_number 001 -i %d.jpg -an -c:v libx264 -preset slower -crf 23 vid.mp4 [21:13:41 CEST] <yvear> llogan, no it doesn't http://pastebin.com/VtVD8Ahy [21:18:42 CEST] <yvear> llogan, see what I mean? [21:19:09 CEST] <lakelord> Hi...in what repository for Ubuntu 14.04 can I find FFMPEG for install [21:19:34 CEST] <llogan> lakelord: https://launchpad.net/~mc3man/+archive/trusty-media [21:19:44 CEST] <llogan> or use a static build http://johnvansickle.com/ffmpeg/ [21:19:53 CEST] <llogan> or compile http://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu [21:19:59 CEST] <llogan> yvear: hold on [21:21:28 CEST] <llogan> yvear: i meant what happens if you don't use -codec? [21:21:29 CEST] <lakelord> llogan: it giver....Cannot add PPA: 'No JSON object could be decoded'. [21:22:14 CEST] <llogan> lakelord: you'll have to go to #ubuntu for that [21:22:36 CEST] <lakelord> llogan: ok...i figured out...I'm a dumbass [21:25:38 CEST] <lakelord> llogan: I added the repository....now ehat ? sudo apt-get install ffmpeg ?? that's it ? [21:26:33 CEST] <llogan> read the link i gave you [21:26:37 CEST] <_DragonsLordKERN> sudo apt-get update [21:26:45 CEST] <_DragonsLordKERN> sudo apt-get install ffmpeg [21:26:47 CEST] <_DragonsLordKERN> I guess [21:33:50 CEST] <llogan> DrSlony: -vf "pad=900:900:(ow-iw)/2:(oh-ih)/2" [21:36:05 CEST] <llogan> if you don't like the square box, then you can add a crop. pad,crop=iw:ih/2 [21:36:10 CEST] <DrSlony> llogan tool cool! Thank you! :) [21:38:26 CEST] <llogan> DrSlony: you may have to add format for compatibility with shitty players: pad,crop,format=yuv420p [21:39:04 CEST] <llogan> and -movflags +faststart if your viewers will be watching via progressive download [21:50:02 CEST] <boxgamex> has anyone ever used ffmpeg to capture really large videos (like 4k) via dshow? [21:50:10 CEST] <boxgamex> seems like I cant get the buffer large enough [21:51:51 CEST] <DrSlony> thank you very much llogan [22:10:42 CEST] <Smashcat> Hi, can anyone tell me if it's possible to export a .mov with transparency, as pngs with transparency? I've tried and it seems to add a white background. [22:31:49 CEST] <Anoia> Smashcat: not in any standard way I'm aware of [22:32:33 CEST] <Anoia> it'd be a feature of the player if it supported some sort of alpha [22:32:44 CEST] <Smashcat> Anoia: Yes I've been looking and trying different options and can't see a way to do it. Going to request the designer exports as an image sequence from after effects if possible. [22:37:06 CEST] <Anoia> what does it need to be transparent in/on? [22:41:53 CEST] <anon-15634> Kind of weird question. DTS-HD MA is made by combining a lossy DTS stream with a diff track created by comparing the master track with the compressed track. Thus, when combined it is lossless, but on older/incompatible players, the lossy core is still playable. Has any research been done on doing this with lossy & lossless video? Ex. a lossy 1080p core + diff to get it back to source video [22:42:20 CEST] <anon-15634> ffmpeg = audio / video enthusiasts; don't know where else to ask except maybe AVS forum [22:45:11 CEST] <llogan> Smashcat: you could try -c:v qtrle [22:45:26 CEST] <Smashcat> Anoia: It's part of an event installation, the image strips are over a coloured background. [22:45:38 CEST] <Smashcat> llogan: I'll try it thanks [22:47:22 CEST] <Smashcat> llogan: Not sure what that did, but I can't open the images it's created! [22:47:35 CEST] <Anoia> in what? web browser? an app? an encoded video? [22:47:59 CEST] <Smashcat> Anoia: In preview (Windows) - the file command reports it's just "data" [22:48:58 CEST] <Smashcat> Command I used was: ~/ffmpeg -i iris-open.mov -c:v qtrle -f image2 open_%05d.png [22:49:13 CEST] <llogan> oh, you're doing the opposite if what i thought you were [22:49:20 CEST] <Smashcat> :) [22:49:33 CEST] <Smashcat> Yeah the movie has transparency. I need to preserve it [22:49:52 CEST] <llogan> please show the complete console output of: ffmpeg -i iris-open.mov [22:49:55 CEST] <llogan> use a pastebin site [22:50:06 CEST] <llogan> and then provide the link [22:50:46 CEST] <Smashcat> http://pastebin.com/HSaASrzs [22:53:39 CEST] <llogan> you should first get a recent ffmpeg build [22:54:11 CEST] <llogan> then try with: ffmpeg -i input output_%05d.png [22:54:28 CEST] <llogan> http://johnvansickle.com/ffmpeg/ [22:54:46 CEST] <Smashcat> llogan: Ok I'll try that [22:58:56 CEST] <Smashcat> llogan: Nice! Thanks that worked great. [22:59:43 CEST] <Smashcat> I used the exact same command line as above, and it preserved the transparency. So I guess it was added after the older version I was using [23:01:00 CEST] <llogan> development is very active [23:01:32 CEST] <Smashcat> llogan: haha, yeah I'll try to keep up to date with it :) [23:02:44 CEST] <Smashcat> It's nice to be able to convert the video to images, then pipe that through "convert" to resize them all, then finally pipe that into "montage" to create the sprite strip automatically :) [23:09:48 CEST] <llogan> Smashcat: you can resize them with ffmpeg [23:09:55 CEST] <llogan> with scale filter [23:10:09 CEST] <llogan> and a montage with tile filter [23:10:29 CEST] <Smashcat> llogan: Couldn't get that to work - might have been the older version I was using. So I just created a shell script to do it all. Works fine though! [23:25:03 CEST] <Smashcat> #politics [23:25:33 CEST] <Smashcat> oops - pasting into wrong window - hahah [23:43:58 CEST] <yvear> llogan, same as first [23:50:10 CEST] <llogan> yvear: i can't really look into it now. try the ffmpeg-user mailing list. [23:53:59 CEST] <yvear> llogan, ok ty [00:00:00 CEST] --- Fri May 8 2015

1 0

ffmpeg-devel.log.20150506
by burek 07 May '15

07 May '15

[00:27:29 CEST] <cone-890> ffmpeg 03Anton Khirnov 07master:3735b5c61677: Revert "lavfi: always check av_expr_parse_and_eval() return value" [00:27:30 CEST] <cone-890> ffmpeg 03Michael Niedermayer 07master:af6739d6daf7: Merge commit '3735b5c616770429572f86aabdaec39c6ebb8818' [01:16:49 CEST] <cone-890> ffmpeg 03Andreas Cadhalpun 07master:9e66b39aa87e: diracdec: avoid overflow of bytes*8 in decode_lowdelay [02:27:06 CEST] <cone-890> ffmpeg 03Andreas Cadhalpun 07master:d93181ef3eac: diracdec: check if reference could not be allocated [03:03:48 CEST] <cone-890> ffmpeg 03Michael Niedermayer 07master:a2190de52d6c: avcodec/diracdec: Fix types and wraparounds in dirac_decode_picture_header() [03:18:04 CEST] <cone-890> ffmpeg 03Andreas Cadhalpun 07release/2.4:0d3a7dd26490: aasc: return correct buffer size from aasc_decode_frame [03:18:05 CEST] <cone-890> ffmpeg 03Andreas Cadhalpun 07release/2.4:97010c74cbff: alsdec: limit avctx->bits_per_raw_sample to 32 [03:18:06 CEST] <cone-890> ffmpeg 03Michael Niedermayer 07release/2.4:4ba11b944e46: Merge commit '0d3a7dd26490156b607541dd2e1faeaa0fc61a88' into release/2.4 [03:18:07 CEST] <cone-890> ffmpeg 03Andreas Cadhalpun 07release/2.4:1cb470934a26: alsdec: limit avctx->bits_per_raw_sample to 32 [03:18:08 CEST] <cone-890> ffmpeg 03Michael Niedermayer 07release/2.4:8d3a967635c3: Merge commit '97010c74cbff177b58daf9a092b4e37a7da26f85' into release/2.4 [03:33:51 CEST] <cone-890> ffmpeg 03Andreas Cadhalpun 07release/2.4:7b66cf5ce7fd: aacpsy: correct calculation of minath in psy_3gpp_init [03:33:52 CEST] <cone-890> ffmpeg 03Michael Niedermayer 07release/2.4:97d7c8d58488: Merge commit '7b66cf5ce7fdb8b3fa13459aab3f4d6ab559f1ea' into release/2.4 [03:48:21 CEST] <cone-890> ffmpeg 03Andreas Cadhalpun 07release/2.4:41a89cba6086: alsdec: check sample pointer range in revert_channel_correlation [03:48:22 CEST] <cone-890> ffmpeg 03Andreas Cadhalpun 07release/2.4:378ee3bad5b9: alsdec: only adapt order for positive max_order [03:48:23 CEST] <cone-890> ffmpeg 03Michael Niedermayer 07release/2.4:843b860ad974: Merge commit '41a89cba6086de2bd24f9ec7e21200fa162505e9' into release/2.4 [03:48:24 CEST] <cone-890> ffmpeg 03Michael Niedermayer 07release/2.4:146648771c95: Merge commit '378ee3bad5b99e8f90864af9bc851590e0f64825' into release/2.4 [04:12:53 CEST] <cone-890> ffmpeg 03Anton Khirnov 07release/2.4:8b86c2ed2eb2: Revert "lavfi: always check av_expr_parse_and_eval() return value" [04:12:54 CEST] <cone-890> ffmpeg 03Michael Niedermayer 07release/2.4:f28d3f98c5bb: Merge commit '8b86c2ed2eb29dce137b26e66a9bddd50f72817f' into release/2.4 [11:51:21 CEST] <cone-461> ffmpeg 03Dave Yeo 07master:22a0387df29c: OS/2:Makedef.cmd cleanup [12:30:06 CEST] <Daemon404> i really enjoy when people just ignore reviews that say their patch is malformed and to use send-email [12:30:11 CEST] <Daemon404> and just paste it in the client every time [12:30:27 CEST] <JEEBsv> lol [12:34:10 CEST] <wm4> they probably don't even get what the issue is? [12:34:36 CEST] <Daemon404> ... i explicitly told them to use send-email [12:34:39 CEST] <Daemon404> it's a direct instruction [13:41:20 CEST] <cone-461> ffmpeg 03Nicolas George 07master:7971fa9ce0ea: lavd/xcbgrab: fix comparison with screen size. [14:08:25 CEST] <kierank> Daemon404: shouldn't you be in berlin now [15:45:33 CEST] <kierank> j-b: lol kaltura violating gpl [15:49:00 CEST] <kierank> https://github.com/kaltura/platform-install-packages/issues/392 [15:49:29 CEST] <j-b> kierank: my my my [15:51:15 CEST] <kierank> just to make it clear they do --enable-nonfree --enable-libfdk-aac twice in their build [15:54:55 CEST] <j-b> which makes them not allowed to use ffmpeg anymore [15:54:57 CEST] <j-b> at all. [15:57:07 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:890b874889bc: avutil/internal: Suppress empty body warning from emms_c() [16:39:46 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:0a6b410edf79: avcodec/vp8: Suppress empty body warnings [16:39:47 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:3c7d3d27be98: tools/cws2fws: Use a similar implementation in dbgprintf() as in ff_dlog() [17:00:29 CEST] <cone-461> ffmpeg 03Thomas Guillem 07release/2.2:bac0850fbf87: matroskadec: fix crash when parsing invalid mkv [17:00:30 CEST] <cone-461> ffmpeg 03Andreas Cadhalpun 07release/2.2:941972d9e980: aasc: return correct buffer size from aasc_decode_frame [17:00:31 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07release/2.2:b34afd94ec90: Merge commit 'bac0850fbf870d4354d0f68840c0b1c9c1425c4a' into release/2.2 [17:00:32 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07release/2.2:55376ef978f8: Merge commit '941972d9e9806e30f978e8fd033671fd30a80a07' into release/2.2 [17:07:00 CEST] <cone-461> ffmpeg 03Andreas Cadhalpun 07release/2.2:d0e66cb1c7cf: alsdec: limit avctx->bits_per_raw_sample to 32 [17:07:01 CEST] <cone-461> ffmpeg 03Andreas Cadhalpun 07release/2.2:b0b92ee04d1e: aacpsy: correct calculation of minath in psy_3gpp_init [17:07:02 CEST] <cone-461> ffmpeg 03Andreas Cadhalpun 07release/2.2:645bc997722e: alsdec: limit avctx->bits_per_raw_sample to 32 [17:07:03 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07release/2.2:74409bff4ce9: Merge commit 'd0e66cb1c7cf2a9ef12963cdc6fd49a35599a9dd' into release/2.2 [17:07:04 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07release/2.2:36c5bb6a721a: Merge commit 'b0b92ee04d1ee49a5d8b1aca718ea7d7d44932aa' into release/2.2 [17:26:16 CEST] <cone-461> ffmpeg 03Andreas Cadhalpun 07release/2.2:d36d3ae02ca7: alsdec: check sample pointer range in revert_channel_correlation [17:26:17 CEST] <cone-461> ffmpeg 03Andreas Cadhalpun 07release/2.2:eaa9693fed36: alsdec: only adapt order for positive max_order [17:26:18 CEST] <cone-461> ffmpeg 03Andreas Cadhalpun 07release/2.2:f0bf5c538a85: alsdec: ensure channel reordering is reversible [17:26:19 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07release/2.2:f0cfa5d98a69: avcodec/alsdec: Use av_mallocz_array() for chan_data to ensure the arrays never contain random data [17:26:20 CEST] <cone-461> ffmpeg 03Andreas Cadhalpun 07release/2.2:deb0f487bdd7: alsdec: validate time diff index [17:26:21 CEST] <cone-461> ffmpeg 03Andreas Cadhalpun 07release/2.2:3a0a2c2586cd: alsdec: check sample pointer range in revert_channel_correlation [17:26:22 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07release/2.2:0756fa87bffd: Merge commit 'd36d3ae02ca7d535de2b26fa0b3ab44387d19b6a' into release/2.2 [17:26:23 CEST] <cone-461> ffmpeg 03Andreas Cadhalpun 07release/2.2:e93c46b3ccc1: alsdec: only adapt order for positive max_order [17:26:24 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07release/2.2:2a947c938261: Merge commit 'eaa9693fed36c29dfbac896946de31eb186ac5d0' into release/2.2 [18:00:49 CEST] <cone-461> ffmpeg 03Andreas Cadhalpun 07master:4b13a542a27d: dirac: use and forward error codes [18:03:25 CEST] <ubitux> mmh we can't adjust fps without decoding with ffmpeg? [18:04:41 CEST] <ubitux> like a basic setpts even without eval, just a factor [18:06:32 CEST] <ubitux> typically doing something like ffmpeg -r 120 -i 30fps.nut 120fps.nut [18:06:39 CEST] <ubitux> but with -c copy [18:07:08 CEST] <BBB> michaelni: as always, github has a mergeable tree if you want it for the net patchset [18:07:26 CEST] <BBB> https://github.com/rbultje/ffmpeg/commits/vp9-profile23-wip [18:08:00 CEST] <BBB> michaelni: and fate samples are at http://downloads.webmproject.org/test_data/libvpx/vp92-2-20-1{0,2}bit-yuv420.webm [18:08:01 CEST] <BBB> and [18:08:25 CEST] <BBB> http://downloads.webmproject.org/test_data/libvpx/vp93-2-20-1{0,2}bit-yuv4{… [18:08:34 CEST] <BBB> michaelni: let me know if you need anything else [18:12:12 CEST] <cone-461> ffmpeg 03Andreas Cadhalpun 07master:75fc81c83185: diracdec: check that block length is valid [18:12:24 CEST] <j-b> yay, MORE PIX_FMT [18:12:30 CEST] Action: j-b summons wm4 [18:12:37 CEST] <BBB> sorry [18:12:38 CEST] <wm4> huh [18:12:56 CEST] <BBB> not my choice [18:13:06 CEST] <wm4> oh dear [18:13:14 CEST] <wm4> yuv440p10be [18:13:18 CEST] <wm4> the most useful format ever! [18:13:19 CEST] <j-b> again, LE vs BE [18:13:25 CEST] <j-b> Why, why why? [18:13:28 CEST] <BBB> edontcare [18:13:32 CEST] <wm4> that's 1/4 subsampling isn't it [18:13:34 CEST] <BBB> no [18:13:37 CEST] <wm4> ah no [18:13:41 CEST] <BBB> its 422 transposed [18:13:44 CEST] <wm4> + .log2_chroma_w = 0, [18:13:44 CEST] <wm4> + .log2_chroma_h = 1, [18:13:48 CEST] <wm4> why does this exist? [18:13:54 CEST] <BBB> I cannot explain [18:14:04 CEST] <wm4> please whack google for me [18:14:14 CEST] <BBB> I agree that its a stupid format [18:14:31 CEST] <BBB> I think some engineer was like, whoa, wouldnt ie be cool if subsampling was a variable [18:14:31 CEST] <j-b> Can't we trivially convert to 422 ? [18:14:36 CEST] <wm4> didn't read the patches, but I suppose the format lets you specify arbitrary depths/subsampling [18:14:43 CEST] <wm4> so we need to add all combinations thereof? [18:14:46 CEST] <BBB> and then he created 444, 422 (yay!) and happened to create 440 also (oh well) [18:14:49 CEST] <BBB> and then he moved on [18:14:59 CEST] <BBB> right [18:15:09 CEST] <BBB> there is no useful use case for 440 that I can think of [18:15:13 CEST] <j-b> Do we _really_ need both BE and LE of this stupidity? [18:15:20 CEST] <BBB> but theres conformance suite samples so Im going to assume that its required to be supported [18:15:31 CEST] <BBB> BE/LE is cargo cult, feel free to fix it but outside scope of this patch [18:16:03 CEST] <wm4> j-b: many encoders/decoders for stupid legacy image formats need them [18:16:17 CEST] <wm4> so cleaning up this shit would take a while [18:17:28 CEST] <j-b> wm4: of course, but why add it for new ones? [18:17:30 CEST] <j-b> like 440 [18:17:31 CEST] <BBB> Im also wondering if I should consolidate some of the dsp functions [18:18:07 CEST] <BBB> like copy for 8bpp and 16bpp is identical if size is double [18:18:13 CEST] <wm4> j-b: because then you'd get swapped endian on a BE machine [18:18:17 CEST] <BBB> and intra pred for 10 and 12bpp are identical except for tm [18:18:27 CEST] <wm4> and we all care about BE machines, right [18:18:29 CEST] <j-b> wm4: which BE machine? [18:19:09 CEST] <j-b> wm4: and, so? BE machines decoding 440 10bits, would get a swap to do... [18:19:32 CEST] <BBB> the format is always native [18:19:41 CEST] <wm4> then the format would be native endian, which would probably conflict with raw encoders or some shit [18:19:43 CEST] <BBB> the only case where Be/LE comes in is if we store it in a raw .yuv file [18:19:51 CEST] <BBB> and read that on a machine of different endianness [18:20:09 CEST] <BBB> in any other case the machine only ever sees the native form [18:20:17 CEST] <wm4> currently, a pixfmt is supposed to define a byte representation, endian-independent [18:20:37 CEST] <wm4> (not that I agree with this, but that's how it is) [18:20:59 CEST] <wm4> Libav is open to cleaning it up [18:22:56 CEST] <BBB> I encourage you guys to clean it up [18:23:00 CEST] <BBB> but its orthogonal to the patchset [18:23:25 CEST] <BBB> its like bringing up ambiguous cosmetic shit in new decoder submissions [18:24:04 CEST] <j-b> I kind of disagree [18:24:05 CEST] <BBB> please align horizontally and shit like that, Im completely allergic to that sort of stuff [18:24:38 CEST] <j-b> Your patch is adding a new pixfmt to libavutil, it's not orthogonal to the patchset [18:24:50 CEST] <j-b> and it's very very different from stupid cosmetic discussions [18:25:11 CEST] <j-b> Once it's in the headers, you're screwed [18:25:28 CEST] <BBB> yuv420p9be is already in the headers [18:25:31 CEST] <BBB> so the problem exists [18:25:37 CEST] <BBB> if you want to fix it, fix it for all of them [18:25:45 CEST] <BBB> saying that 440p10le is somehow special, is silly [18:26:46 CEST] <wm4> yeah, in theory the addition of your pixfmt is fine [18:26:54 CEST] <wm4> it follows the existing rules [18:27:21 CEST] <BBB> I will whack google about the more general concept of adding 440 [18:27:31 CEST] <BBB> I agree that 440 is stupid and unnecessary [18:28:14 CEST] <wm4> they are probably fascinated by the idea that they can implement different subsampling so generically and easily [18:28:25 CEST] <wm4> but that doesn't mean it's a good idea to allow such obscure/useless formats [18:30:49 CEST] <iive> BBB: is there point in using something else than native endiannes for decoder? [18:31:43 CEST] <iive> aka, are bitstreams different for 420p10be and 420p10le ? [18:31:54 CEST] <BBB> decoders only output native [18:32:05 CEST] <BBB> but rawvideo from a yuv file could be anything [18:32:53 CEST] <iive> i'm not against adding new pixfmts, rgb vs bgr is also endian issue. [18:33:19 CEST] <wm4> iive: we've discussed why both are needed above [18:35:51 CEST] <BBB> wm4: https://code.google.com/p/webm/issues/detail?id=996 [18:37:34 CEST] <wm4> huh, they're still on google code? [18:38:37 CEST] <wm4> also, it's not strictly about the needed amounts of code, but the explosion of (obscure) pixel formats, which directly burdens applications [18:39:14 CEST] <wm4> even vlc is complaining! [18:39:50 CEST] <BBB> I dont think it matters what its about [18:39:51 CEST] <BBB> it should go [18:40:10 CEST] <BBB> feel free to voice your outrage in that bug report :) [18:40:38 CEST] <wm4> you pasted the rage into your bug report [18:40:46 CEST] <BBB> more rage is good [18:42:28 CEST] <michaelni> BBB, the patchset needs some version check i think, it fails building libavcodec/libvpxdec.c here with libavcodec/libvpxdec.c:75:10: error: VPX_IMG_FMT_I440 undeclared (first use in this function) [18:44:04 CEST] <BBB> hm... [18:44:08 CEST] <BBB> thats odd [18:44:16 CEST] <BBB> let me check [18:45:39 CEST] <BBB> youre right, it was added in libvpx commit a30774c50d2b1b626b05a99dcd29aa766f0b223b [18:45:54 CEST] <BBB> what do you want me to do? update configure check? [18:48:52 CEST] <Daemon404> kierank, no [18:48:54 CEST] <Daemon404> im flying tomorrow [18:48:57 CEST] <Daemon404> im not going to the first day [18:49:07 CEST] <kierank> lol [18:49:07 CEST] <Daemon404> why would i want to sit in the middle of a circlejerk for longer than needed [18:49:09 CEST] <michaelni> BBB, no idea, whatever you prefer, or ask James Zern (he is listed in MAINTAINERs) [18:49:19 CEST] <wm4> is this about the smpte conference? [18:49:24 CEST] <kierank> Daemon404: because v-nova [18:49:26 CEST] <kierank> dugh [18:49:33 CEST] <kierank> changes everything [18:49:33 CEST] <michaelni> BBB, also you need a update for fate-filter-pixfmts-crop and a few others [18:49:45 CEST] <BBB> oh right [18:49:48 CEST] <BBB> I forgot [18:50:50 CEST] <Daemon404> kierank, they even offered me a job [18:50:52 CEST] <Daemon404> lul [18:52:23 CEST] <j-b> Daemon404: haha. you accepted, yes? [18:52:33 CEST] <kierank> amazing how much noise you can make over vapourware [18:52:44 CEST] <Daemon404> j-b, lol [18:53:12 CEST] <Daemon404> i should probab;ly practice my slides at some point [18:53:14 CEST] <Daemon404> probably. [18:53:23 CEST] <j-b> Daemon404: you should have met them [18:53:29 CEST] <j-b> and report the bullshitware [18:53:36 CEST] <Daemon404> vittorio met them iirc [18:54:04 CEST] <j-b> ah? [18:54:20 CEST] <Daemon404> id rather not ;) [18:54:27 CEST] <kierank> i saw a lot of them at tv connect [18:54:43 CEST] <j-b> they are on the awarding list BS [18:55:06 CEST] <kierank> you just pay money and you get an award [18:55:12 CEST] <kierank> it's not hard [18:55:16 CEST] <kierank> they call me all the time [18:55:34 CEST] <Daemon404> sounds like a posh special ed school [20:22:28 CEST] <cone-461> ffmpeg 03Nicolas George 07master:01fdfa51aca9: xcbgrab: Accept geometries matching the screen size [20:22:29 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:958ddef46c22: Merge commit '01fdfa51aca9086e04bd354fe3f103a49352c085' [20:31:28 CEST] <cone-461> ffmpeg 03Vittorio Giovara 07master:d00f1e0fc10b: quickdraw: Convert to bytestream2 [20:31:29 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:d423fc12e6e1: Merge commit 'd00f1e0fc10bb945cdf140a6c320eb0c9cd25162' [20:50:39 CEST] <cone-461> ffmpeg 03Vittorio Giovara 07master:34efb8a169e4: quickdraw: Support direct pixel blocks [20:50:40 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:39e89f521159: Merge commit '34efb8a169e4551326e069be47125c6c2cb7ab90' [20:51:18 CEST] <jamrial> BBB: yeah, works now [20:58:56 CEST] <kierank> j-b: asking the lawyers: https://github.com/kaltura/platform-install-packages/issues/392#issuecommen… [21:00:20 CEST] <BBB> jamrial: do you know anything about configure version checks to get libvpxs 440 symbol to be there? [21:03:29 CEST] <jamrial> not really. VPX_IMG_FMT_HIGHBITDEPTH is currently checked for profile 2, so it probably guarantees profile 1 440 but not profile 3 [21:05:34 CEST] <jamrial> and the check should be done in libvpx*.c rather than configure, where we just check for a new enough version to enable the vp8/vp9 decoders and encoders [21:08:33 CEST] <BBB> the same commit introduced highbitdepth 440 as well as normal 440 [21:08:43 CEST] <BBB> so I think the define was added after high bitdepth in the code [21:08:52 CEST] <BBB> also, running all of fate is REALLY slow [21:08:58 CEST] <BBB> as in, we have a shitton of tests [21:09:02 CEST] <BBB> are all of these tests useful? [21:11:02 CEST] <BBB> michaelni: pushed new github tree which adds/updates relevant references for 440 [21:16:18 CEST] <cone-461> ffmpeg 03Vittorio Giovara 07master:7b734ee55dbb: lavf: Open PICT images with Quickdraw [21:16:19 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:ffed987a4d4a: Merge commit '7b734ee55dbb8476d7ad63c7daf55c534cf82d5d' [21:29:34 CEST] <jamrial> BBB: checking for VPX_IMAGE_ABI_VERSION >= 3 should work [21:30:44 CEST] <BBB> ok, pushed a new version (to github) that does that [21:30:54 CEST] <jamrial> that define value was bumped about two months after 440 support was added. not ideal, but i guess it's better than nothing [21:31:04 CEST] <BBB> its good enough I guess [21:46:07 CEST] <cone-461> ffmpeg 03Vittorio Giovara 07master:d81fb63d8769: fate: Add a PICT test [21:46:08 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:20ce69e5f85d: Merge commit 'd81fb63d87692765c004c19934b49427df434a07' [22:28:57 CEST] <cone-461> ffmpeg 03Vittorio Giovara 07master:04070dbca068: libx265: Fix 'braces around scalar initializer' warning [22:28:58 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:1caff5731d9a: Merge commit '04070dbca0688ab1e24528ce5c135254a9a79c47' [22:35:48 CEST] <cone-461> ffmpeg 03Vittorio Giovara 07master:6c8c7c246280: dump: Print AVAudioServiceType side data [22:35:49 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:0318013afcff: Merge commit '6c8c7c246280cdcd23e4d99006b95245ae837278' [22:44:26 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:e0010bb91fc0: mux: Do not leave stale side data pointers in ff_interleave_add_packet() [22:44:27 CEST] <cone-461> ffmpeg 03Michael Niedermayer 07master:2284c38e4c68: Merge commit 'e0010bb91fc0b067be8aefd2f77e6838537da433' [00:00:00 CEST] --- Thu May 7 2015

1 0

ffmpeg.log.20150506
by burek 07 May '15

07 May '15

[00:03:48 CEST] <NoTiTo_> c_14: do you have link to a fool proof guide to building / compiling your own ffmpeg [00:03:50 CEST] <NoTiTo_> ? [00:04:57 CEST] <NoTiTo_> c_14: thanks mate, the stream is working now, albeit not perfect. I didn't realise it was the 'broken' ffmpeg giving me issues [00:08:46 CEST] <c_14> NoTiTo_: https://trac.ffmpeg.org/wiki/CompilationGuide [00:28:42 CEST] <NoTiTo_> c_14: is there a way for ffmpeg to output a CBR MPEG TS? [00:29:34 CEST] <c_14> ffmpeg -i video -b:v xk -minrate xk -maxrate xk -f mpegts out.ts [00:30:13 CEST] <Mavrik> NoTiTo_, there's a muxrate parameter that stuffs MPEG-TS stream with NULL packets [00:30:17 CEST] <Mavrik> but it's not always reliable [00:30:43 CEST] <Mavrik> c_14, that won't necessarily make a properly compliant CBR TS stream :) [00:31:04 CEST] <c_14> meh, compliancy [00:31:11 CEST] <c_14> :P [00:34:17 CEST] <NoTiTo_> Mavrik, thanks. I was just looking into that [00:51:26 CEST] <NoTiTo_> thanks and catch you later peeps! [02:47:21 CEST] <Norwack> Don't know if this is the right place to ask, but anyways; Is there any documentation available on librtmp? The man page isn't very detailed [04:18:10 CEST] <jjohn> Ahoy, guys. I do have a problem with the amerge mixer in ffmpeg and thought that maybe you guys could help me out there. [04:18:17 CEST] <jjohn> The problem: http://pastebin.com/R4KQ4yPd [04:35:52 CEST] <klaxa> jjohn: i had a similar problem and resorted to using sox separately to ffmpeg [04:37:38 CEST] <jjohn> klaxa no option here, though. The audio comes from two devices. No files. [04:37:44 CEST] <klaxa> what you want is merge on the sox manpage (http://sox.sourceforge.net/sox.html) [04:37:58 CEST] <klaxa> you could do it with pipes [04:38:05 CEST] <klaxa> but i don't know how much delay would be added [04:38:40 CEST] <klaxa> you could probably duplicate the mono input and generate a stereo input [04:38:52 CEST] <jjohn> Also, the audio needs to be muxed into FLV as well. [04:39:54 CEST] <jjohn> Am I not already doing this? Accoding to this site: https://trac.ffmpeg.org/wiki/AudioChannelManipulation [04:40:13 CEST] <jjohn> Mono -> stereo: simply add -ac 2 to the output options. [04:41:21 CEST] <klaxa> that is for the output stream though [04:41:35 CEST] <klaxa> you could do this: https://trac.ffmpeg.org/wiki/AudioChannelManipulation#monostereo [04:41:51 CEST] <klaxa> and then use amix [04:42:02 CEST] <klaxa> to see if it makes it any better [04:43:16 CEST] <jjohn> You mean, by first changing the volume of each stream separately, and then using amix? [04:45:27 CEST] <klaxa> yes [04:45:53 CEST] <klaxa> your filtergraph could look something like this: "[0:0]volume=0.20:precision=fixed[a0];[1:0]volume=6.0:precision=fixed[a1];[a1][a1]amerge=inputs=2[a2];[a0][a2]amix[aout]" [04:53:47 CEST] <jjohn> Hm ... does not work out. "Stream specifier 'a1' in filtergraph description [0:0]volume=0.20:precision=fixed[a0];[1:0]volume=6.0:precision=fixed[a1];[a1][a1]amerge=inputs=2[a2];[a0][a2]amix[aout] matches no streams." [05:03:48 CEST] <klaxa> that's weird [05:04:28 CEST] <jjohn> Before you ask: I have the latest ffmpeg version installed. Freshly downloaded from git and even more freshly compiled and installed. [05:06:43 CEST] <klaxa> i have to agree, it works not as i expected [05:13:08 CEST] <fffan> AV_SAMPLE_FMT_S16 means singed 16 bit integer, and range is -32767 to +32767 and 0 means silence, so, my question is what does negative value mean? [05:13:37 CEST] <fffan> 0 means silence, what does -1 means? [05:15:32 CEST] <fffan> more silent? [05:20:49 CEST] <jjohn> fffan: my guess would be that the negative values are the 1's complement encoded values for the 32767 - 65534 range. [05:22:15 CEST] <jjohn> -1 would be 65534 in the 1's complement system. [05:27:57 CEST] <fffan> if so, why AV_SAMPLE_FMT_S16, why not AV_SAMPLE_FMT_u16 [05:28:45 CEST] <fffan> AV_SAMPLE_FMT_u16 is more intuitionistic [05:29:12 CEST] <jjohn> Because hardware can be picky about what to understand, I'd say. [05:30:21 CEST] <jjohn> u16 does not use any complement, so you get rid of the two-representations-for-0 problem in the 1's complement system and have a 65535 value available in u16. [05:30:41 CEST] <fffan> you know there is a definition of another pcm type [05:30:42 CEST] <fffan> AV_SAMPLE_FMT_U8, ///< unsigned 8 bits [05:32:59 CEST] <jjohn> u8 would be the same. Instead of your -127 - 127 range, you can have a 255 value (which is -128 in the 2's complement system which is not available in the 1's complement system). [05:37:56 CEST] <fffan> one's complement system is rare [05:39:19 CEST] <jjohn> Can be, but it's simple enough for hardware decoders. [05:41:48 CEST] <fffan> thanks, I'll google details about pcm16. [05:42:52 CEST] <jjohn> Also: http://wiki.multimedia.cx/?title=PCM [05:43:30 CEST] <jjohn> Looks like I was wrong, it always uses 2's complement. [05:43:40 CEST] <jjohn> > If the sample is unsigned, the sample range is 0..255 with a centerpoint of 128. If the sample is signed, the sample range is -128..127 with a centerpoint of 0. If a PCM type is signed, the sign encoding is almost always 2's complement. [05:45:16 CEST] <fffan> AV_SAMPLE_FMT_S16 means always signed, so always 2's compement [05:51:59 CEST] <jjohn> Anyhow: does anyone have an idea for this one? http://pastebin.com/R4KQ4yPd [06:15:11 CEST] <k_sze[work]> Does zeranoe's builds for Windows make use of advanced Intel CPU features? [07:37:16 CEST] <fffan> zeranoe's build sucks [07:39:46 CEST] <k_sze[work]> fffan: as in they are hardly optimized? [07:41:13 CEST] <fffan> I don't think so. [08:32:57 CEST] <k_sze[work]> I wish I know how to build ffmpeg on windows. [08:33:24 CEST] <k_sze[work]> I still haven't managed to get the mingw-w64 toolchain and everything working. [08:33:44 CEST] <k_sze[work]> I kind need to make an optimized build for 4th gen Core i5 [08:34:31 CEST] <k_sze[work]> We're encoding 4 streams in real time simultaneously and the Core i5 can hardly catch up. [09:04:41 CEST] <fffan> so k_sze, which country are you from? [09:05:24 CEST] <jjohn> k_sze may I ask what you have been trying to do? Because I can cross-compile ffmpeg on my Linux box. [09:06:45 CEST] <jjohn> OK, it cannot link the final file, because the linker of mingw32 seems to be broken and exists with a segfault, but I guess that's because my toolchain is outdated. [09:06:58 CEST] <jjohn> But the libraries do compile. [09:07:38 CEST] <jjohn> exits* [10:12:53 CEST] <Mavrik> k_sze[work], yes, the most important encoders do. [10:13:15 CEST] <Mavrik> compiling it yourself won't really add any performance [10:15:04 CEST] <k_sze[work]> fffan: Hong Kong [10:26:37 CEST] <k_sze[work]> Mavrik: I'm especially interested in H.264 and FFV1 encoding optimization. [10:26:53 CEST] <Mavrik> H.264 uses libx264 library which always autodetects CPU features [10:26:56 CEST] <k_sze[work]> but I guess those are already highly optimized in the default build? [10:27:51 CEST] <k_sze[work]> I'm doing two H.264 streams, yuyv 1920x1080 and yuyv 1600x1200, 30fps [10:27:52 CEST] <Mavrik> ffv1 encoder doesn't have any specific CPU optimizations though [10:28:11 CEST] <k_sze[work]> the Core i5 can barely keep up using ultrafast preset. [10:28:55 CEST] <k_sze[work]> I wonder if QSV will be any better. [10:31:24 CEST] <jjohn> Does anyone have an idea for this problem? http://pastebin.com/R4KQ4yPd [10:34:28 CEST] <Mavrik> jjohn, I think you're supposed to use amix, not amerge [10:34:48 CEST] <Mavrik> jjohn, since amerge will create a 3 channel output [10:34:58 CEST] <Mavrik> not a stereo output like you want [10:35:12 CEST] <Mavrik> also try using HE-AAC profile which will improve quality for spoken voice [10:36:41 CEST] <jjohn> It's not only spoken voice, though - only the second one. [10:39:27 CEST] <jjohn> Also, I have now changed the filter to "[1:0]volume=0.20:precision=fixed[a0];[2:0]volume=6.0:precision=fixed[a1];[a0][a1]amix[aout]". But it still sounds metallic. [10:40:12 CEST] <jjohn> Mavrik [10:40:50 CEST] <Mavrik> did you actually check ffmpeg output to see what you're getting encoded? why are you forcing fixed precision? [10:42:14 CEST] <jjohn> Because I only have s16 coming from the input ... shouldn't I do this? [10:43:56 CEST] <jjohn> And yes, I checked the output. Input: two streams, first one: 0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s, second one: 1:0: Audio: pcm_s16le, 48000 Hz, mono, s16, 768 kb/s [10:44:24 CEST] <jjohn> Output: one stream: amix (graph 0) -> Stream #0:0 (libfdk_aac) [10:46:04 CEST] <jjohn> BTW, it also sounds bad if I increase the bitrate to 320 kbits/s - 320 kbits/s should be enough in every case, whether it's voice or not. [10:46:16 CEST] <jjohn> Mavrik [11:23:19 CEST] <jjohn> No idea? [13:14:00 CEST] Last message repeated 1 time(s). [14:05:58 CEST] <haxie> are palettegen and paletteuse only available in recent ffmpeg versions? do you need to set special flags for those? [15:37:42 CEST] <captain-codec> Hello! I don't have not so much experience with codecs transcode ... my question is: I have an Intel(R) Core(TM) i7-4600M CPU @ 2.90GHz. Is it possible to transcode a mkv movie and stream it over mpeg-ts with a bitrate about 4mbit. the cpus are at 20% but I get some dups. I think the dups are responsible for the artifacts / false pixel. am i rig [15:37:42 CEST] <captain-codec> ht that my hardware must be able to transcode and stream it? [15:45:04 CEST] <Mavrik> you didn't really say which codecs and encoders you use [15:45:17 CEST] <Mavrik> but that CPU should handle H.264/AAC encode for a single stream yeah [15:54:10 CEST] <captain-codec> input codec is h256 and the output stream uses mpeg2 [15:54:51 CEST] <captain-codec> when i tried h264/aac I had 100% cpu utilization [16:00:51 CEST] <Mavrik> And that's a bad thing? [16:01:18 CEST] <captain-codec> well yes cause my stream is laggy then [16:01:27 CEST] <captain-codec> and I have some dups [16:02:01 CEST] <captain-codec> that's my command line ffmpeg -re -i file.mkv -vcodec libx264 -s 1280x720 -r 25 -b:v 8M -acodec libfaac -ac 2 -b:a 192k -f mpegts udp://239.0.0.1?pkt_size=1316 [16:02:27 CEST] <RobotsOnDrugs> x264 has some tuning options for streaming [16:03:30 CEST] <Mavrik> What are these dups you're talking about? [16:04:24 CEST] <captain-codec> dup [16:04:34 CEST] <captain-codec> I think dup are duplicated pictures [16:04:55 CEST] <captain-codec> for instance : "frame= 324 fps= 25 q=2.0 Lsize= 7104kB time=00:00:13.45 bitrate=4325.8kbits/s dup=13 drop=0 " [16:13:21 CEST] <captain-codec> @RobotsOnDrugs are this tuning options minrate maxrate and bufsize ? [16:15:19 CEST] <RobotsOnDrugs> those are a few that might be helpful [16:15:28 CEST] <RobotsOnDrugs> there's also -tune zerolatency [16:15:42 CEST] <RobotsOnDrugs> (i forget how to map that to ffmpeg though) [16:16:42 CEST] <c_14> -tune zerolatency [16:16:54 CEST] <captain-codec> well with this cmd then the stream is more unstable than without :/ [16:20:04 CEST] <captain-codec> ok i wil lgive it a try with zerolatency [16:20:58 CEST] <captain-codec> how would you stream a h265 over h264 ? [16:21:38 CEST] <c_14> ^what [16:26:21 CEST] <captain-codec> I meant, how would you stream a file which has a codec h265 over multicast with codec h264/aac ? [16:31:16 CEST] <c_14> The same way I'd stream any other file? [16:31:45 CEST] <c_14> ffmpeg -i file -c:v libx264 -c:a aac -strict -2 -f matroska udp://wheeee [16:32:29 CEST] <captain-codec> so and there might be no problems ? [16:32:34 CEST] <captain-codec> with artifacts [16:33:23 CEST] <captain-codec> or shouldn't i transcode while streaming? [16:33:24 CEST] <c_14> Unless the udp borks, no. [16:33:35 CEST] <c_14> As long as you can decode/encode at realtime it shouldn't be a problem. [16:34:39 CEST] <Mavrik> -tune zerolatency will fuckup quality [16:34:55 CEST] <Mavrik> for no reason since you don't need zerolatency when reading a FILE [16:35:11 CEST] <Mavrik> you're getting artifacts probably due to issues on UDP layer, try using TCP [16:35:14 CEST] <Mavrik> and fixing network [16:37:00 CEST] <captain-codec> well i am streaming over my private network laboratory. I cant see any drops or crc on the interfaces [16:37:40 CEST] <captain-codec> well maybe the iptv boxes can be crap.... but when i try it with my iptv provider i dont have any issues [16:38:37 CEST] <Mavrik> you're not going to get artifacting on encoder layer without explicit error messages [16:38:44 CEST] <Mavrik> "dup" isn't really important in your context [16:39:03 CEST] <Mavrik> so if you're getting blocking artifacts (missing parts of new picture) you're losing data on a network layer or ordering is getting messed up [16:56:07 CEST] <jasonwhite> Hi guys, I am having a problem and I hope some guru here could help me :D I want to compare bitrate,file size ... of two Xvid and H.264 videos of the same quality. With x264, I only know -crf option (default 23) to set quality ( I dont know -qp, quantization param, since the official guide did not mention). With Xvid I know -qscale option. My question is, since -crf range is 0-51, -qscale range is 0-31, what is the proper value o [16:56:19 CEST] <jasonwhite> and -qscale to have those 2 videos the same quality? [16:56:40 CEST] <jasonwhite> Oopss. Sorry. Split line... I will post my question again. [16:56:43 CEST] <jasonwhite> ####################### [16:56:47 CEST] <jasonwhite> Hi guys, I am having a problem and I hope some guru here could help me :D I want to compare bitrate,file size ... of two Xvid and H.264 videos of the same quality. With x264, I only know -crf option (default 23) to set quality ( I dont know -qp, quantization param, since the official guide did not mention). With Xvid I know -qscale option. [16:56:53 CEST] <jasonwhite> My question is, since -crf range is 0-51, -qscale range is 0-31, what is the proper value of -crf (or -qp if possible ) and -qscale to have those 2 videos the same quality? [17:04:15 CEST] <ps-auxw> jasonwhite: There isn't one, really. You'll need to try a bunch of values and visually compare the results to see if they are sufficiently close. It's subjective of course. IMHO it would be better to encode the to have the same bitrate using 2 pass encoding and then compare quality. Depending on what you are going for, try to match encoding speed too. That's much easier to do in a fair way. [17:07:06 CEST] <jasonwhite> ps-auxw: First, you are my favorite command :D Second, I have done the test with constant bit rate and now try to do with varying bit-rate. As you said, I must have try bunch of value, and then I can do PSNR test instead of subjectively judge 2 videos, right ? [17:07:33 CEST] <captain-codec> thank you guys. I will try it again! bye [17:12:23 CEST] <ps-auxw> jasonwhite: Hehe. :) If you are going to go with an objective measurement, I would recommend SSIM rather than PSNR. It's not perfect, but it's closer to human perception than PSNR. Also, you'll want to use the ssim tune for x264, as the psy optimizations (which make things look better, but lower PSNR and SSIM) will reduce SSIM otherwise. [17:14:38 CEST] <ps-auxw> You can also take a look at this for things to avoid when doing a comparison: http://x264dev.multimedia.cx/archives/472 [17:16:23 CEST] <jasonwhite> ps-auxw: I dont know about psy optimizations part, seems like needing a lot of reading ... The link sounds fun. Thanks for your information ! [17:21:04 CEST] <ps-auxw> You're welcome. :) [17:27:39 CEST] <jjohn> Does anyone have an idea for this problem? http://pastebin.com/R4KQ4yPd [20:10:25 CEST] <jjohn> Noone? [20:11:41 CEST] <llogan> jjohn: you didn't provide the complete console outputs [20:17:53 CEST] <jjohn> llogan there you go: http://pastebin.com/ABgSiX1F [20:22:51 CEST] <seasc> jjohn, whats the question? just loged in after you asked 'noone'? [20:23:09 CEST] <seasc> erm, just before :p [20:23:49 CEST] <jjohn> seasc this one: http://pastebin.com/R4KQ4yPd [20:25:10 CEST] <jjohn> Thee reason why in the previous pastebin the command has a bitrate of 320kbits/s was to show that it's not an issue due to the lossy encoding of AAC. [20:25:26 CEST] <seasc> jjohn, by merge, you mean join as in 'append'? [20:25:59 CEST] <jjohn> No, as in "Both audio sources play at the very same time". [20:27:08 CEST] <jjohn> Both amerge and amix are capable of doing that. But the quality loss is just huge - I don't expect transparency here, but we are talking about AAC 128 kbits/s here. [20:29:23 CEST] <jjohn> Such quality loss is not acceptable - and please note that the loss is even there at 320 kbits/s, so the bitrate can't be the culprit. [20:32:29 CEST] <seasc> why are you using microphone and loop_out as inputs? [20:33:06 CEST] <seasc> and i'm confused to see 'mono' and 'stereo' as input-files [20:34:06 CEST] <seasc> jjohn, saying, the command looks too long to my eyes, thinking that ffmpeg automaticly reckognizes the stream/file as mono/stereo, but either way, this is too complex for me, sorry [20:34:11 CEST] <jjohn> mono = microphone, stereo = loop_out. I meant to change it for clearity (so that you can see what's what), but because I am stupid I did the job only to 50%. :( [20:34:35 CEST] <seasc> :) no worries, thought so, its just confusing a bit [20:35:20 CEST] <jjohn> What do you mean with "too long for your eyes"? Something's bothering you? [20:51:19 CEST] <jjohn> c_14 got that? [20:56:28 CEST] <c_14> Can you create input files that this happens with? Cause I just tested with a stereo + mono file I had lying around and I can't notice any quality loss exceeding that caused by lossy reencoding. (though I used opus instead of aac, but if the problem is really the mixing that shouldn't matter) [20:57:09 CEST] <jjohn> Gimme a second, I'll try. [20:57:59 CEST] <jjohn> Input files in FLAC format? [20:58:54 CEST] <c_14> Sure, doesn't really matter as long as it's reproducible. [21:05:34 CEST] <jjohn> Why, that's strange. I did as you said - create one stereo and one mono file (stereo.flac and mono.flac), and if I don't grab it directly from the ALSA interfaces, the audio in the output file is fine, even with 128 kbits/s. [21:06:30 CEST] <jjohn> That is, if I just exchange the ALSA grabbing parameters with stereo.flac and mono.flac, I don't have any problems. [21:08:07 CEST] <c_14> Can you trie getting rid of the -channel_layout input options? [21:08:19 CEST] <c_14> *try [21:08:48 CEST] <jjohn> Already did so since seasc said to drop them. [21:09:10 CEST] <c_14> oh, right [21:09:33 CEST] <c_14> how about the thread_queue_size ? [21:11:31 CEST] <jjohn> Got rid of the parameters for both streams. Now I have the warning "Thread message queue blocking; consider raising the thread_queue_size option (current value: 8)" showing up twice, and the output sounds bad again. [21:12:58 CEST] <c_14> no clue. Try grabbing the streams from alsa in one ffmpeg process and then piping them as pcm (inside wav or something) to another ffmpeg process that does the filtering? [21:22:15 CEST] <jjohn> c_14 hm. I did that. The quality loss is gone ... but it seems that there is either an echo now, or the stereo channel are not in sync anymore, because I hear the sound from the stereo source twice. [21:23:16 CEST] <jjohn> I created the pipes with mknod stereo p and mknod mono p and started two more ffmpeg processes that would grab from each ALSA interface. [21:24:13 CEST] Action: c_14 was thinking of `ffmpeg -f alsa -i 1 -f alsa -i 2 -c:a pcm_s16le -f wav pipe:1 | ffmpeg -i pipe:0 -filter_complex ' [21:24:58 CEST] <jjohn> Lol didn't know it was possible like that. I will try that out. [21:30:42 CEST] <jjohn> Can I grab both audio streams with only one ffmpeg process? Because now I have: /opt/local/ffmpeg/bin/ffmpeg -threads 7 -thread_queue_size 4096 -f alsa -ac 2 -i loop_out -f alsa -ac 1 -i microphone -f wav -c:a pcm_s16le pipe:1 | /opt/local/ffmpeg/bin/ffmpeg -threads 7 -i pipe:0 -filter_complex "[1:0][1:0]amerge=inputs=2[a1];[0:0]volume=0.20:precision=fixed[a0];[a1]volume=6.0:precision=fixed[a2];[a0][a2]amix [21:30:48 CEST] <jjohn> [aout]" -map [aout] -c:a libfdk_aac -cutoff 15000 -b:a 128k -f flv /home/jjohn/test_recording.flv, but it only seems to grab one stream. [21:31:46 CEST] <c_14> eh, add -map 0 -map 1 to the first ffmpeg process [21:33:32 CEST] <jjohn> Blech. "[wav @ 0x1b7d1e0] WAVE files have exactly one stream". That's what it's giving me right now. [21:33:45 CEST] <c_14> use matroska [21:33:49 CEST] <c_14> -f matroska [21:34:35 CEST] <seasc> c_14, for audio only ? [21:34:48 CEST] <c_14> why not [21:34:57 CEST] <c_14> Know of another format that eats 2 streams of raw pcm? [21:35:09 CEST] <seasc> afiu: he doesnt want to 'switch' streams,but mix them? [21:35:12 CEST] <seasc> nah [21:35:19 CEST] <seasc> oh wait, webm? :p but i get you [21:35:54 CEST] <c_14> Don't think you can put pcm in webm [21:36:00 CEST] <c_14> Think it only accepts vorbis and opus [21:36:21 CEST] <jjohn> Doesn't matter. Because ... it does work, but the audio sounds bad again. :) [21:36:39 CEST] <c_14> meh [21:36:51 CEST] <c_14> Is the act of having ffmpeg grab more than one alsa stream making it sound bad? [21:36:55 CEST] <jjohn> Just a "meh"? I'd rather say "bloody hell". [21:37:11 CEST] <jjohn> I don't know TBH. [21:37:22 CEST] <seasc> did you upchannel the mono thing yet as source? [21:37:40 CEST] <c_14> try `ffmpeg -f alsa -i foo -c:a pcm_s16le -f matroska pipe:1 | ffmpeg -i pipe:0 -f alsa -i bar -f matroska -map 0 -map 1 pipe:1 | ffmpeg -i pipe:1 -filter_complex ' [21:37:41 CEST] <jjohn> seasc: opt/local/ffmpeg/bin/ffmpeg -threads 7 -thread_queue_size 4096 -f alsa -ac 2 -i loop_out -f alsa -ac 1 -i microphone -map 0 -map 1 -f matroska -c:a pcm_s16le pipe:1 | /opt/local/ffmpeg/bin/ffmpeg -threads 7 -i pipe:0 -filter_complex "[0:1][0:1]amerge=inputs=2[a1];[0:0]volume=0.20:precision=fixed[a0];[a1]volume=6.0:precision=fixed[a2];[a0][a2]amix[aout]" -map [aout] -c:a libfdk_aac -cutoff 15000 -b:a 1 [21:37:47 CEST] <jjohn> 28k -f flv -y /home/jjohn/test_recording.flv [21:37:52 CEST] <seasc> or make them as other streams, like dunno if that is any better, wav or mp3? [21:38:41 CEST] <seasc> its above my understanding, just trying to put some ideas [21:39:13 CEST] <jjohn> c_14 I'll try it out. [21:44:24 CEST] <jjohn> Holy shit. Quoting myself here: The quality loss is gone ... but it seems that there is either an echo now, or the stereo channel are not in sync anymore, because I hear the sound from the stereo source twice. [21:44:46 CEST] <jjohn> /opt/local/ffmpeg/bin/ffmpeg -threads 7 -thread_queue_size 4096 -f alsa -ac 2 -i loop_out -c:a pcm_s16le -f matroska pipe:1 | /opt/local/ffmpeg/bin/ffmpeg -threads 7 -thread_queue_size 4096 -i pipe:0 -thread_queue_size 4096 -f alsa -ac 1 -i microphone -f matroska -map 0 -map 1 pipe:1 | /opt/local/ffmpeg/bin/ffmpeg -threads 7 -i pipe:0 -filter_complex "[0:1][0:1]amerge=inputs=2[a1];[0:0]volume=0.20:precision [21:44:52 CEST] <jjohn> =fixed[a0];[a1]volume=6.0:precision=fixed[a2];[a0][a2]amix[aout]" -map [aout] -c:a libfdk_aac -cutoff 15000 -b:a 128k -f flv -y /home/jjohn/test_recording.flv [21:44:56 CEST] <jjohn> What a monster. [21:47:42 CEST] <c_14> eeeeh, try using asetpts to set the timestamps to RTCTIME (using time(0) or something) [21:48:05 CEST] <jjohn> Wha?! [21:48:07 CEST] <c_14> for each of the 2 processes capturing from alsa on the stream that the process is capturing [21:48:37 CEST] <c_14> (I'm thinking that the "echo" is coming because one of the two processes is starting slightly delayed from the other so the timestamps don't match up perfectly) [21:48:53 CEST] <c_14> By swapping the timestamps out with your computer's time, that should go away [21:48:55 CEST] <c_14> (hopefully) [21:49:45 CEST] <c_14> basically something like -filter_complex '[0:0]asetpts=time$0$[out]' -map '[out]' <- just make sure that you only apply that to the stream you're currently capturing and that the second process maps both stream [21:49:55 CEST] <c_14> In this case, you could probably go back to your old setup with the fifos though [21:50:00 CEST] <c_14> That might make it easier [21:50:06 CEST] <c_14> Since your last command produced the same result [21:51:35 CEST] <jjohn> So ... only apply the asetpts filter to the ffmpeg process that is mixing both sources ... correct? [21:51:54 CEST] <c_14> No, to each of the source processes [21:51:59 CEST] <c_14> The ones capturing from alsa [21:52:16 CEST] <jjohn> OK, I'll try. [21:55:48 CEST] <jjohn> >[libfdk_aac @ 0x1ba49c0] Trying to remove 1024 samples, but the queue is empty [21:55:50 CEST] <jjohn> >[flv @ 0x1ba3ba0] Encoder did not produce proper pts, making some up. [21:55:55 CEST] <jjohn> >av_interleaved_write_frame(): Immediate exit requested [21:56:20 CEST] <jjohn> With this command: /opt/local/ffmpeg/bin/ffmpeg -threads 7 -thread_queue_size 4096 -f alsa -ac 2 -i loop_out -c:a pcm_s16le -filter_complex '[0:0]asetpts=time$0$[aout]' -map [aout] -f matroska pipe:1 | /opt/local/ffmpeg/bin/ffmpeg -threads 7 -thread_queue_size 4096 -i pipe:0 -thread_queue_size 4096 -f alsa -ac 1 -i microphone -filter_complex '[1:0]asetpts=time$0$[aout]' -f matroska -map 0 -map [aout] pi [21:56:26 CEST] <jjohn> pe:1 | /opt/local/ffmpeg/bin/ffmpeg -threads 7 -i pipe:0 -filter_complex "[0:1][0:1]amerge=inputs=2[a1];[0:0]volume=0.20:precision=fixed[a0];[a1]volume=6.0:precision=fixed[a2];[a0][a2]amix[aout]" -map [aout] -c:a libfdk_aac -cutoff 15000 -b:a 128k -f flv -y /home/jjohn/test_recording.flv [21:57:01 CEST] <jjohn> ... should we just cancel it off as a bug in ffmpeg? :/ [21:59:37 CEST] <c_14> You'd probably have to (in the third command) find the first common pts and cut all samples before that, then reset the PTS to start from zero [21:59:39 CEST] <c_14> or something [22:00:09 CEST] <jjohn> How? [22:01:45 CEST] <c_14> An excellent question, I was hoping it'd happen automatically but apparently it didn't. [22:01:57 CEST] <c_14> though [22:02:11 CEST] <c_14> Can you try outputting to pcm_s16le into a wav file? [22:02:29 CEST] <c_14> Cause if that doesn't error we could just hang another ffmpeg onto the back [22:02:50 CEST] <jjohn> After the first or the second process? [22:03:02 CEST] <c_14> the third one [22:03:16 CEST] <c_14> so after the second [22:03:25 CEST] <c_14> But try outputting to a file first [22:04:26 CEST] <jjohn> I thought wav wouldn't function with two pcm_s16le streams? [22:04:39 CEST] <jjohn> Shouldn't I use mkv then? [22:04:41 CEST] <c_14> Well, this is after you mix them [22:04:46 CEST] <c_14> so there's only 1 stream left [22:07:14 CEST] <jjohn> Just let me get this straight: first process records stereo, pipes it to second process, second process takes the data from the first process, records mono, saves both streams into a mkv container, and pipes the data to the third process. The third process now should mix them and write them (without encoding, as I always did before) to the disk as a single WAV. [22:08:10 CEST] <c_14> Well, it's still encoded. It's just encoded as raw pcm [22:08:32 CEST] Action: c_14 is hoping that that isn't as picky with the timestamps and will just write it out. [22:08:44 CEST] <jjohn> OK, without lossy encoding then. [22:14:53 CEST] <jjohn> OK ... I changed it to "/opt/local/ffmpeg/bin/ffmpeg -threads 7 -i pipe:0 -filter_complex "[0:1][0:1]amerge=inputs=2[a1];[0:0]volume=0.20:precision=fixed[a0];[a1]volume=6.0:precision=fixed[a2];[a0][a2]amix[aout]" -map [aout] -c:a pcm_s16le -y /home/jjohn/test_recording.wav" - it maps to "amix -> Stream #0:0 (pcm_s16le)" ... but then it just hangs. The last ffmpeg process just comes to a halt. [22:15:20 CEST] <jjohn> pressing "q" does not do anything. Pressin CTRL + C gives me "Could not write header for output file #0 (incorrect codec parameters ?): Immediate exit requested". [22:16:01 CEST] <jjohn> And to be honest - this is just too big a workaround for something as simple as recording two streams and mixing them properly. [22:16:45 CEST] <c_14> q doesn't do anything because it's already using stdin for input, and yeah no idea why recording from 2 alsa sources causes that quality loss [22:19:44 CEST] <jjohn> So ... ? [22:22:35 CEST] <c_14> The only other thing I could put out there would be to do all the mixing alsa side, and then only capture that with ffmpeg [22:22:40 CEST] <c_14> Something like https://trac.ffmpeg.org/wiki/Capture/ALSA#Recordaudiofromanapplicationwhile… [22:22:51 CEST] <c_14> just with a bit of volume changing first [22:23:02 CEST] <c_14> hmm [22:23:14 CEST] <c_14> actually iirc you should just have to change the ttable values [22:23:32 CEST] <c_14> Though you'd have to duplicate the mono to stereo [22:26:12 CEST] <jjohn> And what is the common approach to dublicate the channel in ALSA? [22:27:12 CEST] <c_14> mhm, actually you should only have to modify that ttable and the slaves.a.channels a bit [22:27:20 CEST] <c_14> sec [22:29:58 CEST] <c_14> http://pb.c-14.de/t/kng.CHYLtA [22:30:25 CEST] <c_14> except I just deleted the wrong lines [22:30:36 CEST] <c_14> http://pb.c-14.de/t/kng.ZG5sSg [22:31:17 CEST] <jjohn> .de domain? Also auch ein Deutscher? ;) [22:32:03 CEST] <c_14> Nur wenn keiner hinschaut. [22:51:40 CEST] <jjohn> Hm ... that config just gives me the error messages "Invalid or missing schannel for channel 0" and "cannot open audio device multi (Invalid argument)". [22:56:44 CEST] <c_14> eh, right [22:56:47 CEST] <c_14> got the order mixed up [22:57:06 CEST] <c_14> at least I think I ded [22:57:08 CEST] <c_14> *did [22:57:29 CEST] <c_14> swap all the .numa.numb in the ttable so its .numb.numa [22:58:54 CEST] <jjohn> table.0.0 6; ttable.0.1 6; ttable.2.0 0.2; ttable.3.1 0.2; ? Doesn't work either. [23:00:36 CEST] <c_14> Hmm, try asking #alsa maybe. My alsa really isn't up to par. [00:00:00 CEST] --- Thu May 7 2015

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