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January 2014

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ffmpeg-devel.log.20140105
by burek 06 Jan '14

06 Jan '14

[00:07] <cone-115> ffmpeg.git 03Alexander Strasser 07master:77015443a84b: lavf/file: file_check: Handle file URLs that start with "file:" [00:25] <ubitux> ok, found my bug [00:25] <nevcairiel> \o/ [00:25] <ubitux> i'm obviously not unpacking correctly [00:25] <ubitux> i need a signed b->w unpacking [00:25] <ubitux> i was using the <0 mask as an unpack 2nd operand [00:26] <ubitux> (so it was unpacking with ff with <0 and 0 with>=0) [00:26] <ubitux> but obviously it's not the correct representation for <0 ones [00:27] <ubitux> still not sure how to do that [00:27] <ubitux> mmh actually it should be [00:27] <nevcairiel> yes, it should [00:28] <nevcairiel> extending a signed value means you put the signed bit in all the new bits [00:28] <nevcairiel> so 0 or ff should be fine [00:28] <ubitux> yeah well something is wrong around here anyway [00:34] <kurosu_> punpack{h,l}bw with itself, psraw 8 [00:35] <kurosu_> or the bxb->w instruction whatever its name is [00:36] <kurosu_> pmaddusbw or something [00:39] <ubitux> wtf is going on... [00:45] <ubitux> oh FFS [00:46] <ubitux> this time i got the bug for real [00:47] <ubitux> i'm doing clip_i8(i8 - i8) instead of clip_i8(u8 - u8) [00:47] <ubitux> how am i going to do this now. [00:48] <ubitux> kurosu_: no that trick was fine, my brain is just melting [00:48] <nevcairiel> putting it in the top bits and doing a signed shift may be easier though [00:48] <nevcairiel> arithmetic shift or how its officially called [00:49] <ubitux> well it's just one additionnal instr., it's really fine [00:56] <cone-115> ffmpeg.git 03Stefano Sabatini 07master:98ecbf0093e5: doc/protocols/file: document general file protocol URL syntax [00:56] <cone-115> ffmpeg.git 03Stefano Sabatini 07master:92e145acbb9b: doc/protocols/file: fix semantical reverse [00:56] <cone-115> ffmpeg.git 03Stefano Sabatini 07master:22fa50d15938: lavf/file: fix help message first character casing for trunc option [01:02] <ubitux> brain is off, i'll solve that tomorrow i think [01:11] <cone-115> ffmpeg.git 03Michael Niedermayer 07master:7546ac2fee40: avformat/mp3dec: fix start time in light of initial skip samples [01:49] <cone-115> ffmpeg.git 03Alex Converse 07master:e2096e2eaa9e: fate: Add a downsampled SBR testvector [01:49] <cone-115> ffmpeg.git 03Michael Niedermayer 07master:c69906151b50: Merge commit 'e2096e2eaa9e75663d6bf0c37d342752aa5a146d' [02:11] <cone-115> ffmpeg.git 03Alex Converse 07master:b2212dec0f01: aac: Fix TNS decoding for the 512 sample window family. [02:11] <cone-115> ffmpeg.git 03Michael Niedermayer 07master:4195ae0fd815: Merge commit 'b2212dec0f011893ec68eecaa990170fa24050d7' [02:20] <cone-115> ffmpeg.git 03Alex Converse 07master:42d1b4198397: fate: Add a test vector for AAC ELD with TNS. [02:20] <cone-115> ffmpeg.git 03Michael Niedermayer 07master:f200ec20b815: Merge commit '42d1b41983971da63302ac3d12091cad1f3d6324' [02:26] <cone-115> ffmpeg.git 03Alex Converse 07master:7f29644108c5: aac: Fix low delay windowing. [02:26] <cone-115> ffmpeg.git 03Michael Niedermayer 07master:fd53f9d985c9: Merge commit '7f29644108c5fbd80f160930b31b78b8704c1a49' [02:39] <cone-115> ffmpeg.git 03Alex Converse 07master:9d18a7d3ec09: fate: Update AAC ELD 5.1 ref for recent bugfixes. [02:39] <cone-115> ffmpeg.git 03Michael Niedermayer 07master:a57da850c0ae: Merge commit '9d18a7d3ec09d6d933d648570643fde924aa391a' [02:45] <cone-115> ffmpeg.git 03Martin Storsjö 07master:82b9799bb211: sdp: Check that fmt->oformat is non-null before accessing it [02:45] <cone-115> ffmpeg.git 03Michael Niedermayer 07master:70937d9708bc: Merge remote-tracking branch 'qatar/master' [04:53] <cone-115> ffmpeg.git 03Peter Ross 07master:f5f6e59495a3: avformat/matroskaenc: warn when muxing video codec not supported by format [13:14] <ubitux> fixed, but another mismatch :( [13:14] <ubitux> it never ends :p [13:14] <ubitux> hopefully the last one this time [13:28] <nevcairiel> where would be the fun if it worked right away [13:33] <ubitux> :( [13:33] <ubitux> it's easier to debug though now [13:33] <ubitux> since i have a shitload of c printing everywhere [13:38] <BBB> that's useful [13:39] <BBB> do any of the samples decode correctly now (and still actually invoke this code)? [13:39] <ubitux> yes, the first one at least passes [13:39] <ubitux> (07 one) [13:39] <ubitux> i still have a single off-by-one in 08 [13:40] <ubitux> dunno about the rest, maybe hidding some other surprises [13:44] <cone-251> ffmpeg.git 03Diego Biurrun 07master:52ccc4a0ece8: configure: Support preprocessor macros as header names [13:44] <cone-251> ffmpeg.git 03Michael Niedermayer 07master:05286b6a9cc3: Merge remote-tracking branch 'qatar/master' [13:59] <ubitux> ah, found the bug... [13:59] <ubitux> annoying one :( [15:53] <kurosu_> BBB: probably you aren't interested, but do you think the tricks with few dct coeffs for idct would work for h264? or is its ict too simple to show benefits? [15:54] <kurosu_> haven't checked in details, but I guess you just check the number of coeffs. In the jpeg implementation, they were doing such things for vertical/horizontal-only coeffs [15:55] <wm4> Daemon404: whatever happened to your idea to add libcurl to libavformat? [16:09] <BBB> kurosu_: it's worth trying, I've had very mixed results in vp9 [16:09] <BBB> kurosu_: the gains in h264 will not be as big since even the biggest transform (8x8) still fits in a single iteration (unlike 16x16, which requires 2 16x8s) [16:10] <BBB> 16x8 as in 1d idcts over half the rows/cols of coefficients [16:10] <BBB> kurosu_: it's made harder b/c h264 has things like doing 2 4x4s at a time for inter blocks, and then this gets very hard very quickly [16:11] <kurosu_> 2 4x4 at a time? that's a decoder trick, right ? [16:12] <kurosu_> in that case, I guess so [16:13] <kurosu_> and probably wouldn't apply to hevc/vp9 because 4x4 usually means high bitrate/intra and therefore the trick hardly applies [17:05] <BBB> kurosu_: right, or lossless [17:05] <BBB> kurosu_: it may work for the 8x8 in h264, but I don't know how common that is [17:05] <BBB> I guess it can happen at higher resolution [17:06] <BBB> like I said, worth a try but I'm not sure it'll be as transferrable [17:19] <BBB> ubitux: ok sub8x8 is now on github also, I think I'm done with this [17:19] <BBB> ubitux: let me know when lf is done and we'll do some blogpost to announce that it's "good enough". other simd (intra pred and iadst) and other minor optimizations can be done later after that [17:20] <BBB> I have many ideas, but we can call it finished anytime after lf is done I think [17:20] <ubitux> i found the bug, i need to think a bit about it, maybe done tonight, then cleanups and stuff, and i'll need to do the transpose so the horizontal lpf can work [17:23] <ubitux> (remaining bug is that i'm re-reading already processed input in filter14(), so i need to review my "read-dep tree" :p) [17:25] <BBB> ah right [17:25] <BBB> shall I re-send patches to ML [17:25] <BBB> or is github OK? [17:25] <BBB> (also for michaelni once we're ready to merge) [17:26] <ubitux> github is fine with me [17:26] <ubitux> but i've something to do right now, bbl [17:26] <cone-251> ffmpeg.git 03Federico Simoncelli 07master:b1ad93123317: v4l2: setting device parameters early [17:28] <BBB> yeah no hurry [17:28] <cone-251> ffmpeg.git 03Michael Niedermayer 07release/1.2:71b3235cea40: avformat/oggdec: dont read timestamps from EOS pages of ogm videos [17:28] <cone-251> ffmpeg.git 03Michael Niedermayer 07release/2.1:8763aca389f0: avformat/oggdec: dont read timestamps from EOS pages of ogm videos [18:18] <Daemon404> [14:55] <+wm4> Daemon404: whatever happened to your idea to add libcurl to libavformat? <-- on my Things I'd Love To Do list [18:54] <cone-251> ffmpeg.git 03Michael Niedermayer 07master:d04aceb7d032: avformat/nutdec: remove unused variable [18:54] <cone-251> ffmpeg.git 03Federico Simoncelli 07master:b13d6c837fd7: pulse: set time_base as multiple of sample_rate [18:54] <cone-251> ffmpeg.git 03Federico Simoncelli 07master:b53d6ce3fdef: pulse: get latency only when needed [19:16] <ubitux> yay it's now passing 6 of them. [19:16] <ubitux> ...and yet another mismatch :)) [19:22] <BBB> ubitux: \o/ gj [19:22] <ubitux> not yet :( [19:22] <BBB> 6 is pretty good [19:25] <ubitux> YAY [19:25] <ubitux> IT FUCKING WORKS [19:26] <JEEB> \o/ [19:26] <ubitux> now comes the cleanup and shit [19:29] <BBB> ubitux: weeh [19:30] <BBB> uh weird [19:30] <BBB> can I not log with a parsing context? [19:32] <ubitux> ? [19:32] <ubitux> av_log(NULL ... works for me [19:32] Action: ubitux is using av_log(0,0,...) all the time [19:35] <BBB> see ML (or that trac ticket from ami_stuff just now) [19:35] <BBB> I know you filed one also but I didn't care yet so far [19:35] <BBB> I can look if you want [19:36] <BBB> hm frame size change [19:36] <BBB> I hate it when that happens [19:37] <BBB> I sort of expected that to not work, I'll fix it [19:37] <BBB> ideally we'd make scalable work at some point, but I just couldn't care less about scalable [19:49] <ubitux> 17553 decicycles in loop_filter_v_16_16_c, 8387632 runs, 976 skips [19:49] <ubitux> 3977 decicycles in ff_vp9_loop_filter_v_16_16_ssse3, 8387836 runs, 772 skips [19:50] <ubitux> i think that's a bit better :) [19:50] <ubitux> i can probably win a few more cycles [19:51] <ubitux> BBB: weird error :p [19:51] <ubitux> probably stack thrashed or something, exploding elsewhere [19:53] <BBB> ? [19:53] <BBB> what error? [19:53] <BBB> oh in ami_stuff's bug? [19:53] <ubitux> yes [19:53] <BBB> no I was using a AVParseCodecContext as log ctx [19:53] <BBB> that's not legal [19:53] <BBB> so it died [19:54] <ubitux> ah [19:54] <BBB> so I changed it to an avcodeccontext and then it's fine [19:54] <BBB> we should probably fix that in avparsectx [19:54] <BBB> but that's a abi break so not now [19:55] <ubitux> LGTM :P [19:55] <BBB> I think I also know how to fix your crash but it's a little more involved [19:55] <BBB> review my github when lf cleanup is done, I'll review your code [19:55] <BBB> and then we're getting somewhere [19:55] <BBB> how much overall speedup? [19:55] <BBB> in wallclock decoding time [19:56] <kierank> 6:50 PM <"ubitux> 17553 decicycles in loop_filter_v_16_16_c, 8387632 runs, 976 skips [19:56] <kierank> 6:50 PM <"ubitux> 3977 decicycles in ff_vp9_loop_filter_v_16_16_ssse3, 8387836 runs, 772 skips [19:56] <kierank> nice [19:58] <BBB> kierank: we'll get another 10% per function with avx :) [19:58] <BBB> (or something like that) [19:58] <BBB> at least for the idcts [19:58] <ubitux> BBB: 8.32s 9.32s on ped1080p.webm with 1 thread [19:59] <BBB> huh [19:59] <ubitux> huh sorry [19:59] <BBB> it got slower? [19:59] <ubitux> the other way around [19:59] <ubitux> ofc [19:59] <BBB> lol [19:59] <BBB> ok cool [19:59] <ubitux> (tested c after) [19:59] <kierank> if there was a place in london I could colo a 1U box I could give you access to an avx2 machine [19:59] <BBB> I think with my and your patch we'll be beating libvpx then [19:59] <BBB> and we're not even done with simd [19:59] <BBB> so this is really very cool [20:00] <ubitux> :) [20:00] <ubitux> that's only the vertical one vtw [20:00] <ubitux> btw [20:00] <ubitux> i need to write the tranpose so we can have about 1 less second with horizontal [20:01] <BBB> cool [20:02] <BBB> which ones does this include? [20:02] <BBB> just 16_16? [20:02] <BBB> or 16_16, 16_8? [20:02] <BBB> or all 6 8/16 ones and the 2 mix ones? [20:02] <BBB> or some other combination? [20:03] <ubitux> just 16_16 [20:03] <ubitux> vertical [20:03] <BBB> oh whoa [20:03] <BBB> that's a massive speedup really for just that one [20:03] <BBB> good job [20:04] <BBB> I see about 50% of calls to v_16_8_c coming from v_16_16, and the other 50% from decode_frame [20:05] <BBB> which means if you make a mmx version that operates on 8 pixels (if that's possible; else just work on full sse registers and ignore the upper halves) you'll get another big gain [20:05] <BBB> (if you didn't already) [20:05] <BBB> hm I'm not sure that makes sense, this shouldn't happen except on picture edges [20:05] <BBB> ohwell [20:05] <BBB> what do I know [20:05] <BBB> anyway [20:05] <BBB> g [20:05] <ubitux> :p [20:06] <BBB> ood job [20:06] <BBB> maybe instruments isn't so good after all [20:16] <cone-251> ffmpeg.git 03Michael Niedermayer 07master:ee16e0cacc16: avfilter/vf_format: check that the format list is not empty [21:16] <smarter> <kurosu_> BBB: probably you aren't interested, but do you think the tricks with few dct coeffs for idct would work for h264? or is its ict too simple to show benefits? [21:16] <smarter> what trick? [21:38] <ubitux> 3936 decicycles in ff_vp9_loop_filter_v_16_16_ssse3, 8387211 runs, 1397 skips [21:38] <ubitux> faster \o/ [21:38] <ubitux> now i'm going to try < 3900 [21:40] <ubitux> oh yeah i can do something cool [21:48] <beastd> ubitux: what is it? [21:48] <ubitux> avoid 1 or 2 write [21:48] <ubitux> that might allow me to reach < 3900 [21:51] <cone-766> ffmpeg.git 03Ronald S. Bultje 07master:847072873c95: vp9_parse: don't use AVCodecParserContext as av_log context. [21:56] <ubitux> 3905 decicycles in ff_vp9_loop_filter_v_16_16_ssse3, 8387654 runs, 954 skips [21:56] <ubitux> aha! [21:57] <nevcairiel> your goal, you did not reach [21:58] <ubitux> yeah but i micro-optimized 300/500 lines [21:58] <ubitux> maybe there is some room for improvements later on [22:10] <cone-766> ffmpeg.git 03Tim Walker 07master:5c437fb672b6: lavu: Add values for various Dolby flags to the AVMatrixEncoding enum. [22:10] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:751385fe3f65: Merge commit '5c437fb' [22:20] <cone-766> ffmpeg.git 03Tim Walker 07master:5b4797a21db9: avframe: add AV_FRAME_DATA_MATRIXENCODING side data type. [22:20] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:4cf4da9dc581: Merge commit '5b4797a21db900b7d509660b7a4d49829089b004' [22:33] <cone-766> ffmpeg.git 03Tim Walker 07master:6bfdb2de8813: dcadec: set the output channel mode more accurately. [22:33] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:bc7f76377c1e: Merge commit '6bfdb2de881372048be7fbda643417e1fd3ce93c' [22:38] <cone-766> ffmpeg.git 03Tim Walker 07master:30d70e79a6b4: dcadec: set AV_FRAME_DATA_MATRIXENCODING side data. [22:39] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:ae01af24756d: Merge commit '30d70e79a6b4ac7f4eb66446a9da275161ef6ea7' [22:52] <ubitux> ok i don't think i'll reach below [22:52] <ubitux> :( [22:58] <cone-766> ffmpeg.git 03Tim Walker 07master:4b7f1a7ced0e: mlp: Parse TrueHD decoder channel modifiers and set the AVMatrixEncoding for each substream. [22:58] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:b4107f7805be: Merge commit '4b7f1a7ced0e98f2cc698d896f7ebab8d30eaa09' [23:06] <cone-766> ffmpeg.git 03Tim Walker 07master:e92123093dfd: mlpdec: set AV_FRAME_DATA_MATRIXENCODING side data. [23:06] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:85b424a45e38: Merge commit 'e92123093dfdca0ef6608998240e2f9345d63bff' [23:14] <cone-766> ffmpeg.git 03Tim Walker 07master:13345fc1f86f: (e)ac3: parse and store the Dolby Surround, Surround EX and Headphone mode flags. [23:14] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:7b3c78b5e6de: Merge commit '13345fc1f86fc3615789e196d5a339c1c27c9068' [23:21] <cone-766> ffmpeg.git 03Tim Walker 07master:7840c40445c9: (e)ac3dec: set AV_FRAME_DATA_MATRIXENCODING side data. [23:21] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:d18234f31915: Merge commit '7840c40445c9f52aeccba96de3d27613398bfbf2' [23:29] <cone-766> ffmpeg.git 03Johan Andersson 07master:7ce88e5ec414: cmdutils: update copyright year to 2014. [23:29] <cone-766> ffmpeg.git 03Michael Niedermayer 07master:67999d3d1275: Merge remote-tracking branch 'qatar/master' [00:00] --- Mon Jan 6 2014

1 0

ffmpeg.log.20140105
by burek 06 Jan '14

06 Jan '14

[00:01] <rolf> I have a problem with an HD movie which I get converted to .mp4 under linux but not under Windows 7 64 Bit. This is the problem: http://pastebin.com/izDWUCMW [00:09] <Mavrik> rolf, I'm rather sure that vo_aacenc audio encoder does not support 5.1 audio [00:09] <Mavrik> you probably have different versions of ffmpeg and they choose different audio streams to transcode [00:09] <Mavrik> since your input has 3 different audio streams, two stereo and one 5.1 [00:10] <rolf> i do not need this 5.1, I only need stereo [00:11] <Mavrik> well then use one of the stereo inputs as output [00:11] <Mavrik> or tell ffmpeg to remix audio to stereo from 5.1 [00:11] <Mavrik> depending what you want [00:11] <Mavrik> see documentation on "-map" parameter. [00:12] <rolf> -map under windows? I am at the moment on my linux, is the output here the same? [00:14] <rolf> ok, ffmpeg -help under linux tells: Advanced options: [00:14] <rolf> -map file.stream[:syncfile.syncstream] set input stream mapping , I will try to understand what to do. [00:23] <BeWilled> how do I improve the quality of the jpg file output using this command -filter:v scale=150:-1 -vframes 1 -codec:v mjpeg [00:23] <BeWilled> right know I am getting a 4 kb jpeg file [00:23] <relaxed> BeWilled: -q:v 1 [00:24] <relaxed> ^^ should give you the highest quality [00:24] <relaxed> the -q_v scale goes from 1 to 31 [00:24] <relaxed> er, -q:v scale [00:31] <rolf> Mavrik, thank you very much, it is running on windows now. I have this options: -map 0:1 -ar 48000 [00:32] <rolf> but I have an AMD PhenomII X4 and ffmpeg is using ony one core, is there an option to use all 4? [00:32] <relaxed> -threads 4 [00:32] <relaxed> although a recent ffmpeg should use them all by default [00:33] <BeWilled> thank you relaxed [00:33] <rolf> relaxed, thank you, I will try [00:34] <relaxed> you're welcome [00:43] <BeWilled> need help with this error while extracting images from flv http://pastebin.com/KAxvxfhZ [00:44] <relaxed> I see no error. Always provide the command in your pastebin. [00:45] <BeWilled> muxing overhead -100.00000 its -filter:v scale=250:-1 -vframes 1 -codec:v mjpeg -q:v 1 [00:46] <relaxed> what is the complete command and what's your goal? [00:47] <BeWilled> ffmpeg -ss 20 - i intput.flv -filter:v scale=250:-1 -vframes 1 -codec:v mjpeg -q:v 1 output.jpg [00:47] <rolf> with -map 0:1 the generated .mp4 has only the german stereo sound. I do not understand how to set correct options for -map . [00:48] <BeWilled> I'm succesfully extracting jpgs from flv files in a batch process... but seem to be getting this error on a specific file [00:49] <rolf> or this is to to with two steps? first to get the audio and the the rest? [00:49] <relaxed> you only want one frame? does that command output an image? [00:50] <BeWilled> yes [00:50] <relaxed> rolf: pastebin ffmpeg -i yourinput [00:50] <relaxed> BeWilled: there were two questions there [00:50] <relaxed> hurry, because I'm heading out for pints [00:50] <BeWilled> yes, and yes [00:50] <relaxed> great, then what's the problem? [00:51] <BeWilled> muxing overhead -100.000000% [00:51] <relaxed> why are you hung up on that? [00:52] <relaxed> there's no error- move on [00:52] <relaxed> do you know how to check the exit status of a command? [00:52] <relaxed> to see if there was an error? [00:53] <BeWilled> no [00:53] <relaxed> echo $? [00:53] <relaxed> if it's zero there was no error [00:54] <BeWilled> ok [00:54] <relaxed> or: ffmpeg -i .... || echo "Houston, we had a problem" [00:54] <relaxed> ffmpeg makes it very clear when there's actually an error [00:55] <relaxed> And for more precise seeking you may want to move the -ss after the input. It will be a little slower though. [00:55] <relaxed> rolf: waiting for pastebin.. [00:56] <relaxed> nevermind scrolled up. you want the french audio? [00:59] <rolf> yes , one moment [01:00] <rolf> http://pastebin.com/YUaTBCMY [01:00] <relaxed> ok, what is your goal? [01:01] <rolf> to get video and audio combined in one .mp4, audio only german stereo [01:03] <relaxed> you want german? [01:03] <rolf> Stream #0:0 and Stream #0:1 I think I want to have [01:03] <rolf> yes german [01:04] <relaxed> ffmpeg -i input -map 0:0 -map 0:1 -c copy -f mpegts output.ts [01:29] <rolfgerman> http://pastebin.com/PZqMQcj1 I think I solved my prolblem, he is on step 2, tomorrow I will see if the .mp4 is OK. Thank you very much, thank you very much relaxed for your help. [01:46] <average> is it possible to draw an arrow with ffmpeg on a video ? [01:46] <average> I know drawtext is possible and I'm using it [01:47] <sacarasc> You can overlay images, I don't know how myself, but I know it can be done. [01:48] <sacarasc> Using the overlay filter, I think. [01:48] <average> I could do it with imagemagick , separating video, audio, superimposing image over frames in video, reassemble video, then merge video/audio again, but that would be tedious [01:48] <average> well, yeah, the overlay filter, but that would require all this disassmble/reassemble thing. I don't mind it that much, I did it for a sox-based noise-reduction, it's fine but.. it's a bit tedious [01:49] <average> I was hoping if ffmpeg has drawtext it should maybe has some sort of "drawarrow" somewhere.. [01:49] <average> ofcourse that is just a hope.. and I don't have anything that leads me to believe that such a thing actually exists [01:49] <konflict> Hi, Im currently trying to solve -timecode sync issue. Im piping two SDI streams from BlackMagic Decklink into ffmpeg to capture two DNXHD mxf files. I need to get identical timecodes on both clips based on either system time or some external source for later "frame precise" editing. So far I wasnt able to find any solution. Could anyone help? [01:51] <average> konflict: what's a timecode ? [01:51] <average> konflict: sorry I'm like.. clueless on that topic.. [01:55] <average> konflict: is timecode that piece of information that lets the video player know how big the video is and stuff like that ? [01:55] <average> like the current second etc [01:55] <average> ? [01:55] <konflict> No worries, thanks for feedback in the first place. Well its basically time information for each individual frame of the video. Its significant say when you want to sync video and audio from separate sources, if both of them have timecode you can sync them perfectly. [01:56] <average> how can you check if both sources have timecode ? [01:57] <konflict> well most of the NLEs like premiere, final cut or media composer supports this feature as its crucial for multicam editing [01:57] <konflict> in my scenario my sources doesnt have any but i want to add real time as my reference timecode [01:57] <average> also, have you seen this ? http://stackoverflow.com/questions/18253340/keep-timecode-in-ffmpeg/1825949… [01:57] <konflict> it that make sense :) [01:58] <average> options -copyts or -copytb <mode> with <mode> one of { 1 , 0 , -1 } [01:58] <average> what you wrote makes sense [01:59] <average> moreover, I also need it for one of my things because it seems that somewhere across the post-processing I mess up my timecodes and the video player does not know how long the video is anymore [02:00] <average> and doesn't know how to seek properly in the video [02:00] <average> so I guess that's timecode-related too right ? [02:01] <konflict> well I will certainly give this a try. You see even without timecode syncing two camera feeds would be piece of cake. Just find reference point like light flash or something audible. But I want to capture between 4 - 12 streams for multicam edit and that would be very time consuming and whats worse probably inaccurate [02:02] <average> konflict: then maybe you will have to inject some sort of external signal so that you can check for it in post-processing ? [02:02] <average> for example a red-dot somewhere [02:03] <average> konflict: but your cameras are filming the same place/area but from different angles. is that correct ? [02:03] <konflict> exactly [02:04] <relaxed> average: you could overlay a transparent png of an arrow or try to use drawtext with a unicode arrow, www.alanwood.net/unicode/arrows.html [02:04] <average> and you want to join all the cameras in one single video in some sort of 3x4 grid (if you have 12 streams for example ) ? [02:05] <relaxed> https://trac.ffmpeg.org/wiki/Create%20a%20mosaic%20out%20of%20several%20inp… [02:05] <average> konflict: are you working on a live stream or something that you have time to post-process on ? [02:05] <konflict> not really, I want to keep them as they are in full resolution and just edit cuts between them [02:06] <konflict> well, Im trying to shoot in small studio then post process end then distribute [02:06] <average> oh I see, like a football match for example, where different angles are needed in different moments of the game.. [02:07] <konflict> yeah, thats exactly it [02:07] <average> don't your cameras start recording all at the same time ? [02:07] <average> if you have them starting all at the same time then 01:20:31 will mean the same thing for all of them [02:08] <average> but I suspect they don't start at the same time for some reason [02:08] <average> and you don't know the relative differences between them.. or do you ? [02:08] <average> or you want to close some of them sometimes and they de-sync for that reason [02:09] <konflict> well Im capturing with ffmpeg and every "capture" starts in slight offset to previous. Its not not much, difference is in seconds maybe frames only but its still there and its noticeable [02:11] <average> ok, let's suppose you have the videos already stored somewhere 1.avi , 2.avi , 3.avi . [02:12] <average> you can put a bulb, led, whatever, somewhere in the set, as you said [02:12] <konflict> so this way if I specify -timecode 00:00:00,00 starting point will be same for all recorded clips but it wont represent same point in real time when recorded [02:12] <average> yeah, I understood [02:12] <average> thanks for the explanation btw [02:12] <average> here's what I think [02:12] <average> you talked about a light, that's a very good idea [02:12] <average> now what's left, is identifying that light [02:14] <average> maybe you don't want to identify light, maybe you throw some object in the image to sync http://www.imagemagick.org/script/command-line-options.php#subimage-search [02:14] <konflict> Yes, you are right because this is certainly one way. Im not editor really but afaik most of pro editing stations allow you to specify reference point for each individual clip and it will line up on timeline automatically [02:14] <average> you can search for that object with compare -subimage-search from ImageMagick [02:16] <konflict> interesting, thank you for the link... [02:17] <average> but in order to search for light, you can just check the frame luminocity or brightness http://www.imagemagick.org/discourse-server/viewtopic.php?f=6&t=24046 [02:18] <average> here's another one that finds the average gray-level http://www.wizards-toolkit.org/discourse-server/viewtopic.php?f=1&t=14073#p… [02:18] <average> which is also for the purpose of brightness level [02:19] <konflict> thank you sir :) [02:20] <average> no problem. you will have to take frames and so forth, but, maybe it will work :) I wish you the best of luck :) [02:21] <konflict> even though I still not sure if thats going to work, I guess it wont be so difficult feature to add. I mean for example take the -timecode information from system time or extract it from audio input which is by the way typical way of distributing timecode [02:21] <konflict> anyway, thank you for the input. Definitely interesting ideas [02:21] <konflict> good luck with your project too [02:31] <average> "Dejavu can memorize audio by listening to it once and fingerprinting it. Then by playing a song and recording microphone input, Dejavu attempts to match the audio against the fingerprints held in the database, returning the song being played" [02:31] <average> https://github.com/worldveil/dejavu#dejavu [02:31] <average> konflict: this is the method you mentioned just now. of finding things through audio input [02:31] <average> you can use dejavu [02:31] <average> it's written in python [02:32] <average> you'll probably have to adapt it to your needs as it's made for a much general purpose I think (it's made for things like soundhound.com or midomi.com or shazam) [02:36] <konflict> Thank you so much for keeping me in mind. This is actually very nice project and I think this could work provided coding skills which I lack almost completely. This certainly is beyond my level :) [02:36] <konflict> But hey, thank you! [03:10] <average> konflict: hope is not lost my friend, you can still do it. here is codecademy, where you can learn some python for the greater good :) http://www.codecademy.com/tracks/python [03:12] <Fusl> is ffmpeg buffering stdin? [03:12] <Fusl> pipe:0? [03:14] <Fusl> i'm reading pipe:0 with -re and try to omit overflows in node.js at the same time but after a couple of time input gets unsync and it takes ffmpeg a minute to hear a sound we injected into the stream :/ [06:58] <jimi_> I have a movie with audio. I want to keep the first N second of video and audio, but then after N seconds to end of file, I want only video, how can I do this? [07:10] <DeadSix27> jimi_: i'd cut the audio to N seconds, then mux it with the full video it should continue to play even though the sound has ended [07:11] <DeadSix27> atleast thats what happends when i merge a shorter audio and a longer video all the time [07:12] <DeadSix27> but that ofc only works when its supossed to be at beginning [07:13] <jimi_> DeadSix27, yes, i want to keep seconds 0-N, and then replace N-EOF with another audio track, all over the same underlying video [07:16] <DeadSix27> ah dunno how to do that [07:16] <DeadSix27> with ffmpeg i mean. [07:16] <jimi_> DeadSix27, ill google abit just adnt found antyhing [07:17] <DeadSix27> maybe you can cut 2 audios, and then align them both together and then mux em [07:17] <DeadSix27> the only thing i dont know then, is how to align them both together [07:18] <DeadSix27> maybe this helps: https://trac.ffmpeg.org/wiki/How%20to%20concatenate%20(join,%20merge)%20med… [08:35] <Zeranoe> Is the -b:a for fdk-aac per channel? Meaning, if I set the bitrate to 128k, will that be for the total, or per channel [08:40] <Zeranoe> I see that yes it is per channel [11:56] <haarp> why does ffmpeg in example -1- of http://pastebin.com/Li3dHQk8 display wrong stats? the time is counting up way too fast, although the stream is obviously realtime. it doesnt happen in example -2-, where it can recognize a duration. is this a bug? should i file it? [13:58] <nano-> Isn't the following supposed to work? ffmpeg -i test.wav -ac 2 -c:a tta test.tta (input is a s16 3 second wav) [13:58] <nano-> I get: [13:58] <nano-> [NULL @ 0x7febdb02ca00] Unable to find a suitable output format for 'test.tta' [13:58] <nano-> test.tta: Invalid argument [14:00] <spaam> nano-: to old version of ffmpeg ? [14:01] <nano-> ffmpeg version 2.1.git-a044a18 Copyright (c) 2000-2013 the FFmpeg developers [14:02] <JEEB> tta _encoding_ ? [14:02] <nano-> 20131206 [14:02] <nano-> yes. encoding. [14:03] <JEEB> ok, there is a ttaenc... but is that lavf or lavc [14:03] <JEEB> ok [14:03] <JEEB> lavc [14:03] <JEEB> anyway, by that I would guess that there is no tta muxer [14:04] <JEEB> try muxing into mkv [14:04] <nano-> Tried. That works. But I don't hink that's what I want, and VLC didn't play it (which isn't the target anyway so that doesn't matter). [14:05] <JEEB> then you can extract that bit stream if you want to [14:05] <JEEB> or, if tta files are just the raw bit stream [14:05] <JEEB> you could test -f rawvideo out.tta [14:05] <JEEB> yes, that format sounds completely wrong but that generally means "just output the data" [14:06] <nano-> Humm.. Output file #0 does not contain any stream [14:07] <JEEB> oh, it might not work when encoding... [14:07] <JEEB> in any case, there is no tta muxer [14:07] <JEEB> just checked the source tre [14:07] <JEEB> *tree [14:07] <JEEB> there's a demuxer, decoder and even an encoder [14:07] <JEEB> I'm surprised paul actually implemented encoding because having Yet Another Lossless Audio Encoder is kind of getting out of hand [14:08] <JEEB> not that you gain much if anything compared to flac or wavpack [14:08] <nano-> Yep, perhaps I should just drop it instead of putting effort into generating test files :) [14:08] <JEEB> if you need test files [14:08] <JEEB> there are such on the FATE server [14:08] <JEEB> that's used for testing ffmpeg [14:10] <nano-> Thanks for your help. [14:11] <JEEB> np [14:12] <spaam> nano-: beer later today? [14:13] <nano-> spaam: Nah, going for a first run of the latest Pandemic expansion. [14:14] <spaam> nano-: okey :) [14:14] <nano-> http://boardgamegeek.com/boardgameexpansion/137136/pandemic-in-the-lab [14:29] <average> hello ? [14:30] <sacarasc> Greetings. [14:30] <average> do you know someone who makes music ? [14:31] <average> I'm making a screencast and I need some music to put in the intro [14:31] <average> I'll give props/credits to the artist [14:31] <average> but I need a 12-15 second intro that sounds good for it [14:31] <sacarasc> I do, but as it's 05:30 for them currently, I don't think they're around. [14:32] <average> sacarasc: hook me up [14:32] <average> sacarasc: and I'll talk to them when they're available [14:33] <sacarasc> Join us in #DigitalGunfire. [00:00] --- Mon Jan 6 2014

1 0

ffmpeg-devel.log.20140104
by burek 05 Jan '14

05 Jan '14

[00:04] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:5047849584f2: avformat/utils: fix order of buffers in timestamp update code [00:24] <cone-17> ffmpeg.git 03Anton Khirnov 07master:feded990e3ef: mpegvideo: set reference/pict_type on generated reference frames [00:24] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:532b93b51631: Merge commit 'feded990e3ef9af4a0b827d5b6d8fe86f0b94942' [00:47] <cone-17> ffmpeg.git 03Anton Khirnov 07master:a6a2282c25ab: rv30: fix extradata size check. [00:47] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:3cf1503beb35: Merge commit 'a6a2282c25abe43e352010a7c3fbc92994c0bc1c' [01:06] <cone-17> ffmpeg.git 03Anton Khirnov 07master:5569146d48f0: adx: check that the offset is not negative [01:06] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:c13e73c25b6d: Merge commit '5569146d48f06564e8fa393424782cceed510916' [01:13] <cone-17> ffmpeg.git 03Anton Khirnov 07master:24057c83207d: eacmv: check the framerate before setting it. [01:13] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:405b1ef898bf: Merge commit '24057c83207d6ea8bfd824155ac37be8a33dfd0c' [01:35] <cone-17> ffmpeg.git 03Anton Khirnov 07master:94a417acc05c: mathematics: remove asserts from av_rescale_rnd() [01:35] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:3929c17405e0: Merge commit '94a417acc05cc5151b473abc0bf51fad26f8c5a0' [01:42] <cone-17> ffmpeg.git 03Anton Khirnov 07master:1b5d065ca722: pmpdec: check that there is at least one audio packet. [01:42] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:4a055f91bec2: Merge commit '1b5d065ca722eb8028c7a08e054b6da3419faf5d' [02:06] <cone-17> ffmpeg.git 03Anton Khirnov 07master:e89aa4bf56e5: lzw: switch to bytestream2 [02:06] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:508229adb930: Merge commit 'e89aa4bf56e5b5c45f569eb12733519789e057da' [02:12] <cone-17> ffmpeg.git 03Anton Khirnov 07master:58312b2472d3: h264: reset data_partitioning if decoding the slice header for NAL_DPA fails [02:12] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:8e6af036b957: Merge commit '58312b2472d3a44d7458865c459d59ef2e02bf1a' [02:18] <cone-17> ffmpeg.git 03Anton Khirnov 07master:3d95d27376e5: audio_mix: initialize the data pointers to NULL [02:18] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:fadec4f6dbc7: Merge commit '3d95d27376e59de14f984e7a22a52e066d85df35' [02:31] <cone-17> ffmpeg.git 03Anton Khirnov 07master:fc6a3ef40d34: audio_mix: fix zeroing output channels in certain cases [02:31] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:4b8bc6d2b0a9: Merge commit 'fc6a3ef40d34ce8443ae57c2452f3f273d7d4891' [02:36] <cone-17> ffmpeg.git 03Anton Khirnov 07master:cc976a75dffa: audio_mix: print (SKIP) instead of 0.0 for matrix columns removed along with output zeroing [02:36] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:781fd58838ca: Merge commit 'cc976a75dffa148d655b52604331679ff669e8a2' [02:41] <cone-17> ffmpeg.git 03Anton Khirnov 07master:a8cc88b1a23d: tests/Makefile: allow FILTER* to be called with lists of filter names [02:41] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:cf867517f15b: Merge commit 'a8cc88b1a23dc1515f27cfa98af16a273c539091' [02:49] <BBB> ubitux: ok 32x32 works now, not too bad [02:51] <BBB> 7.6sec (libvpx) vs. 8.2sec (ffvp9) without lf simd [02:51] <BBB> and idct32x32 gave me about 1sec speedup [02:51] <BBB> so I think after lf simd we'll beat libvpx by quite a bit [02:53] <cone-17> ffmpeg.git 03Anton Khirnov 07master:b318106fae65: FATE: add a test for the lavr mixing case fixed in fc6a3ef40d34ce8443ae57c2452f3f273d7d4891 [02:53] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:ad5e4e0f947a: Merge commit 'b318106fae65149356934fc72feafef3272fd4ea' [03:03] <cone-17> ffmpeg.git 03Anton Khirnov 07master:aec25b1c4650: mpegvideo: split the encoding-only parts of frame_start() into a separate function [03:03] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:fb17f03dadc3: Merge commit 'aec25b1c4650944d32706bfd40eb02bbd5587303' [03:13] <cone-17> ffmpeg.git 03Anton Khirnov 07master:a4d0c6e05035: mpegvideo: move dct_unquantize functions up to avoid forward declarations [03:13] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:9b9f64fe315c: Merge commit 'a4d0c6e0503562d4cc8f9f6d02d84d7b32583b15' [03:21] <cone-17> ffmpeg.git 03Anton Khirnov 07master:a3a55645f09f: mpegvideo: remove disabled bfin asm [03:21] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:adc09a353c5f: Merge remote-tracking branch 'qatar/master' [04:36] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:958e31197431: avcodec/rv10: cleanup rpr handling [04:36] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:a881b9aa08af: avcodec/rv30: cleanup rpr handling [04:36] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:d08c6e1101aa: avcodec/rv30: check rpr before use against maximum [04:36] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:be524ffc16bf: avcodec/rv30: put the rpr check back in init with the max vs bits bug fixed [09:44] <cone-644> ffmpeg.git 03Clément BSsch 07master:307b6b8cb4f8: avfilter/lut3d: fix channel order in Iridas format parsing. [10:43] <ubitux> can i compare strings in yasm macro? [12:02] <ubitux> the lpf byte clipping in vp8/9 really is sick [12:35] <nevcairiel> if 8 also had, steal from there! :D [12:36] <BBB> ubitux: maybe check how vp8 did it (it's likely similar), or how libvpx' simd does it, if you're having trouble [12:41] <ubitux> nevcairiel, BBB, yes the part i'm messing with is mostly the same as vp8, and yes i'm looking [12:47] <BBB> and ask questions, sometimes I have ideas [12:47] <BBB> if this is still about clip_u8(u8+i8), what michaelni said is probably a good idea [12:48] <BBB> (i.e. split i8 over two registers, negative and positive, and then use add/sub instructions per-register) [12:48] <BBB> (paddusb/psubusb) [12:49] <ubitux> yes i did that [12:49] <ubitux> just like in vp8 [12:50] <ubitux> i'm trying to understand what you did for the (f1+1)>>1 [12:50] <ubitux> f1 being of course int8_t [12:50] <ubitux> (and 127>>1 ` 128>>1 so i cant just paddsb :p) [12:59] <ubitux> > According to the WebM mailing lists, the VP9 codec supports 4:4:4 chroma subsampling in profile 1. [12:59] <ubitux> ah? [13:03] <j-b> you mean, according to the spec, right? [13:04] <JEEB> > VPx > spec > laughing_schoolgirls.jpg [13:04] <j-b> libraries using abort > * [13:05] Action: JEEB ^5s j-b [13:07] Action: ubitux would use andthenisaid.jpg [13:09] <BBB> ubitux: sounds like a candidate for psrl{d,w,q}(pand(paddb(x, 1), 0xfe), 1) [13:09] <BBB> or pxor(pavgb(pxor(x, 0x80), zero), 0x80) [13:14] Action: j-b gives libdvdread/libdvdnav source code to JEEB and laughs [13:17] Action: JEEB quietly marks the package "undeliverable" and pokes it back to the mailer daemon [13:20] <cone-815> ffmpeg.git 03Martin Storsjö 07master:2ad4ee345a42: arm: Add a missing endfunc macro call [13:20] <cone-815> ffmpeg.git 03Michael Niedermayer 07master:63ce041a7d06: Merge commit '2ad4ee345a4216aef3999f57dd14c56128d27a13' [13:39] <cone-815> ffmpeg.git 03Martin Storsjö 07master:3348e3492d0c: arm: Use the matching endfunc macro instead of the assembler directive directly [13:39] <cone-815> ffmpeg.git 03Michael Niedermayer 07master:d8e65e9224b9: Merge remote-tracking branch 'qatar/master' [13:48] <BBB> ubitux: can I haz review plz? [13:53] <ubitux> BBB: give me a few hours plz :p [13:53] <ubitux> like 3-4 [13:53] <BBB> lol [13:53] <BBB> ok [13:53] <BBB> I'll work on a fixme to prevent another 3 shuffles in pass 2 and work on the sub-idcts or so [13:54] <BBB> kierank: can I borrow your sandybridge machine at some point in the next few days? I'd like to test some basic avx functions for speed and accuracy [15:04] <BBB> ubitux: also, my repo has some small modifications to the simd to fix an outstanding fixme (prevent 3 stack moves) [15:04] <BBB> ubitux: I can send new patch but basically look at github, not ML [15:04] <ubitux> noted :) [15:05] <BBB> it's 10-20 cycles faster overall in the nodc version, so quite a bit [15:05] <BBB> (it's basically what you'd expect) [15:06] <ubitux> btw, a load from rodata is generally faster than from stack, right? [15:06] <BBB> I'm not sure [15:07] <BBB> it all depends on where it is in your cache right? [15:07] <BBB> if you just put it there, it's proabbly in L1 = very fast [15:07] <BBB> if you put it there last week, it's probably not so fresh [15:07] <BBB> = slow [15:07] <BBB> rodate ... I don't know how that's scheduled, but I'm sure the schedulers are quite smart there [15:08] <BBB> maybe pengvado knows [15:08] <cone-815> ffmpeg.git 03Michael Niedermayer 07master:8113e838a811: avformat/nut: add support for per frame side & meta data with version 4 [15:22] <cone-815> ffmpeg.git 03Michael Niedermayer 07master:de5b6c736bcb: Changelog: add nuts side & metadata support [15:22] <cone-815> ffmpeg.git 03Michael Niedermayer 07master:b317f9459f4c: avutil/mathematics: add av_add_stable() [15:22] <cone-815> ffmpeg.git 03Michael Niedermayer 07master:cd7037dd7ac1: avutil/rational: av_add_stable() test code [15:22] <cone-815> ffmpeg.git 03Michael Niedermayer 07master:863f4c3c7170: avformat/utils: fix rounding error accumulation for generated dts in compute_pkt_fields() [15:32] <pengvado> BBB: caching is totally independent of sections [15:33] <BBB> ok, good to know, thanks [15:34] <BBB> pengvado: does the scheduler do anything clever w.r.t. pre-caching rodate sections? or is it just on-use? [15:37] <pengvado> I don't know what you mean by "pre-caching" [15:37] <pengvado> if you meant "prefetching" then no. [15:45] <ubitux> BBB: btw, do you have the same stack valgrind issue with 32x32? [15:51] <BBB> pengvado: yes I meant prefetching [15:51] <BBB> thanks [15:51] <BBB> ubitux: possibly, didn't check [15:51] <ubitux> + VP9_UNPACK_MULSUB_2W_4X 2, 13, 6270, m15137, [pd_8192], 0, 7 ; y21, t26 [15:51] <ubitux> t21* [15:52] <BBB> oh nice typo [15:54] <ubitux> BBB: http://pastie.org/8600340 i think m0 is scrapped in this area, it shouldn't be in the comments, no? [15:54] <BBB> fixed (locally) [15:55] <BBB> sort of ... yes m0 is scratch there, but that's just because we've moved the instructions that do m0 a little higher up [15:55] <BBB> (since we don't have enough instructions) [15:55] <BBB> yes it's a little confusing but the comments try to follow the c code whereas the real code follows the flow of "I don't have enough registers" [15:56] <BBB> I'll think of how to do that cleverer [15:56] <BBB> this is indeed kinda weird [15:56] <ubitux> i guess it's the same for "; store t16-23 - keep t24-31 in registers for the final sumsub"? [15:56] <ubitux> (it's storing t17+) [15:57] <ubitux> t17-22 even [16:00] <BBB> hm no that comment is broken, it stores only t16-t19 and t23 [16:01] <BBB> and t16/t23 are done earlier [16:01] <BBB> so it only stores t17-19 [16:01] <ubitux> you could write a macro for the bunch of stores in 2nd pass [16:01] <ubitux> also, haven't you free a reg at that point for pw_512? [16:03] <ubitux> don't you have*, :english: [16:08] <ubitux> no more comment from me, except that maybe in this bunch of store we have the possibility to jungle with some dstq+N*strideq in the store macro to drop a few "add dstq, stride2q" [16:08] <ubitux> might not be possible without a neg strideq [16:08] <ubitux> but well, maybe not worth the effort [16:08] <ubitux> LGTM otherwise :p [16:08] <ubitux> i'd really like to have valgrind happy though :( [16:10] <BBB> so valgrind has been unhappy ever since the original first 16x16 landed? [16:11] <BBB> (i.e. only the full) [16:11] <ubitux> wasn't it pushed with all of them at once? [16:11] <ubitux> ah the sub might have appear after.. mmh i don't remember [16:13] <BBB> holy shit [16:13] <BBB> it just went from 2300 to 1450 cycles [16:13] <BBB> (by introducing the sub16x16 idct) [16:13] <BBB> sub landed later, I think you said this happened with the ful [16:13] <BBB> I'll add it on my plate to look at [16:14] <kierank> BBB: iirc you have an account [16:14] <BBB> kierank: oh it's still active? thanks, will check [16:19] <BBB> yay decoding tiem is under 8sec now [16:19] <ubitux> BBB: did you upload the sample btw? :P [16:19] <BBB> ohright [16:19] <BBB> doesn't ffmpeg have an upload somewhere? [16:19] <Compn> ftp://upload.ffmpeg.org/ [16:20] <BBB> incoming? [16:20] <BBB> or elsewhere? [16:20] <Compn> incoming yes [16:20] <Compn> sorry, the ftp stuff got all jumbled around i dont remember exactly :) [16:21] <BBB> ubitux: ftp://upload.ffmpeg.org/incoming/ped1080p.webm [16:21] <BBB> probably needs to be moved somewhere to be readable [16:21] <Compn> hope ubitux remembers http password to access it [16:21] <ubitux> i think i have an access [16:21] <ubitux> iirc it's a videolan one, dunno how those match [16:22] <Compn> yes [16:22] <Compn> upload.ffmpeg.org points to streams.videolan.org [16:22] <Compn> http://streams.videolan.org/incoming/ped1080p.webm [16:22] <Compn> is the final url of course [16:23] <ubitux> go it, thx [16:24] <ubitux> got* [16:26] <ubitux> 13:09:02 <@BBB> ubitux: sounds like a candidate for psrl{d,w,q}(pand(paddb(x, 1), 0xfe), 1) // will that really work? oO [16:30] <BBB> why not? [16:30] <BBB> sub16x16 uploaded to github [16:30] <ubitux> sounds like it's missing something [16:30] <BBB> will write a sub8x8 also, maybe it's faster [16:40] <ubitux> BBB: it doesn't work for unsigned [16:41] <ubitux> BBB: http://pastie.org/pastes/8600473/text [16:42] <ubitux> this is what you are suggesting, right? [16:42] <ubitux> well anyway, i'll figure out something [16:45] <BBB> oh it's signed [16:45] <BBB> I thought it was unsigned [16:45] <BBB> signed is much easier pavgb(x, zero) [16:46] <BBB> oh wait no that was unsigned [16:46] <BBB> duh [16:46] <BBB> ok [16:46] <BBB> hm [16:46] <BBB> let me think [16:46] <BBB> let's do pxor(pavgb((pxor(x, 0x80), zero), 0x80) [16:52] <ubitux> how does the pavgb() works really? (x+(x<0?-1:1))/2 with x being i8? [16:54] <nevcairiel> pavgb is unsigned, so simply (a+b+1)>>1 [16:56] <ubitux> if so, then the trick doesn't work [16:56] <ubitux> but i'll figure out something [17:08] <ubitux> psubb(pxor(psrlq(pand(x, 0xfe), 1), 0x40), 0x40) this works for a signed x>>1 [17:09] <ubitux> it's basically similar to psubb(pxor(psrlq(pand(x, 0xf8), 3), 0x10), 0x10) for x>>3 [17:10] <ubitux> now if i do psubb(pxor(psrlq(pand(paddb(x, 1), 0xfe), 1), 0x40), 0x40) i will have one mismatch for x=127 [17:10] <ubitux> (only case mismatching) [17:16] <BBB> why a mismatch? I mean, paddb and pand are sign-unaware [17:16] <BBB> so 127 -> 128 -> 128 [17:17] <BBB> psrlq makes it 64 [17:17] <ubitux> mismatch with i=127 (unsigned:127): -64 ` 64 [17:17] <BBB> oh because you pxor and then sub [17:20] <BBB> uint8_t u = x; [17:20] <BBB> u += 0x80; [17:20] <BBB> u = (u + 1) >> 1; [17:20] <BBB> u -= 0x40; [17:20] <BBB> return u; [17:20] <BBB> works for me [17:21] <ubitux> oh :) [17:21] <BBB> so psubb(pavgb(pxor(x, 0x80), zero), 0x40) [17:21] <ubitux> great! [17:21] <ubitux> thanks [17:24] <iive> there is no byte wise shift, so you clear the bit that would move into the other byte. aka the 0xfe constant. [17:25] <ubitux> iive: x is signed full range, and you need (x+1)>>1 [17:25] <ubitux> last suggestion from BBB works great :) [17:25] <iive> i'm talking about some of the previous code... [17:26] <ubitux> ok so it seems masks, filter2 and filter4 now work [17:26] <ubitux> debugging the 2 remaining monsters. [17:26] <BBB> lol [17:27] <ubitux> actually, filter6() seems to work too [17:27] <BBB> so filter14 is left? [17:27] <BBB> that one is kinda big [17:29] <ubitux> https://github.com/ubitux/FFmpeg/compare/vp9-lpf#diff-651121e4f3585d617f2de… [17:29] <ubitux> not "that much" [17:30] <ubitux> btw [17:30] <ubitux> up to how much args can i have? [17:31] <BBB> nice [17:31] <BBB> lf is 30.9% of runtime now [17:32] <BBB> args, you mean gprs? [17:32] <BBB> cglobal name, args, gprs, xmms, [mem,] name1, name2, ... [17:32] <BBB> gprs can be anything up to 7 on x86-32 [17:32] <BBB> and anything up to 15 on x86-64 [17:34] <BBB> I love that simd, very pretty [17:34] <BBB> better commented than most stuff I write ;) [17:34] <ubitux> i won't keep all comments [17:34] <ubitux> at least not in that state [17:34] <ubitux> that's just markers for debugging mostly [17:35] <BBB> sure [17:35] <BBB> http://pastebin.com/vEa2JZ9m [17:35] <BBB> profile [17:36] <ubitux> :) [17:36] <ubitux> did you get this with perf? [17:36] <BBB> I'm a maccie [17:36] <BBB> I use instruments [17:37] <ubitux> ah, ok [17:37] <ubitux> and yeah i meant gprs, ok for 15, too bad i wanted 16 (and actually already wrote that value :p) [17:38] <nevcairiel> its not too bad to load one or two args from the stack later on [17:38] <BBB> e.g. https://developer.apple.com/library/mac/documentation/developertools/concep… [17:39] <BBB> the values will be on stack anyway [17:39] <ubitux> lol at the lcd style on top [17:39] <BBB> anything after arg 6 is on stack (r15m) [17:41] <BBB> or argm [17:41] <BBB> e.g. cglobal name, 5, 5, 0, a,b,c,d,e,f,g [17:41] <BBB> gm is the stack location of that argument [17:42] <BBB> gmp is the same as <ptrsize>word gm [17:42] <BBB> e.g. qword gm [17:42] <ubitux> shouldn't it be 5,6,0 in this case? [17:42] <BBB> yeah, I guess, or even 6, 6, 0 [17:42] <ubitux> ah well [17:42] <ubitux> 6,7,0 i'd say [17:42] <ubitux> g being local [17:42] <ubitux> no? [17:44] <ubitux> well anyway, i'll see later [17:48] <BBB> 6,7 means load 6 arguments into registers, but reserve 7 registers [17:49] <BBB> so it just depends on what you want :) [17:49] <ubitux> i can't reserve 16 reg right? [17:51] <BBB> no [17:52] <BBB> just 15 [18:16] <Compn> starving the registers again? :) [18:23] <ubitux> Compn: no it's fine [18:23] <ubitux> i just wanted to do something nasty :p [18:47] <wm4> it seems yadif is still broken with some input pixel formats, like 10 bit [18:48] <nevcairiel> its intentional [18:48] <JEEB> to get prores users to pay for an update? :D [18:48] <nevcairiel> so that we can brand mark everyone that reports it as a interlace sympathizer [18:48] <JEEB> (10bit 4:2:2) [18:49] <wm4> why would an interlace sympathizer use yadif? [18:51] <ubitux> yay it almost works [18:52] <ubitux> it visually works, but not bitexact [18:52] <ubitux> it's gonna be really fun to find the problem now [18:53] <ubitux> wm4: it's broken because of the legal reverts [18:53] <ubitux> iirc [18:54] <nevcairiel> which can be reverted now :p [18:54] <ubitux> https://trac.ffmpeg.org/ticket/3250 [18:54] <nevcairiel> post on the ML said that the agreement came in,a nd the reverts can be reverted [18:55] <wm4> clearly a license change to make questionable external entities hapopy is much better than working code [19:21] <cone-115> ffmpeg.git 03Michael Niedermayer 07master:5e0c7eab2a9d: avformat/oggdec: dont read timestamps from EOS pages of ogm videos [19:22] <ubitux> http://b.pkh.me/vp9lpf-ref.png http://b.pkh.me/vp9lpf-out.png [19:22] <ubitux> who wants to play the N differences game with me? [19:23] <Daemon404> well first of all give us yuv [19:23] <Daemon404> not rgb :P [19:23] <Daemon404> cmp(1) that stuff! [19:23] Action: Daemon404 runs [19:25] <ubitux> :) [19:29] <JEEB> michaelni, thank you :) [19:33] Action: Daemon404 prods nevcairiel [19:34] <nevcairiel> wat [19:34] <Daemon404> do you accept problem samples for our mkv demuxer code over irc? [19:34] Action: Daemon404 runs [19:34] Action: Daemon404 is :lazy: [19:34] <Daemon404> s/our/your/ [19:34] <nevcairiel> sure [19:34] <Daemon404> notice'd you it [19:35] <Daemon404> plays fine with haalis mkv demuxer [19:35] <Daemon404> borks with lav [19:35] <Daemon404> seems ot be fine in vlc and mplayer [19:35] <nevcairiel> define borks [19:35] <nevcairiel> the timestamp mess? [19:35] <nevcairiel> hm maybe its just vfr [19:36] <Daemon404> it i vfr but thats never been an issue [19:36] <Daemon404> it only happens wth native (non-vfw) mpeg4-asp [19:36] <nevcairiel> so whats the issue with the file? :P [19:36] <Daemon404> im dlgn to remember [19:36] <Daemon404> <_< [19:36] <Daemon404> it was very obvious [19:36] <nevcairiel> despite the odd vfr timestamps it looks okish smooth and plays fine [19:36] <Daemon404> oh right [19:37] <Daemon404> it looked horribly not-smooth for me and JEEB [19:37] <Daemon404> i thought it ws the encode itelf, but it is fine with haali [19:38] <JEEB> I'll retest with current git after I finish this thing [19:38] <Daemon404> yeah its very obvious on e.g. the scrolling credits [19:39] <nevcairiel> bet i can blame avformats timestamp code somehow [19:39] <Daemon404> lol [19:46] <nevcairiel> hm ok, every frame in the mkv has a proper timestamp and the demuxer outputs it too, most interestingly is that every frame has a duration of 41ms though :p [19:47] <wm4> where's the sample? [19:47] <Daemon404> wm4, noticed it to you [19:47] <wm4> thanks [19:48] <Daemon404> nevcairiel, it seems to be dropping frames maybe in the 30fps section [19:48] <Daemon404> but the 24 fps sections are jerky too [19:49] <nevcairiel> appears that avformat doesnt mess with the timestamps [19:50] <nevcairiel> i get them out of avformat the same way the mkv reader has them [19:50] <Daemon404> do they appear correct? [19:50] <nevcairiel> well they are re-ordered, so its always hard to judge [19:51] <Daemon404> ah [19:52] <nevcairiel> i wonder [19:52] <nevcairiel> found the difference [19:53] <nevcairiel> i bet its those stupid peoples fault that mux vfw mpeg4 [19:53] <Daemon404> ? [19:53] <Daemon404> as in hacks for vfw mpeg4 broke my native stream? [19:53] <nevcairiel> kinda [19:53] <nevcairiel> well [19:54] <nevcairiel> vfw containers use dts timestamps [19:54] <nevcairiel> and native uses pts [19:54] <nevcairiel> so decoders usually can deal with both and need to somehow decide which to use [19:54] <Daemon404> ... ew [19:54] <nevcairiel> but since mpeg4 in avi is really quite common, thats probably the fallback [19:54] <Daemon404> [18:54] <+nevcairiel> vfw containers use dts timestamps <-- thats pretty gross imo [19:55] <nevcairiel> avi has no timestamps as such, so they cant have re-ordereing [19:55] <JEEB> and yeah, most people muxed mpeg-4 part 2 from avi [19:55] <nevcairiel> just a frame count and a timebase [19:55] <wm4> and there's no way to get correct timestamps with lavf [19:55] <Daemon404> well yes [19:55] <wm4> err, lavc [19:55] <JEEB> and mkvmerge just muxed everything as-is from avi [19:55] <wm4> because lavc assumes mpeg style timestamps [19:55] <JEEB> thus almost everyone encoding mpeg-4 part 2 ended up with avi-mkv [19:55] <JEEB> ( [19:55] <nevcairiel> but no worries [19:55] <nevcairiel> i can fix it [19:56] <Daemon404> ;) [19:56] <Daemon404> \o/ [19:56] <JEEB> nais [19:56] <nevcairiel> mpeg4asp in mp4 for example is always native [19:56] <nevcairiel> at least as far as i've seen [19:56] <Daemon404> mpeg4 in mkv is easily distiguishable from vfw-mpeg in mkv [19:56] <Daemon404> seems liek a 2 line fix [19:56] <nevcairiel> yeah [19:56] <nevcairiel> just need to figure out if that information is actually available where i need it [19:57] <Daemon404> lavf should expose the matroska codec string no? [19:57] <nevcairiel> not sure where it would be in [19:57] <wm4> hehe, lavc probably has the same problem as nevcairiel [19:58] <wm4> or at least somewhat similar [19:58] <nevcairiel> in the demuxer i have a "ms_compat" flag which is set for those vfw streams [19:58] <nevcairiel> should add a private option to read it [19:59] <wm4> e.g. if I use the lavf demuxer with my player, vfw muxed mkvs will have the wrong timestamp for the first frame, because the avi-style timestamps clash with lavc assuming mpeg style timestamps [19:59] <nevcairiel> if my code knows that its getting dts timestamps it will simply handle them outside of lavc, so thats not a problem [20:00] <nevcairiel> but it needs that information, which is the problem here :D [20:00] <Daemon404> most of this seems the rsult if lavf insanity [20:00] <Daemon404> afaik [20:00] <Daemon404> s/if/of/ [20:00] <nevcairiel> not really [20:01] <Daemon404> the dts handling in lavf has always been insane (in that its not really dts sometimes) [20:01] <nevcairiel> its not lavfs fault that there are files with the same codec but different muxing modes which result in completely different timestamps [20:01] <Daemon404> and shouldnt that be handled in the matroska demuxer [20:02] <Daemon404> transparently [20:02] <nevcairiel> whats the demuxer supposed to do? know about all codecs and generate pts from the dts encoded in the file? [20:02] <wm4> Daemon404: the lavf mkv demuxer does return dts for vfw muxed shit, but lavc assumes it's mpeg style dts, which means the timestamps get "delayed" - which apparently means that mpeg streams starting at 0 have negative dts for the first frames [20:02] <wm4> Daemon404: at least that was my problem wrt. difference between vfw muxed and native [20:02] <Daemon404> nevcairiel, to an extent that its possible [20:03] <Daemon404> vfw stuff is all sub-ids [20:03] <Daemon404> with a common prefix [20:03] <nevcairiel> but it cant generate pts [20:03] <nevcairiel> it also shouldnt really [20:03] <nevcairiel> it should output whats in the file [20:03] <nevcairiel> and for mkv that usually works ok [20:03] <Daemon404> yes but it cant have pts anyway [20:04] <Daemon404> or expose some sort of dts == pts flag [20:04] <Daemon404> wm4, the mpeg delay stuff has bit me a few times [20:05] <wm4> so to handle avi timestamps correctly, I had to special case avi and vfw-mkv, and enable the old mplayer timestamp sorting code [20:11] <nevcairiel> ah now the sample looks really smooth [20:12] <nevcairiel> now i just need to find a vfw sample to make sure i didnt break those [20:12] <JEEB> ugh, wanted to link you one but then remembered that it was highly VFR [20:12] <JEEB> as in [20:12] <JEEB> very loooong pictures [20:12] <nevcairiel> i found one [20:13] <wm4> JEEB: long pictures? [20:13] <JEEB> wm4, http://www.cccp-project.net/beta/test_files/cbed_lavc_vfr.mkv [20:14] <wm4> oh I see [20:37] <ubitux> http://pastie.org/8601146 [20:37] <ubitux> b hello_gdb [20:37] <ubitux> profit [20:37] <ubitux> (yes, fuck gdb scripting) [20:37] <nevcairiel> lol [20:38] <nevcairiel> conditional break points in gdb are hard, i take it? [20:38] <ubitux> i didn't even try [20:39] <nevcairiel> do I just not see them, or did the yadif patches not reach the ML at all? [20:40] Action: beastd doesn't see them too [20:41] <Daemon404> nevcairiel, yeah i didnt either [20:41] <Daemon404> they might be in moderation [20:41] <nevcairiel> its a bit weird, since he said only one patch didnt show up for him, which is why i was confused [20:41] <nevcairiel> but i guess so [20:41] <Daemon404> i saw his post on libav's and went to look [20:41] <Daemon404> and i saw nothing [20:42] <nevcairiel> I can see that the fourth patch with the actual license change [20:42] <nevcairiel> has not made it to the list yet. [20:42] <nevcairiel> that kinda implies the first 3 made it to the list [20:42] <nevcairiel> but maybe i read it wrong =p [20:43] <nevcairiel> also, i should really stop being lazy and publish a new release already [22:06] <mark4o> michaelni: re oggparseopus issue: [22:06] <mark4o> try: ffmpeg -i http://opus-codec.org/examples/samples/music_orig.wav -b:a 34k -vbr off -frame_duration 60 out.opus [22:06] <mark4o> ffplay out.opus [22:06] <mark4o> click middle of window to seek to middle [22:25] <michaelni> does this need a specific version of libopus ? [22:25] <Daemon404> i must ask [22:25] <Daemon404> (unrelated) [22:26] <mark4o> michaelni: I am using 1.1 [22:26] <Daemon404> why do we set the frame duration differently than libopus [22:26] <Daemon404> by default [22:26] <Daemon404> seems silly. [22:26] <Daemon404> s/libopus/opus-tools/ [22:26] <michaelni> mark4o, i think its 1.0.1 here [22:26] <michaelni> ill update it [22:26] <mark4o> Daemon404: that was fixed to 20ms for ffmpeg. I think it is still 10ms in libav. [22:27] <Daemon404> oh so it was fixed [22:27] <Daemon404> dice [22:27] <Daemon404> er nice [22:28] <mark4o> Daemon404: 74906d3727ec3bd9b7b28dfa7a98ff6e8cf8b6d7 [22:30] <mark4o> michaelni: yeah I think there were some CBR issues in older libopus versions, where it didn't output the exact rate that was requested [22:31] <bais> there is a web gui for ffmpeg ? [22:31] <michaelni> mark4o, i confirm its reproduceable with 1.1 [22:32] <Daemon404> g 30 [22:32] <mark4o> michaelni: ok cool [22:46] <cone-115> ffmpeg.git 03Mark Harris 07master:262451878bab: avformat/oggparseopus: fix segmented timestamps [00:00] --- Sun Jan 5 2014

1 0

ffmpeg.log.20140104
by burek 05 Jan '14

05 Jan '14

[00:01] <relaxed> if that fails and you just to concat the streams, maybe mkvmerge would be a better tool. [00:01] <relaxed> *just want to* [12:58] <vvu|Log> can ffmpeg get video from DVD EZMaker 7 dongle ? as i know Linux has kernel support for it and it should see it as v4l device *think so* [13:32] <DrSlony> Hi, I need to convert videos to divx/xvid because my dvd player only handles these old formats, whats the most compatible ffmpeg way to convert to that? [13:34] <Katharsis> which of the software could you recommend me for screencast (not CLI based) recording? [13:36] <Katharsis> we've used ffmpeg for last days [13:36] <Katharsis> ffmpeg -f alsa -i pulse -f x11grab -r 30 -s 1366x768 -i :0.0 -vcodec libx264 -r 25 -preset ultrafast -threads 4 -y -sameq file_name.mp4 [13:38] <Katharsis> we can make and play mp4 file on Linux without any problems [13:40] <Katharsis> Windows software play the movie without screen (voice only), the same problem is with Adobe Software [13:42] <Katharsis> to fix it, we had to conver this ffmpeg mp4 movie into mp4 file type with FreeMake Video Converter [13:42] <DrSlony> Katharsis recordmydesktop kazam istanbul vokoscreen vlc eidete tibesti byzanz [13:42] <Katharsis> then, we realized that the sound isn't synch with video [13:43] <DrSlony> Katharsis ive also had issues with windows people not being able to play mp4 made with ffmpeg in linux, but they upgraded their vlc or mplayer and it worked [13:43] <Katharsis> DrSlony: which of them is universal for linux/windows and userfriendly? [13:43] <DrSlony> most gui screen recording programs use ffmpeg anyway, so that probably wont work around the issue [13:44] <DrSlony> Katharsis check each one's website for screenshots, see what you like [14:36] <konflict> Hi, Im currently trying to solve -timecode sync issue. Im piping two SDI streams from BlackMagic Decklink into ffmpeg to capture two DNXHD mxf files. I need to get identical timecodes on both clips based on either system time or some external source for later "frame precise" editing. So far I wasnt able to find any solution. Could anyone help? [15:13] <Katharsis> DrSlony: we can't upgrade Adobe on Windows, it seems that v5 doesn't support libx264 + MP3 inside of MP4 file [15:13] <Katharsis> any solution for this problem? [15:24] <Trashlord> is there any option I can pass to possibly speed up the conversion time? I've spent the last 7 hours converting a 2 hour video file from a bunch of VOBs to an mp4 file [15:27] <Trashlord> it's converting at 7.1 fps [16:06] <Katharsis> is it possible to resize 1360x768 to 1280x720 without lost? [16:06] <Katharsis> (mp4 movie) [16:12] <Katharsis> i'm using ffmpeg -i movie_a.mp4 -s 1280x720 -acodec copy -sameq -sws_flags lanczos -f mpegts -vcodec libx264 -threads 0 movie_b.mpg for that [16:12] <Katharsis> but i have a little lost (blur on strings) [16:16] <saste> Katharsis, no [17:19] <ac_slater> hey guys, I'm pretty new the world of video encoding/decoding/containers. I'm trying to find the best container/format for nice balance of file size and enc/decode complexity. I'll be creating these containers for use on an embedded system, hence the avoidance of h.264. Any suggestions? [17:19] <ac_slater> (also, I dont care about audio right now) [17:20] <ac_slater> (and i know h.264 is awesome and all, I just do not have access to hardware encoder) [17:21] <klaxa> try xvid/mpeg4 part 2 [17:21] <klaxa> also, that's a codec, not a container [17:21] <ac_slater> klaxa: right, I understand that much ;) [17:21] <klaxa> container would be matroska or avi or mp4 [17:21] <ac_slater> is there a comparison matrix for popular containers? [17:22] <klaxa> http://en.wikipedia.org/wiki/Comparison_of_container_formats [17:22] <ac_slater> haha [17:22] <ac_slater> klaxa: apologies mate. [17:23] <klaxa> huh? why? [17:23] <ac_slater> didn't do my googling before bothering people here [17:24] <ac_slater> (other channels I'm on HATE that) [17:24] <klaxa> well usually yes [17:25] <ac_slater> aside from xvid / mpeg4pt2, any other suggestions? [17:25] <klaxa> if xvid is still too complex maybe mpeg2 [17:25] <klaxa> if that is too complex i hope the databus is fast enough for rawvideo [17:26] <klaxa> http://en.wikipedia.org/wiki/Comparison_of_video_codecs here's another hopefully helpful list/comparison [17:26] <ac_slater> klaxa: in my case, I *can* do raw video, but there might be mixins with streaming protocols later - like RTSP which have certain codec restrictions [17:26] <ac_slater> yea I just saw that link ... sadly a lot of stuff on there is "unknown" [17:27] <ac_slater> oh but that's just software support - nevermind [17:27] <klaxa> well what are the specs of the device? [17:27] <ac_slater> 1Ghz dual-core ARM cortex A9 SoC [17:27] <klaxa> doing video decoding on embedded devices without hardware decoding is pretty... eh i guess [17:27] <klaxa> oh that should be able to handle xvid just fine [17:28] <ac_slater> klaxa: yea I think so as well, it just sucks bad at encoding h.264 [17:28] <klaxa> or mpeg 4 part 2 [17:28] <klaxa> i think xvid is even mpeg 4 part 2 compliant [17:28] <klaxa> let me check [17:29] <klaxa> >Xvid (formerly "XviD") is a video codec library following the MPEG-4 standard, specifically MPEG-4 Part 2 Advanced Simple Profile (ASP). [17:29] <klaxa> at least wikipedia says so [17:29] <ac_slater> badass. [17:33] <ac_slater> klaxa: by any chance, do you know much about RTSP ? [17:33] <klaxa> actually not [17:33] <ac_slater> ah thanks anyways. It seems to be a difficult topic to get some solid details on [17:34] <klaxa> i tried to get into video streaming with ffserver, but i didn't get how it worked so i kinda gave up and settled for setting myself a goal of writing a proper streaming server for matroska over http [17:34] <klaxa> no idea when that will take off though [17:35] <ac_slater> klaxa: that's pretty awesome. I'd be interested in taking a look at that if it's opensource - mostly cause I have no idea what streamers have to manage in terms of packetization. Sadly, I have to keep my protocols standardized. [17:35] <klaxa> haven't seen any streaming software that supports matroska for streaming so far, i don't even know why, it's designed for live-streaming and the most sophisticated container format available [17:36] <klaxa> well http is pretty standardized [17:36] <klaxa> i have some other projects still open, i want to finish those first, yesterday i finally managed to compile libmatroska2 [17:37] <klaxa> i'll probably go through the source and try to jot down *some* sort of documentation [17:37] <klaxa> there is neither documented how to compile it, nor how to use it, lol [17:37] <ac_slater> klaxa: typical ;) [17:37] <klaxa> i've heard the same thing about libmatroska [17:38] <klaxa> "Well... if you need documentation, maybe read the vlc source..." [17:38] <ac_slater> I've streamed mkv containers in the past though. Maybe they were being transcoded or something [17:38] <klaxa> how did you do that? [17:38] <ac_slater> yea, I don't like that indirection either - producing documentation isnt too difficult [17:39] <klaxa> but it's not as much fun as coding the thing in the first place :) [17:39] <ac_slater> klaxa: recently, I used `rygel` to broadcast some mkv files via DLNA. [17:39] <klaxa> i can relate, i'm not fond of writing documentation, but reading documentation is greatly appreciated [17:39] <ac_slater> (not sure what implications lay in the DLNA/UPnP method though) [17:40] <ac_slater> writing docs help me produce usable interfaces ;) [17:40] <klaxa> hmm yeah i haven't dealt with DLNA either [17:40] <klaxa> but i don't think it's actually the sort of live-streaming i want to achieve [17:40] <ac_slater> klaxa: but http is much more convenient anyways [17:45] <ac_slater> man, stepping into the world of video enc/dec/containers is overwhelming. [17:47] <klaxa> well at least you are already starting with ffmpeg [17:47] <klaxa> that's good because it's the de-facto open-source standard [17:47] <klaxa> most open-source video projects have ffmpeg or its libraries as dependencies [17:48] <ac_slater> right, I've been using it for years as a converter,etc ... but never as an SDK [17:48] <klaxa> ah, well yeah i haven't looked into the libraries either really... [17:48] <klaxa> i'd much rather have some nice python bindings than writing everything in C :x [17:49] <ac_slater> klaxa: haha yea I second that. It would make prototyping faster anyways [18:12] <lnostdal> so, "compilation; it just takes a few minutes" says the wiki .. but then why aren't there any up to date, official ppa for ubuntu (yes; for the masses and the noobs) out there which is up to date in the same way the ppa for e.g. vlc is up to date? [18:13] <lnostdal> instead, the masses end up with libav..... unfortunate [18:18] <klaxa> good question, i just compile from source :x [18:18] <klaxa> or get a static build [18:19] <klaxa> which are up to date [18:21] <lnostdal> yeah, the static build (well one of them; i did not test both of them) doesn't include pulseaudio (or alsa; i'm not sure) support .. x11grab support is also missing [18:22] <lnostdal> that's the two things i know are missing; perhaps more things are missing too [18:22] <klaxa> true, i think x11grab is not possible with static binaries [18:22] <Mavrik> yeah, because they're system dependant [18:22] <klaxa> well feel free to set it up yourself if you feel that it's missing :I [18:22] <Mavrik> also, making a PPA for ffmpeg is PITA [18:22] <klaxa> that's what linux lives off after all [18:22] <Mavrik> due to wierd Debian rules for building debian packages [18:25] <JEEB> I think in general there's less weird rules and more that you can't make a package that provides the basic libraries that has a different API/ABI. So I think you're left with 1.xx ffmpeg if I recall correctly. Might have been pushed further because the libav 9 migration has finally finished. They're now doing libav 10 migration to be ready for its release so you should then be able to package newest ffmpeg as well [18:25] <JEEB> it's kind of sad to look at the migration status when someone is trying to update X amount of downstream projects by himself [18:25] <JEEB> something you don't think about at all when you just use the ffmpeg command line app [18:26] <lnostdal> yeah, i can compile this from source myself .. i've done so many times .. or i can go grab up to date .deb from some semi(?) dodgy ppa (e.g. https://launchpad.net/~samrog131/+archive/ppa ) .. *shrug* [18:36] <lnostdal> even with ABI / API problems - some official ppa for ffmpeg that states that "this will or might cause problems" (because of API / ABI changes) would be great .. it doesn't have to be "perfect"; something official would already be better than the alternative ways (unofficial ppa or from source) .. perhaps the packages and executables could have different names .. anyway, just rambling; i'll go back to compiling this now .. :P [18:39] <beastd> lnostdal: I agree that we would be better off with some .deb package [18:39] <beastd> but thing are never easy :8 [18:39] <beastd> :( [18:40] <beastd> a .deb delivering static only binaries (and libs maybe) should be OK [18:41] <JEEB> libs can kind of lead to unforeseen consequences unless you really work it out [18:41] <JEEB> ffmpeg/-probe/-play can be built with static libav* of course [18:41] <JEEB> which could be packaged [18:41] <beastd> anything that delivers dynamic libs, will cause problems [18:42] <JEEB> even static [18:42] <beastd> and even worse the problems will most likely end up in the wrong mailbox :((( [18:42] <beastd> *the problem reports [18:43] <JEEB> anyways, most people just want the tools [18:43] <JEEB> which I guess isn't that hard to deliver [18:43] <JEEB> even if they would go against official debian policies (static linking) [18:43] <beastd> JEEB: What scnenario do you have in mind where static libs would cause problems? (just curious) [18:43] <JEEB> just means it wouldn't get into mainline [18:43] <JEEB> beastd, if they are put into the search path basically [18:43] <JEEB> so you have libav's shared libs [18:43] <JEEB> and ffmpeg's static ones [18:44] <JEEB> linker's search path of course is what I mean [18:44] <JEEB> there are ways around this of course [18:46] <beastd> JEEB: that is only a problem when building stuff and you do not want to use current FFmpeg (e.g. a project that only works with Libav or older versions of FFmpeg), right? [18:46] <JEEB> well, it's always a problem depending on which the linker happens to pick [18:46] <JEEB> and which you wanted [18:47] <JEEB> just means that if you want to package libraries you'll probably want to do it in some way that won't make the distro's stuff overlap [18:47] <JEEB> and headers [18:47] <beastd> yeah, probably [22:12] <CoJaBo> I have a bunch of 3GP files where the "moov atom" is missing; is there any hope of recovery whatsoever, or should I just delete them? [22:12] <haarp> when ffmpeg cant know the duration of any of the input streams (http://pastebin.com/Li3dHQk8 -1-), the stats shown during operation are completely wrong. the time count up way faster than realtime, even though it can only capture in realtime. when i add a duration to the audio (-2-), the stats are correct again. whats going on? [22:24] <konflict> Hi, Im currently trying to solve -timecode sync issue. Im piping two SDI streams from BlackMagic Decklink into ffmpeg to capture two DNXHD mxf files. I need to get identical timecodes on both clips based on either system time or some external source for later "frame precise" editing. So far I wasnt able to find any solution. Could anyone help? [23:52] <BeWilled> Is there a parameter to tell ffmpeg to overwrite everything and don't ask? [23:53] <haarp> -y [23:53] <relaxed> BeWilled: ... -y output_name [00:00] --- Sun Jan 5 2014

1 0

ffmpeg-devel.log.20140103
by burek 04 Jan '14

04 Jan '14

[00:02] <cone-48> ffmpeg.git 03Dale Curtis 07master:4feca2214a0b: h264: Clear ERContext.cur_pic when unref'ing current picture. [03:55] <cone-742> ffmpeg.git 03Michael Niedermayer 07master:8a1714ad85dd: ffmpeg: do not fail when options are routed to libavformat and libavcodec and only one can be used [03:56] <Daemon404> uhhhh shouldnt that at least issue a warning? [05:11] <cone-742> ffmpeg.git 03James Almer 07master:2c759d7018fa: matroskadec: Export the MuxingApp element value as metadata [05:36] <michaelni> Daemon404, in the use case that lead to it, -frame_size was causing it and it was specified for the demuxer/decoder side. The problem occured because theres a frame_size for encoders too [05:37] <michaelni> but noone specified it for encoders so i think it shouldn need a warning [05:37] Action: michaelni goes to bed, will read log tomorrow [13:27] <saste> anyone has hints about #3265? [13:27] <saste> michaelni, what about the fm check in the code? [13:27] <saste> audio-only seems to work fine [13:33] <michaelni> saste, seems to be checking for libavformat/ffm* PACKET_ID [13:34] <michaelni> i dunno that part of the code though so ... [13:39] <saste> i wonder why it only seems to affect libvpx... [14:05] <cone-17> ffmpeg.git 03Peter Ross 07master:0588acaffaf6: avformat/bink: seek to first frame [14:06] <cone-17> ffmpeg.git 03Anssi Hannula 07master:857841c1b63b: avformat/http: always allow no-op seek [14:06] <cone-17> ffmpeg.git 03Anssi Hannula 07master:9371d70bad2e: avformat/hls: decouple playlists from variants [14:06] <cone-17> ffmpeg.git 03Anssi Hannula 07master:6275d93d1f19: MAINTAINERS: add myself for spdif* and hls.c [14:39] <cone-17> ffmpeg.git 03Peter Ross 07master:57cfcbf34776: avformat/bink: display audio track ids [15:34] <j-b> "HLS provides MPEG TS timestamps via ID3 tags in the beginning of each segment of elementary audio streams." this is a joke, right? [15:34] <av500> lol [15:34] <JEEB> lol [15:37] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:0389f9abe9b1: avcodec/avcodec: document that some video decoders do not support linesizes changing between frames. [15:41] <Compn> id3 ! of course! [15:41] <Compn> the best metadata format ever! [16:07] <Anssi> j-b: I wished it was :/ [16:07] <j-b> Anssi: come on... [16:31] <kierank_> j-b: wtf [16:33] <Anssi> (here's the spec paragraph: http://tools.ietf.org/html/draft-pantos-http-live-streaming-12#page-23 ) [16:35] <kierank> Anssi: any idea why? [16:36] <Anssi> kierank: not really [16:36] <kierank> why the hell do they need to do that [16:36] <wm4> ubitux: anything new on the gif encoding front? [16:38] <Anssi> kierank: well, they had a raw audio stream (no timestamps) and wanted to sync it with an actual MPEG-TS, so they needed timestamps in the raw stream [16:38] <kierank> then why can't they add timestamps the normal way [16:42] <Anssi> kierank: you mean not use a raw format? not sure... maybe they wanted to avoid MPEG-TS overhead but didn't want to use a non-MPEG format so they went with raw + id3 timestamps [16:42] <kierank> usually you'd just add a PES header [16:42] <kierank> which isn't that large [16:43] <kierank> unless they want to hack round the padding requirements at the end [16:44] <nevcairiel> the timestamps dont apply to the mpeg-ts, but a separate raw audio stream, cant put timestamps in there [16:45] <Anssi> kierank: right, no idea why they didn't go with that [16:46] <kierank> nevcairiel: eugh and they can't just multplex that with the ts? [16:46] <nevcairiel> who knows why they decided what they did [16:46] <Anssi> kierank: they want to save bandwidth so that client can decide what streams to stream [16:47] <kierank> i mean they can have a separate TS? [16:48] <Anssi> kierank: hm, didn't get what you meant [16:48] <kierank> instead of providing a separate raw audio track with timestamps, just provided another TS file [16:49] <nevcairiel> that reminds me that i need to implement hls streaming into the media server, i bet that'll be much fun :( [16:51] <Anssi> kierank: with just the audio or video+audio? if you mean the former, I guess to avoid TS overhead (I'm not sure how much that is, though), if the latter, since there may already be X different video options and then Y audio options (and Z subtitle options) [16:51] <kierank> former [16:51] <Anssi> the spec allows TS for that though [16:51] <Anssi> but also raw+id3 [17:06] <cone-17> ffmpeg.git 03Stefano Sabatini 07master:d52dd2430bcd: doc/ffserver: mention how to access streams through RTSP [17:06] <cone-17> ffmpeg.git 03Stefano Sabatini 07master:d392619a465e: doc/faq: remove "-profile option fails when encoding H.264 video with AAC audio" entry [17:22] <ubitux> wm4: since last i work on it? no [17:23] <ubitux> not that i know of at least [17:23] <wm4> didn't j-b ask about gif encoding? [17:23] <wm4> anyway, it seems users kind of expect being able to encode color gifs [17:23] <wm4> but that apparently requires pal8 input [17:23] <ubitux> huh? oO [17:24] <wm4> there's also RGB8, but that's pretty useless [17:24] <ubitux> well at least that's 10x smaller than pan [17:24] <ubitux> pal* [17:26] <wm4> also, I have a case where seeking in a mpeg stream doesn't jump to a key frame [17:26] <wm4> is this expected behavior or should I report a bug? [17:26] <nevcairiel> mpeg streams dont seek to keyframes in general, the decoder is expected to pick up the pieces [17:28] <wm4> so it's broken by design on the demuxer level? [17:38] <nevcairiel> on containers who dont have a keyframe flag on packets, sure, the demuxer simply doesnt know whats in the packets [17:39] <wm4> doesn't libavformat use all these codec parsers (and sometimes even full decoders) exactly for that reason? [17:39] <nevcairiel> not for seeking [17:40] Action: Daemon404 hugs his pre-indexed ffms2 [17:40] <nevcairiel> i suppose it could be rigged up to drop packets until it finds a keyframe, but for complex codecs that would cause much logic duplication [17:40] <nevcairiel> take h264 with intra-refresh, whats a keyframe? [17:41] <wm4> there's some functionality in libavcodec to drop "broken" frames [17:41] <nevcairiel> i once considered all these cases to get more accurate and faster seeks, but a generic solution is really hard [17:42] <wm4> that gets frames with missing reference frames, but AFAIK also frames that simply are corrupted [17:42] <nevcairiel> so i spent my time making sure decoders dont output broken frames after any random access into the stream [17:42] <nevcairiel> its not perfect of course, but much better [18:19] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:9c5260e73a7a: avutil/mathematics: fix 2 typos in the doxy [20:13] <ubitux> ok, i can now somehow dicern the original video after the optimized loop filter [20:13] <ubitux> so it's "starting to work but still broken as shit" [20:14] <ubitux> assuming there is no major design issue, this is going to bring about 25 fps (175 -> 200) with only the vertical one [20:17] <ubitux> (thanks the threading ofc) [21:34] <cone-17> ffmpeg.git 03Michael Niedermayer 07master:950fb8acb42f: avcodec/mpegvideo: fix ac/dc_val and coded_block table sizes [21:38] <ubitux> oh... [21:38] <ubitux> now how am i supposed to make add an unsigned byte with a signed byte, with unsigned saturation? [21:41] <ubitux> basically, clip_u8(u8 ± i8) [21:44] <nevcairiel> that seems like a weird task [21:45] <ubitux> tell that to the vp9 authors :( [21:45] <nevcairiel> maybe there are restrictions that the signed byte will never be negative? [21:46] <ubitux> no :P [21:46] <nevcairiel> result of the calculation can be negative and needs to clip to 0? [21:46] <ubitux> 0-255 [21:46] <ubitux> really, it's just clip_u8(u8 ± i8) [21:47] <ubitux> (i need to support both add and sub) [21:47] <nevcairiel> the question is, can u8 be so small that adding a negative i8 would make it negative? [21:47] <ubitux> yes it can be full unsigned byte range [21:47] <nevcairiel> well, 16-bit intermediates then, not sure how else to solve it [21:47] <nevcairiel> but maybe there are some magic tricks [21:47] <ubitux> eeeh :( [22:05] <michaelni> ubitux, split the signed bytes in 2 registers one that holds the positive ones and one that holds the negative ones and then just do 2 unsigned saturating add/sub [22:07] <michaelni> not sure how best to do the split but you have add/sub and min/max available to build it out of [22:10] <JEEB> hmm, anyone interested at looking at https://trac.ffmpeg.org/ticket/2739 when they have time? [22:10] <wm4> oh that again [22:10] <nevcairiel> its ogm, of course not [22:11] <JEEB> wm4, decided to poke the channel for the first time in X months :D [22:20] <JEEB> nevcairiel, and yes -- I wish those files didn't exist. Yet they do :< [22:25] <michaelni> JEEB, where is the file ? [22:26] <JEEB> I uploaded it onto the sample FTP [22:26] <JEEB> although I do IIRC have it on a random server if that's gone away by now [22:26] <michaelni> i think i found it [22:29] <ubitux> michaelni: mmh ok [22:30] <Daemon404> ogm you say? [22:30] <Daemon404> http://chromashift.org/img/%5bAXP%5d-Intro.ogm as usual. [22:36] <ubitux> 3d, matrix, angels, techno font, fire, gothics [22:36] <ubitux> what's missing? :)) [22:36] <llogan> naturally it was prepended to some anime [22:37] <llogan> i imagine he lives in Frankfurt [22:38] <JEEB> oh yes, the classic intros >_> [22:58] <Daemon404> llogan: why indeed. [22:58] <Daemon404> indeed it was. [00:00] --- Sat Jan 4 2014

1 0

ffmpeg.log.20140103
by burek 04 Jan '14

04 Jan '14

[00:16] <matham> Is there a way to set the avframe.data pointers to my own buffers, can I simply just change it's values to point at my buffer while making sure that linesize is correct? [00:49] <msmithng> any docs defining applicable option settings for ffserver.conf entries? [00:50] <llogan> Zeranoe: there's tools/normalize.py but i've never looked at it [00:50] <msmithng> for instance # Ratecontrol buffer size > VideoBufferSize 40 [00:51] <msmithng> what is the double defining? [00:51] <llogan> FunkyELF: using m4a with ffmpeg may use an "ipod muxer" IIRC [00:51] <llogan> *m4v [02:33] <relaxed> average: For x11grab you'll need to build ffmpeg yourself. [02:49] <Lachezar> Can I ask questions about avcodec here, or do I get K-B-ed for that? [04:06] <mark4o> FunkyELF: https://trac.handbrake.fr/wiki/Containers#MP4vsM4v [11:26] <Blacksheep> hi everyone. I have a problem with ffserv. I'm strimmig my webcam via VLC (http) and I want my ffserv to get this 'remote' feed. Is it possible ? [12:08] <brontosaurusrex> how would i check if file is audio (or contains an audio track) with ffmpeg, I just need "true" or "false" ? [12:35] <Blacksheep> Could someone tell me how to configure ffserver so when i stream my webcam with ffmpeg to it, i can see the resoult on my webpage using html5 video tag? [12:36] <Blacksheep> i can not see te resoult of stream neither by ffplay or vlc. [14:08] <Waraqa> Hi, Is there a way to limit mapping to a specific duration? [14:14] <Waraqa> I want to replace only part of the original audio stream with external audio in a video input [14:18] <Waraqa> I need a way to limit the effect of using map option [14:22] <Spideru> MOrning [14:51] <Spideru> I would to stream sound from my alsa audio card to ffserver, using ogg format. Then, on ffserver side, I would to record audio stream to a file and send the stream to ffplay or another multimedia remote player. How I should configure ffserver to receive ogg stream and to record/redirect it? How can I send ogg stream to ffserver? Thabk you [14:51] <Spideru> *Thank you [14:52] <Spideru> Is possibile to use ffserver and the source of the stream separated by public internet connection? [16:55] <Waraqa> I want to replace only part of the original audio stream with external audio in a video input, Is there a way to limit mapping to a specific duration? [17:00] <viric> Waraqa: amix ? [17:01] <viric> maybe not [17:02] <Waraqa> viric: I will look for that one, thanks [18:28] <pyBlob> "-pix_fmt yuv444p will return the components in seperate blocks, while "-pix_fmt rgb24" will return the components in interleaved format [18:28] <pyBlob> ... is there also a seperate format for rgb? [18:30] <JEEB> pyBlob, you mean a planar one, use -pix_fmts and grep for p and see the ones that contain r, g and b [18:30] <JEEB> or feel free to create a nice grep line for all of that at once :) [18:30] <pyBlob> I'm currently on windows :( [18:31] <pyBlob> but I'll have a look [18:31] <JEEB> well, find in that case but generally the same idea [18:31] <JEEB> (except find doesn't like regular expressions, but if you just need the stuff with a "p" it shouldn't be too hard) [18:31] <JEEB> and yeah, you probably want a format that has both I and O in the listing [18:32] <JEEB> with a quick look I noticed a few alternatives, so feel free to go nuts [18:32] <pyBlob> btw ... what does the 444/420/... stuff mean? [18:33] <JEEB> that's chroma subsampling [18:33] <JEEB> 4:4:4 , 4:2:0, 4:0:0, 4:2:2 etc. [18:33] <JEEB> in no specific order [18:34] <JEEB> 4:4:4 is full chroma (aka no subsampling), 4:2:2 is two samples for one real sample in width, and 4:2:0 is four samples (2x2 area) for one real sample [18:37] <pyBlob> so gbrp should be the format I'm searching: "IO... gbrp 3 24" [18:37] <JEEB> one of them yes, most probably [18:38] <pyBlob> strange ordering ... but ok :S [18:38] <JEEB> probably already used by something [19:41] <FunkyELF> is there a command line tool to read tags from mp4 files? I see mp4tags can tag them but don't see that it can read them back [21:25] <Bombo> what can i do to avoid these: 'Invalid pixel aspect ratio 519/520, limit is 255/255 reducing' i have 'Video: h264 (High), yuv420p, 1920x1040, SAR 1:1 DAR 24:13' and want to encode+resize to xvid 'Video: mpeg4 (libxvid) (xvid / 0x64697678), yuv420p, 640x346 [SAR 519:520 DAR 24:13]' [21:25] <Bombo> i'm flooded with that msg [21:26] <Bombo> i can't see the status bar [21:27] <JEEB> basically the sar/par field in MPEG-4 Part 2 is an 8bit integer [21:27] <JEEB> and the SAR/PAR you're setting on it is not within 0-255 [21:28] <JEEB> (and yes, that SAR/PAR pretty much means you might as well have it at 1:1) [21:28] <JEEB> but since it's not exactly 1:1 you end up getting 519:520 [21:29] <JEEB> if your stuff is always output as 1:1 SAR/PAR you could just add setsar filter at the end of your filter chain [21:29] <JEEB> with the values 1:1 [21:31] <Bombo> hmmkay [21:32] <Bombo> [Parsed_setsar_1 @ 0x30b6600] num:den syntax is deprecated, please use num/den or named options instead [21:32] <Bombo> i added setsar=1:1 [21:32] <JEEB> then set it 1/1 [21:33] <JEEB> at least that's what it seems :) [21:33] <Bombo> sorry, of course ;) [21:34] <Bombo> Video: mpeg4 (libxvid) (xvid / 0x64697678), yuv420p, 640x346 [SAR 1:1 DAR 320:173] [21:34] <Bombo> msg is gone, lets see if it looks ok then [21:34] <JEEB> it should, the difference in aspect ratio should be minimal [21:34] <Bombo> (and my player plays it right on the tv) [21:34] <JEEB> 519:520 and 1:1 [21:35] <JEEB> well, your player most probably ignores that aspect ratio information anyways :D [21:35] <JEEB> because pretty much nothing reads it from MPEG-4 Part2 [21:35] <Bombo> hmmyeah ok ;) [21:44] <Bombo> btw is there an audio filter in ffmpeg (or external) to 'balance-source-5.1-ac3-volumes-to-stereo' if you know what i mean? [21:45] <Bombo> i recoded a 5.1ac3 audio stream to mp3, but the volume is quite quiet [21:45] <Bombo> but then when there are explosions its pretty loud [21:58] <JEEB> Bombo, ffmpeg by default if you have a new enough build should do downmixing correctly, and yes this often means the effect you are getting. And as funny as it is, that's actually the _correct_ way to downmix it [21:58] <JEEB> you can probably modify the weights and stuff :P [22:01] <llogan> FunkyELF: what do you mean by, "read tags"? [22:08] <FunkyELF> llogan: like id3 tags or whatever mp4tags tags things with [22:08] <FunkyELF> llogan: I found it... it was mp4info [22:14] <Bombo> JEEB: hmmkay... how would i do that? probably not easy according to your ;P [22:14] <JEEB> I've no idea [22:14] <Bombo> or i could buy a 5.1 set ;) [22:15] <llogan> FunkyELF: why don't you just use ffmpeg or ffprobe instead? [22:16] <Bombo> JEEB: it's quite ok, i turn up the volume, the explosions and stuff may disturb the neighbours only ;) but i need to remember to turn it down again, when i switch to normal TV operations [22:19] <Bombo> i was a bit shocked when jay and seth walked into that shop to buy milky way, when suddenly hell broke loose, and blue transporter beams appeared. if you know what i mean. [22:19] <Bombo> ;) [22:40] <viric> when ffmpeg complains about "[output stream 0:1 @ 0x9bf16e0] 100 buffers queued in output stream 0:1", how can I know what that output 0:1 is? [22:40] <viric> I use filter_complex, and 0:1 is only the input. [22:46] <viric> http://sprunge.us/ibMS [22:47] <viric> It's the ffmpeg output, not the commandline :) [22:47] <viric> Maybe it's the Stream #0:1 (libvorbis) ? [22:49] <llogan> that's what I'd assume [22:50] <Lachezar> Hey all& What format should I use to transcode verbatim a set of VOB to a single file? [22:52] <viric> I'll have to check the last concat [22:53] <viric> Lachezar: my wildcard is always mkv [22:53] <Lachezar> I would like not to decode+encode, but copy the video+audio stream. And it seems every VOB file has the streams for English in different place: VTS_04_1: Stream #0.20[0x80], but VTS_04_4: Stream #0.4[0x80] [22:54] <Lachezar> viric: MKV is fine by me, but can I use -vcodec copy -acodec copy there? [22:54] <viric> I expect so [22:54] <viric> no idea about the different streams placing [22:54] <viric> you could also use 'zip' :) zip myfile.zip *VOB [22:55] <Lachezar> viric: I would like to use those on my TV (Samsung whatever). [22:55] <viric> then you should ask the tv [22:55] <viric> I don't know much more :) [22:56] <Lachezar> viric: I'm trying MKV now. [22:56] <Lachezar> What is: 'mpeg_decode_postinit() failure' [22:57] <Lachezar> viric: Seems MKV does not like copy: [matroska @ 0xff0900] Can't write packet with unknown timestamp [22:57] <viric> hm no idea. [22:57] <viric> I never tried the operation you described. [22:58] <Bombo> does mpeg2 fit into mkv? [22:59] <Bombo> Lachezar: you could write into .vob or .m2t and select the streams to copy with -map 0:20 or -map 0:4 [22:59] <Bombo> dunno if you yould select 0x80 streams or $lang ... [22:59] <Bombo> -y+c [23:02] <llogan> mkv is like a pig. it will eat just about anything. [23:03] Action: Lachezar is trying very hard to understand what was written just now. [23:06] <Bombo> Lachezar: what you want to do? sort the streams? [23:06] <relaxed> Lachezar: ffmpeg -i concat:1.vob\|2.vob\|3.vob -map 0 -c copy -f vob combined.vob [23:07] <relaxed> or instead of '-map 0' which means all the input's streams, use -map 0:v -map 0:a etc... [23:07] <Lachezar> Bombo: I'd like to end up with a single file with 1 Video, 1 Audio and possibly 1 Subtitle. [23:08] <Bombo> Lachezar: what you got? 1.vob 2.vob 3.vob? [23:08] <Lachezar> relaxed: Will this work if the audio stream I want is different ID in every .VOB? [23:08] <Lachezar> Bombo: VTS_04_1.VOB, VTS_04_2.VOB [23:08] <Lachezar> Bombo: ... [23:09] <Bombo> you need to sort them first i guess [23:09] <Lachezar> I think the program hates me: Stream #0:1 -> #0:2 (dvdsub -> ass) [23:09] <Bombo> make new vobs with -map so that you got the same [23:09] <Bombo> then concat them like relaxed said [23:10] <Lachezar> Bombo: So, for every VOB i should 'copy' with -map 0:v:0 -map 0:a:?? and the concat:*.vob ? [23:12] <Bombo> i think its: ffmpeg -i VTS_04_1.vob -vcodec copy -map 0:0 -acodec copy -map 0:20 01.vob [23:12] <Bombo> (or -c:v copy and -c:a copy, depends on version) [23:13] <Bombo> and sub: -scodec copy -map what:ever [23:14] <Lachezar> Bombo: I see... This sounds like a bit more work than I thought, but it is what it is& [23:14] <Bombo> Lachezar: copy video stream from source with id 0:0, and audio stream with id 0:20 from source file, into new [23:14] <Lachezar> Funny thing is: If I do not specify -map ffmpeg seems to get the right streams. Probably because the audio I want is the best quality one. [23:17] <Bombo> *shrug* then try the concat line on the orig vobs, maybe works [23:18] <TylerDurden> Just compiled ffmpeg on Debian. Got a 20GB 3D film that cant fit on a USB. Look as I may I cant find a correct way to split this file with ffmpeg. [23:18] <Lachezar> Bombo: Probably not. I'm actually having problems with the copy: buffer underflow i=1 bufi=16170 size=23816 [23:19] <Lachezar> relaxed: Pardon! [23:20] <relaxed> follow the bot's instructions [23:20] <Lachezar> relaxed: Are there any significant differences between ffmpeg and avconvert? 'Cause I'm using *buntu, and it seems ffmpeg is aliased to avconvert [23:20] <Bombo> TylerDurden: -ss skip to timestamp and -t duration [23:22] <TylerDurden> Bombo the -codec copy -ss xxxx part I think I got but I am not to sure about how duration would be used? [23:24] <Bombo> -ss 00:30:00 seeks to 30. minute -t 00:30:00 copies from 00:30:00 30 minutes (to 00:60:00) [23:24] <relaxed> it also takes seconds [23:24] <TylerDurden> hmm, Ill give that a go. Thx! [23:24] <Bombo> and .ms? ;) [23:25] <TylerDurden> ? [23:25] <relaxed> yes [23:25] <TylerDurden> what is ms? [23:25] <Bombo> milliseconds [23:25] <TylerDurden> ahhh [23:25] <TylerDurden> thought you meant that weird OS thingy... [23:26] <Lachezar> F*ck: & -map 0:0 -map 0:0 -c:v copy -c:a copy & This ends up with two video streams copied (and breaks anyway :(). [23:26] <Bombo> thats spelled with a $ ;) [23:26] <relaxed> also, -ss before the input does a quick seek to that point; -ss after the input decodes to that point. [23:26] <TylerDurden> haha [23:26] <relaxed> -t always belongs after the inputs [23:27] <Bombo> Lachezar: yes 0:0 is the video stream [23:27] <Bombo> Lachezar: should give the audio stream id for the audio stream copy [23:28] <relaxed> Lachezar: read --> man ffmpeg|less +/^'[ ]{7,}-map' [23:31] <Lachezar> Pasted: http://pastebin.com/wCsaS1S2 [23:31] <Lachezar> relaxed: I am /trying/ to read https://trac.ffmpeg.org/wiki/How%20to%20use%20-map%20option [23:33] <Lachezar> Ahhh... I see... The demo has german FIRST, english SECOND :-/ [23:33] <Bombo> Lachezar: try adding -f vob [23:33] <Lachezar> The paste http://pastebin.com/wCsaS1S2 shows some buffer problem. [23:34] Action: Lachezar finds skipping the -map better. [23:34] <Lachezar> Thos 'packet too large' and 'buffer undeflow' seem to break things badly. [23:35] <Lachezar> Bombo: Adding -f vob did not help: still tons of errors. [23:36] <Bombo> and then, try ffmpeg ;) [23:36] <Lachezar> Bombo: I might end up firing a VM with a Debian as next step. [23:37] <Lachezar> Does anyone know how can I find out what language is an Audio/Subtitle sub-stream in a .VOB? [23:37] <Bombo> debian has old ffmpeg version [23:37] <relaxed> Lachezar: try my static build http://johnvansickle.com/ffmpeg/ [23:38] <TylerDurden> awk '/Duration/ {gsub(/,/, "", $2); split($2, a, /:/); print 3600*a[1] + 60*a[2] + a[3]}' ffprobe -i "$@" [23:38] <Bombo> Lachezar: iirc 0x80 is always german [23:38] <TylerDurden> sry darn term is so clear I dont see what is what... =P [23:38] <Bombo> Lachezar: my ffmpeg shows (deu) and (eng) instead of 0x [23:38] <Lachezar> Bombo: It seems that is not a 'rule' in my case BG seems to be 0x80 :-/ [23:39] <Lachezar> relaxed: tar-bunzip and I'm ready to go? [23:40] <relaxed> yes [23:40] <Bombo> Lachezar: BG? [23:41] <msmithng> read my mind [23:41] <Lachezar> Bombo: BG = Bulgarian... [23:41] <msmithng> ah [23:41] <Bombo> try ffmpeg ;) [23:42] <Lachezar> relaxed: Sorry to be N00B-y what is ffmpeg10bit? [23:42] <Bombo> what does it say? [23:42] <Bombo> what does the ffmpeg say? *sing* [23:43] <relaxed> Lachezar: it's used for creating 10bit h264 video. you don't need it [23:43] <Lachezar> Bombo: Stream #0:4[0x80]: Audio: ac3, 48000 Hz, 5.1(side), fltp, 448 kb/s [23:43] <Lachezar> Bombo: That's from ffmpeg, not avconv [23:43] <Bombo> Lachezar: hm. *shurug* [23:45] <Lachezar> relaxed: Same thing with your static version: lots of 'buffer underflow' and 'packet too large'. Do you want another pastebin? [23:45] <relaxed> yes [23:49] <Bombo> (got no vob to check, but a mkv says Stream #0:1(ger) and Stream #0:2(eng), could be possible that vobs just have 0x ids for lang, but thats not the problem anyways ;)) [23:51] <Lachezar> relaxed: http://pastebin.com/diTzZ5Y4 [23:53] <relaxed> you need to map the specific streams you want in the output. ffmpeg -i input -map 0:0 -map 0:21 -c copy -f vob output.vob [23:55] <Lachezar> relaxed: Are you sure? It seems it mapped those correctly automatically: Stream #0:1 -> #0:0 (copy), Stream #0:21 -> #0:1 (copy) [23:57] <relaxed> It's a good practice to get into because ffmpeg won't always do what you think. [23:58] <Lachezar> relaxed: OK, but even if I specify -map 0:1 -map 0:21 -vcodec copy -acodec copy I get the same errors, and the resulting VOB is choppy+skippy+out-of-sync [23:59] <relaxed> try -f dvd [23:59] <relaxed> maybe add -copyts [00:00] --- Sat Jan 4 2014

1 0

ffmpeg-devel.log.20140102
by burek 03 Jan '14

03 Jan '14

[00:06] <Compn> did someone update copyright year yet ? [00:14] <Zeranoe> Is there any preference to use YASM, over nasm [00:18] <saste> Compn, of course [00:18] <JEEBsv> Zeranoe: I was going to say that nasm was dead, but it seems they're alive again (at least there's a release from decemberr). In general though, during the time that nasm was dead, most people moved to yasm. Also I have no idea if yasm still supports more stuff than nasm [00:54] <cone-427> ffmpeg.git 03Stefano Sabatini 07master:f6707f0689e0: doc/muxers/hls: extend documentation [00:54] <cone-427> ffmpeg.git 03Stefano Sabatini 07master:6cceef2ff9d4: doc/muxers/segment: reference hls muxer [00:54] <cone-427> ffmpeg.git 03Stefano Sabatini 07master:0aa325457dde: doc/muxers/segment: clarify ambiguos reference [01:44] <michaelni> Anssi, seems my mail didnt make it (SMTP error from remote server after RCPT command ...) but what i wanted to say, about the hls pull req or that other patch, feel free to cherry-pick/push them if you think they are ok [01:45] <Anssi> michaelni: ok [01:46] <Anssi> michaelni: about the mail error, was it from my end? if so can you give a more complete error message? [01:47] <michaelni> i wish i knew, i do have some mysterious problems with gmx ... the error was "host: mail.iki.fi 4.4.3 Please try again later" but its identified as "A message that you sent could not be delivered to one or more of [01:47] <michaelni> its recipients. This is a permanent error. The following address [01:47] <michaelni> failed:" [01:49] <Anssi> michaelni: "the following address" was the iki.fi one? [01:49] <michaelni> yes, i posted out of order sorry [01:51] <michaelni> Anssi: http://pastebin.com/dkpT9gJX [01:58] <Anssi> michaelni: ok thanks. strange error though, not many hits for "4.4.3 Please try again later" in google either... [02:20] <cone-427> ffmpeg.git 03Anton Khirnov 07master:58f0d164def5: Makefile: add html template files to the doxygen deps [02:20] <cone-427> ffmpeg.git 03Michael Niedermayer 07master:9c29967ae691: Merge remote-tracking branch 'qatar/master' [02:20] <cone-427> ffmpeg.git 03Michael Niedermayer 07master:0f378d86321e: doc/doxy: remove unused files [02:30] <cone-427> ffmpeg.git 03Maxim Poliakovski 07master:4663a550f584: atrac3plus: Add forgotten initialization of an internal struct [02:36] <cone-427> ffmpeg.git 03James Almer 07master:8feaa83d479a: fate: Add OggVP8 demux test [06:24] <Anssi> michaelni: got a reply from iki.fi admins, GMX mail server is presently on several blacklists (SORBS, SPAMCOP) so they are automatically greylisting incoming mails from there (the first message gets a temporary error and the sender server should try again soon), but the gmx mail server isn't handling it properly according to the SMTP spec (which they are saying is the first such server they've seen, they've had this active for several ... [06:24] <Anssi> ... years now) [06:26] <Compn> Anssi : gmx has been on blocklist for years [06:26] <Compn> Anssi : the point is, why is your isp blocking emails without your consent to do so? :) [06:28] <Anssi> Compn: it is not blocking, just greylisting... and it is not my ISP but a separate service... and it is actually in their FAQ, I just had missed it [06:29] <clever> my isp actively blocks ALL outgoing port 25 traffic, except to their own smtp server [06:30] <clever> forcing me to use their servers and their filtering [06:30] <clever> ive found that to be standard in my area [06:30] <Anssi> yes that is the standard here as well [06:32] <Anssi> michaelni: here's the reply I got: http://pastebin.com/GTg29k5P [06:59] <Compn> Anssi : iirc michael and gmx tried to get off the blocklist at sorbs [06:59] <Compn> the consensus was ... once on the blacklists , you aint getting off [07:00] <Compn> and then a bunch of other devels chimed in with how shitty sorbs has become [07:02] <Compn> https://lists.ffmpeg.org/pipermail/ffmpeg-devel/2006-May/015819.html [07:02] <Compn> from 2006 :) [07:02] <Anssi> Compn: ok... so I wonder how gmx not respecting 451 temporary error has gotten unnoticed this far, greylisting is used on quite many servers... [07:03] <Compn> As you will probably know already there is/was an issue with several developers [07:03] <Compn> (rich & oded at least) being unable to email me, i complained to gmx, heres [07:03] <Compn> their awnser (in german!, in english it means something like: we are already [07:03] <Compn> aware of the problem and are working on a solution, we are sorry about the [07:03] <Compn> problems and please be patient...) [07:03] <Compn> working on a solution for 8 years now :) [07:03] <Compn> http://lists-archives.com/mplayer-dev-eng/09441-gmx-sorbs-awnser.html [07:04] <Compn> er, heres the real link to that http://mplayerhq.hu/pipermail/mplayer-dev-eng/2006-May/043345.html [07:08] <Compn> Anssi : but sure, gmx might be doing something wrong with 451 error, no idea :) [07:10] Action: Compn goes sleeps town [09:38] <cone-48> ffmpeg.git 03Stefano Sabatini 07master:e3868cc588c4: doc/muxers/hls: change "can used" to "can be used" [10:31] <j-b> 'morning [10:45] <saste> j-b: 'year [10:48] <michaelni> 'newyear [11:56] <michaelni> ubitux, do you have a moment to look at "[FFmpeg-devel] [PATCH 02/18] avformat/webvttdec: add support for HLS WebVTT MPEG timestamp maps" ? [11:56] <michaelni> i think it needs a comment / review from you [11:57] <ubitux> oh [11:57] <ubitux> i'll have a look, missed that, sorry [12:24] <cone-48> ffmpeg.git 03Peter Ross 07master:7340718d1c45: avformat/bink: recognize bink2 files [14:59] <cone-48> ffmpeg.git 03Michael Niedermayer 07master:4843227b2ca6: avcodec/iff: ensure that runs with insufficient input dont leave uninitialized bytes in the output [18:04] <cone-48> ffmpeg.git 03Michael Niedermayer 07master:6433b393ba2b: avcodec/aacps: fix number of bands used with ipd/opd [18:58] <cone-48> ffmpeg.git 03Michael Niedermayer 07master:c925e960e83d: mp3on4: fail when the header check fails [22:17] <cone-48> ffmpeg.git 03Michael Niedermayer 07master:2e59ffbb7964: avcodec/mace: check buffer size [00:00] --- Fri Jan 3 2014

1 0

ffmpeg.log.20140102
by burek 03 Jan '14

03 Jan '14

[12:51] <kurkale6ka> Hi, I have a bunch of .mp4 videos and lots of them are rotated. Tried mediainfo with grep Rotation... nothing, exiftool always reports Rotation 0 and with ffmpeg -i there is no Rotation info in the metadata. Do I have any remaining options to test this? [12:51] <kurkale6ka> I wanna correct the rotation in batch so I need to know first which one are to be rotated [13:13] <kurkale6ka> Ok, maybe I can play with the width:height. Is there a simpler way of getting them than ffmpeg -i ... | grep '#0:0' | awk ... [13:15] <zap0> there are several formats data can be extruded too. plain text may not be the most reliable [13:17] <kurkale6ka> So what do you suggest [13:42] <zap0> depends on the parsing tools you have [13:44] <kurkale6ka> I am on linux [13:52] <zap0> my condolensenses [14:19] <viric> kurkale6ka: check ffprobe, and the different output mechanisms it has [14:19] <viric> for example: ffprobe -of default FILE [14:19] <viric> | grep whatever [14:22] <kurkale6ka> thanks viric. That's what I get (same output as ffmpeg -i) and there is nothing related to rotation: http://bpaste.net/show/163875/ [14:22] <viric> I've no idea about rotation flags. I never saw one. [14:22] <viric> I'm not sure they exist. [14:23] <kurkale6ka> apparently they do in some cases [14:40] <kurkale6ka> I did the following to rotate my video: ffmpeg -i video-2013-10-19-17-17-30.mp4 -c:a copy -vf 'transpose=1' out.mp4 but I got a bit rate of 216 where it is 750 in the original so I added -b:v 752k. Is this the correct way to proceed? [14:48] <viric> if it were a flag, you could -c:v copy. no? [14:50] <kurkale6ka> viric: I tried this but the transposing is then not done [14:53] <kurkale6ka> Hmm I tried with a bit rate much higher than the original and I got better quality. How is that possible [14:56] <kurkale6ka> Also Duration: 00:00:48.64, start: 0.000000, bitrate: 772 kb/s but in Stream #0:0 it reads 752 kb/s. Why the difference? [14:57] <viric> better quality than the original? [14:57] <viric> it can be either bad perception or a failure of matrix [14:58] <kurkale6ka> Yes, there was less pixelation (actually none visible) [15:03] <DelphiWorld> hi people [15:03] <DelphiWorld> anyone know rtp streaming well? [15:04] <DelphiWorld> ffmpeg -re -f flv -i rtmp://X.X.X.X:1935/live/livestream -acodec copy -vcodec copy -f mp4 -f rtp rtp://239.100.100.1:1234 [15:04] <DelphiWorld> this is not working... [15:05] <DelphiWorld> i want to stream it to RTP, but change the format to mpeg4 [17:00] <kurkale6ka> This command rotates my video and encodes it from mpeg4 part 2 to h264: ffmpeg -i video-2013-10-19-17-17-30.mp4 -c:a copy -vf 'transpose=1' out.mp4. Result is a file which is 1.5M in size as opposed to 4.5M. This is normal I suppose since the compression is stronger and I shouldn't be woried about loss of quality (which I can't detect visually) [17:05] <wald0> where i can found the "best" options for compress videos generically? (movies or camera videos which are huge, to have the smallest size possible with a perfectly-viewable quality) [17:05] <wald0> i assume that i should use h265 + mp3 for that (ogg is better, but i think that is not so compatible with players) [17:06] <wald0> i forgot to say: with a standarized compatibility so that will perfectly work in any player" [17:09] <JEEB> first of all there's still no "usable" H.265/HEVC encoders [17:10] <JEEB> as in such that give better quality within a sane time period and have a sane API [17:10] <JEEB> thus you probably meant H.264/AVC [17:11] <wald0> bad typed, sorry [17:12] <JEEB> anyways, in general it's just libx264 + crf rate control + the slowest preset you can take [17:12] <JEEB> crf default is 23, preset default is medium [17:12] Action: DelphiWorld is back [17:12] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset [17:12] <JEEB> ^ listing of the presets [17:12] <JEEB> anything but ultrafast and placebo should be usable [17:12] <DelphiWorld> hny 2014 JEEB [17:12] <JEEB> with crf you find the highest value that still looks good [17:12] <JEEB> aand, that's pretty much it [17:13] <JEEB> as for audio, if your source has compressed lossy audio already, you might just want to copy that instead of re-encoding [17:13] <JEEB> also I think nowadays AAC is more supported than MP3 in most containers and use cases [17:13] <JEEB> if you need to re-encode [17:13] <JEEB> DelphiWorld, yes, happy new year to you, too [17:14] <DelphiWorld> thank JEEB [17:14] <DelphiWorld> JEEB: ffmpegfull new year:) [17:14] <wald0> but aac has teh same "enough" quality as mp3 ? and... it has also a small size like mp3 ? i noticed that a good amount of the video size is eated by the audio tracks [17:15] <JEEB> AAC is newer than MP3, and good AAC encoders do make a better result at the same bit rate than MP3 encoders [17:15] <sacarasc> Depending on the encoder, AAC is usually equal or better than MP3 at the same bitrate. [17:15] <DelphiWorld> i hate mp3 [17:15] <sacarasc> Why? [17:16] <DelphiWorld> sacarasc: aac is better in 64kbit/s you get better thant a 128kbit/s mp3 [17:16] <sacarasc> Who would use 128kbps MP3, though? :p [17:17] <DelphiWorld> sacarasc: you? :) [17:17] <sacarasc> Nope. [17:17] <sacarasc> I tend to go for, on the rare occasions I need MP3, V0. [17:17] <DelphiWorld> JEEB: if i ffprobe a file or a stream, how should i know if the AAC is LC or He-AAC? [17:17] <JEEB> depends on if the information you can get from ffprobe contains the profile :P [17:18] <JEEB> it's a profile, after all [17:18] <DelphiWorld> JEEB: no, i dont see profile evean if its encoded by ff [17:18] <JEEB> no, I mean if you can output that variable by poking at the settings. if not, then you're out of luck [17:18] <DelphiWorld> lol... [17:19] <JEEB> but I would be surprised if you couldn't output at least some of the decoder info :P [17:19] <JEEB> but as you can guess, I have never needed that part of it and thus I just don't know [17:19] <DelphiWorld> :) [17:22] <wald0> DelphiWorld: i noticed some lossy of quality in mp3, but when used ogg i have not see them at all (same size/bitrate,e tc) [17:22] <DelphiWorld> yeah ogg is good, but not device wide like aac/mp3 [17:23] <JEEB> the vorbis encoders are good, esp. the aotuv one [17:23] <JEEB> and mp3 is _pretty_old_ by now [17:23] <DelphiWorld> JEEB: can you help me with something:) [17:24] <sacarasc> It's only in its 19th year, JEEB. :p [17:24] <wald0> yeah, unfortunately is not widely compatible, thats teh only reason that i dont use it for my videos :/ [17:24] <wald0> ok so... how i can know which AAC encoder I have to see if is enough good in quality/size ? [17:24] <wald0> debian wheezy here [17:25] <DelphiWorld> JEEB: i have a rtmp stream. i want to capture it & output it to a rtp stream. any clue please? [17:25] <JEEB> you'll need to compile ffmpeg with fdk-aac [17:25] <JEEB> because one cannot distribute the binaries [17:25] <JEEB> because of fdk-aac's license [17:25] <DelphiWorld> JEEB: lol dont have any issue with ffmpeg build [17:25] <JEEB> DelphiWorld, I was talking to wald0 there :P [17:25] <DelphiWorld> JEEB: here's the stream: rtmp://5.9.151.112:1935/live/livestream [17:25] <wald0> JEEB: then i have a double problem, i want to distribute the tool that im going to write :/ [17:26] <DelphiWorld> lol JEEB [17:26] <JEEB> you can distro the tool just fine, you just have to handle the fdk-aac part separately :P [17:26] <JEEB> which is unfortunate of course [17:26] <JEEB> but not much you can do [17:26] <JEEB> there's the internal aac encoder, which is the best of the LGPL/GPL compatible ones [17:27] <JEEB> DelphiWorld, I don't know jack about rtp(s) output but I really don't see anything special in the input part because rtmp is supported in ffmpeg itself (thank effing goodness) [17:27] <wald0> if the gpl ones are not "pretty bad to use" by some quality/size reason, i think that i should better go with these ones for now [17:28] <JEEB> the libavcodec internal one isn't exactly fabulous [17:28] <JEEB> there's a patch set on the trac issue tracker being developed to make it better tho [17:29] <wald0> mmmh, and how i could "easly" integrate the fdk-aac one in the default shipped ffmpeg ? [17:29] <wald0> or i -really- need to compile ffmpeg with it ? [17:29] <JEEB> yes, you need to [17:30] <DelphiWorld> JEEB: rtmp is supported no problem. but i dont know how to output it into mpeg4 stream and then distribute it through rtp. [17:30] <DelphiWorld> -f rtp rtp://xxxxxxxxxx this is ok. but how to change the format to mpeg4 [17:30] <JEEB> you mean MPEG-4 Part 2? -c:v libxvid or mpeg4? [17:31] <DelphiWorld> yep [17:31] Action: DelphiWorld guess wald0 is building a IPTV solution... [17:31] <JEEB> I mean, you could have looked at -codecs... [17:31] <DelphiWorld> JEEB: not transcoding... [17:31] <DelphiWorld> copy audio/video codec but just change transport [17:31] <JEEB> so you want to stream the mpeg-4 container? that's not Part 2. [17:32] <JEEB> I'm not sure if you can even do that :P [17:32] <JEEB> I mean, yes -- you can enable "movie fragments" [17:32] <DelphiWorld> JEEB: i'm not expert in rfc's:) [17:32] <JEEB> but then you have rtp(s) [17:32] <JEEB> in other words, unless you really need to stay away from weird setups. for your own good [17:32] <JEEB> I've only heard of rtp(s) and mpeg-ts in rtp(s) [17:33] <JEEB> and I have no effing idea what ffmpeg supports and doesn't support :P [17:33] <DelphiWorld> JEEB: my need: rtmp stream is streaming. want to output another stream in rtsp to be able to view it in other devices like simbian. [17:33] <JEEB> test yourself and look for information on the internets and in the source code [17:33] <JEEB> find someone else to poke. I have said enough already :P [17:33] <wald0> I wonder if will be better to use AAC from gpl or mp3 instead [17:34] <DelphiWorld> wald0: aac-he is your brother [17:34] <JEEB> wald0, generally you want AAC for most things that play video (as it's pretty much the quasi-standard). I've only seen MP3 in some H.264+MP3 in FLV combos [17:35] <wald0> well, in fact my ffmpeg is from marillat, not debian wheezy, how i can know if has fdk-aac ? [17:35] <JEEB> -codecs [17:35] <JEEB> and look for the things there [17:35] <JEEB> although [17:35] <JEEB> if they distro ffmpeg with fdk-aac they are distributing an enable-nonfree binary [17:35] <JEEB> which is not supposed to happen [17:35] <DelphiWorld> lol [17:35] <wald0> DEA.L. aac AAC (Advanced Audio Coding) (encoders: aac libfaac libfdk_aac libvo_aacenc libaacplus ) [17:35] <JEEB> as in, you should not distro that [17:36] <JEEB> heh [17:36] <wald0> tell that to mr. marillat :þ [17:36] <JEEB> I don't give a flying fuck unfortunately :) [17:36] <wald0> i wonder how the marillat guy does for distribute these things then, if is not "legal" ? [17:36] <DelphiWorld> WTF work! [17:36] <JEEB> it is not license-matching, but if they release the source code [17:37] <JEEB> then at least the ffmpeg side doesn't care too much [17:37] <DelphiWorld> but then cant play it... [17:37] <JEEB> also as I said, only binary distribution is NG, so if you compiled that thing on your own machine, that's 100% OK as well [17:37] <JEEB> most people pick the compilation locally way [17:38] <DelphiWorld> ffmpeg -re -f flv -i rtmp://5.9.151.112:1935/live/livestream -f mpegts rtp://239.100.100.1:1234 [17:38] <DelphiWorld> is that ok JEEB ? [17:38] <JEEB> I told you already X times, I have no idea :P [17:39] <wald0> nah, ppl would use the one packaged for debian (or maybe marillat) here [17:39] <DelphiWorld> JEEB: lollol but what about transport [17:39] <wald0> so... if i encode it with fdk-aac, it would be playable in a compatible way ? (in windows machines, or linuxes that doesn't has ffmpeg compiled with that fdk) [17:40] <JEEB> yes, all encoders for a format in theory should output stuff that decodes accordingly to that specification :P [17:40] <DelphiWorld> :) [17:40] <JEEB> there are some examples where some encoder borked something, but in general specifications "work" [17:40] <JEEB> wald0, well I'm talking about people who specifically want to use fdk-aac and don't want to use anything grey [17:41] <JEEB> for those people it's generally the simplest way to just compile fdk-aac and ffmpeg with it :P [17:41] <DelphiWorld> can guys someone help me please...:( [17:41] <DelphiWorld> i want to take my loved tv with me [17:42] <wald0> what should be the fallback codecs to use if there's not fdk ? [17:42] <wald0> DEA.L. aac AAC (Advanced Audio Coding) (encoders: aac libfaac libfdk_aac libvo_aacenc libaacplus ) [17:42] <wald0> just "aac" or "libfaac" ? mmh.. [17:42] <JEEB> quality wise you should basically do: fdk-aac -> libfaac -> aac [17:42] <wald0> ok thx [17:43] <JEEB> funny enough distributing a binary with faac and aacplus is even worse than distributing one with fdk-aac if I recall correctly :P [17:43] <JEEB> since those use the reference code, which the creator of the project surely didn't license [17:44] <DelphiWorld> lol [17:44] <JEEB> fdk-aac is an official thing and you just have to comply to the license [17:44] <JEEB> which by default at least is incompatible with (L)GPL [17:44] <JEEB> vo-aacenc is apachev2 and thus v3 compatible for (L)GPL, it just sucks [17:49] <wald0> ok so... now im thinking on the features that i need, it should generically convert (compress) videos in a simple and basic interface [17:50] <wald0> i have noted options for select quality and for change resolution, what other things can be useful for a simple app ? [17:50] <wald0> handy* app, is the word :) [17:56] <DelphiWorld> JEEB: its pocible to open a full duplex session bethwan 2 ffmpeg's? [18:07] <wald0> i assume that a video compressor doesn't need features/options at all ? :) [18:13] <calcifea> i'm trying to convert a video to vp8 with that command line "-c:v libvpx -crf 10", still crf is not used.. any idea? [18:13] <DelphiWorld> mmmmmmmmmm, no one that use rtp streaming here at all? [18:59] <DelphiWorld> wb saste [20:47] <ddsss> how to convert .swf to .wav? [20:47] <ddsss> ffmpeg -i file.swf -acodec copy out/file.wav -> creates empty wavs [20:52] <viric> why you use acodec copy? [20:53] <viric> maybe a wav cannot store the codec in swf [20:53] <viric> (is swf a video?) [20:54] <ddsss> viric, swf is audio afaik [20:55] <ddsss> viric, when I do ffmpeg -i file.swf it says: Duration: N/A, bitrate: N/A Stream #0.0: Audio: pcm_s16le, 5512 Hz, 1 channels, s16, 88 kb/s [20:55] <viric> No idea. [20:55] <viric> ah, pcms16le. [20:55] <viric> no idea what happens [20:57] <rcombs> SWF is a Flash file [20:57] <rcombs> it's neither video nor audio; it's more like a program [20:57] <rcombs> I wouldn't expect ffmpeg to handle it at all [20:57] <viric> I imagined that, first [20:57] <viric> maybe ffmpeg takes it as raw audio? [20:57] <rcombs> possibly [20:58] <rcombs> either way, you may be able to extract the actual audio files using an SWF decompiler [20:58] <viric> ffprobe on a text file, tells me: Stream #0:0: Video: ansi, pal8, 640x400, 25 fps, 25 tbr, 25 tbn, 25 tbc [20:58] <rcombs> they're probably WAV or MP3 already [20:58] <viric> :) [20:58] <rcombs> yeah, those look like defaults [21:00] <ddsss> k. I'll screw around with them some more:) they do play in audacious btw. [21:46] <wald0> what means profile high? for h264 [21:48] <JEEB> a feature set [21:49] <JEEB> profiles mean available features used by the stream, levels are related to the memory usage on the decoder side [21:52] <llogan> wald0: generally you don't need to explicitly set a profile unless your device requires it [21:52] <llogan> just let the encoder choose [22:20] <wald0> i see [22:27] <wald0> ah, just got a feautre in mind to include in that compressor tool :) --> normalize audio! [22:27] <wald0> pretty wanted thing [22:37] <average> uhm hello ? [22:37] <average> trying to do avi => webm [22:37] <average> I'm losing a lot of video quality over the conversion [22:37] <average> I don't know what switches to use to adjust quality [22:39] <average> I could use pastebin, but it's just 1 line that I need to show [22:39] <average> ffmpeg -i output.avi -acodec libvorbis -ac 2 -ab 96k -ar 44100 -b 345k -s 1366x768 -crf 20 -minrate 700k output.webm [22:40] <average> http://pastebin.com/y685C0s0 [22:40] <average> I don't get an error [22:40] <average> but the problem is I'm losing video quality [22:40] <average> I tried using -minrate and -crf to get better quality [22:40] <average> minrate helped a bit but not much [22:40] <llogan> the complete console output is missing [22:41] <llogan> the console output is important. you don't have to encode the whole thing. you can add -t 10 or something. [22:43] <average> http://pastebin.com/eJdYbi5j [22:43] <average> llogan: added console output [22:43] <average> I don't understand most of those numbers [22:43] <llogan> you're not using ffmpeg from FFmpeg. [22:43] <llogan> so you'll have to get help at libav or use ffmpeg from FFmpeg [22:43] <llogan> http://trac.ffmpeg.org/wiki/UbuntuCompilationGuide [22:44] <llogan> or [22:44] <average> I have a version of ffmpeg compiled manually by the guide you wrote about above [22:44] <llogan> your console output indicates that you're using a Libav product [22:44] <llogan> try #libav [22:45] <llogan> if you want to continue using it [22:45] <average> I don't want to use it [22:45] <average> I'll use a static build of FFmpeg [22:45] <llogan> is ffmpeg in ~/bin? [22:46] <average> downloading static build [22:46] <llogan> but you said you already compiled ffmpeg from FFmpeg [22:47] <average> no, F my compiled ffmpeg [22:47] <average> I'll use what you gave me above [22:48] <llogan> ok. now that you're (probably) using the correct stuff you can refer to http://trac.ffmpeg.org/wiki/vpxEncodingGuide [22:48] <average> http://pastebin.com/LB31zAA4 [22:49] <average> this is the output I get using the static build of FFmpeg [22:49] <average> llogan: there's no avi mentioned in the vpxEncodingGuide [22:49] <average> I was looking for avi because I have an avi.. maybe I'm wrong [22:50] <average> llogan: I tried what you gave me, works great [22:51] <llogan> you can use avi, or any other supported format, as input. [22:51] <average> actually it doesn't work great, it's fantastic ! [22:51] <llogan> the .mp4 in the guide are just examples [22:51] <average> that's some awesome stuff [22:51] <average> llogan: is the static FFmpeg compiled with x11grab too ? [22:51] <llogan> no, you'll have to compile for that [22:52] <llogan> what command did you end up using for the webm output? [22:52] <average> ffmpeg -i input.mp4 -c:v libvpx -crf 10 -b:v 1M -c:a libvorbis output.webm [22:52] <average> in the guide you gave me [22:52] <average> changed input.mp4 to output.avi [22:52] <average> and it worked fine [22:53] <wald0> with CRF we can set the quality, right? but how we can set it in a way that "will not be bigger than the original video" ? (which is totally useless) [22:53] <llogan> average: might want to try out using -qmin and -qmax as shown in the guide too [22:54] <JEEB> wald0, you are not dealing with the original video, you are dealing with raw video. You just set the highest crf value that still looks good to you, that's all [22:54] <llogan> wald0: i assume you mean -crf for libx264. [22:55] <matham> I was wondering what align in e.g. av_image_alloc does? does it affect the total size of the buffer for each plane, or e.g. linesize as well to make to it larger than width? If the latter, why doesn't av_image_fill_linesizes have an align param? [22:55] <average> llogan: those guys who made the static builds of FFmpeg... do they have their build scripts somewhere ? I could use them.. [22:55] <average> llogan: I suppose it won't be trivial to add x11grab to the static build... I've had troubles in the past with static builds... [22:55] <JEEB> wald0, I mean, if it ends up being bigger than your original then that's life :P A slower preset might help there, of course. [22:55] <llogan> average: you can ask them. burek is here sometimes. relaxed is here now, but i don't know if he is afk [22:56] <average> relaxed: you afk ? [22:56] <wald0> mmmh, for my old command to recode (compress) videos i got sometimes the size bigger, so the compressing is useless there then [22:56] <wald0> from* [22:56] <JEEB> if that is really the highest crf value that still looks good [22:56] <JEEB> and if you're already on the slower preset [22:56] <JEEB> then guess what [22:56] <wald0> it is not possible to know/set a limit of quality that will be set "equal" to the original one ? [22:56] <JEEB> no [22:56] <JEEB> we don't have thousands of chinese children in our basements yet [22:57] <JEEB> CRF is the closest thing we have to "constant quality", basically (constant rate factor) [22:58] <ChocolateArmpits> can anyone give an explanation to -bufersize tag and how it works along with -maxrate? [22:58] <llogan> maybe you should ask yourself why you want to re-encode in the first place [22:58] <JEEB> so you really have to find the highest value that still looks for you (the value doesn't keep static between presets because slower presets make the encoder "see" more things and thus same CRF with a slower preset might actually end up bigger) [22:58] <wald0> mmmh, and knowing CRF and the AAC parameters, do we can know an aprox. size of the resulting video? [22:58] <JEEB> no [22:58] <JEEB> the idea of CRF is to be free of any bit rate constraints and just run an ABR rate control over the whole clip that more or less kind of feels like "constant quality level" [22:59] <wald0> not easy to do an "automagic" tool for compress videos then :/ [23:00] <JEEB> esp. since not everyone's eyes are the same, too :P [23:00] <JEEB> *are not the same [23:00] <wald0> from camera's is easy since they are extremely bloated in size, but for other videos... sometimes they are well compressed (very rare) and sometimes their quality is just real crap (see pron) [23:01] <JEEB> yes, and you can't do jack about that. If it's already well compressed or crap (thus already very small), then you really can't do much 'bout it [23:01] <wald0> so compressing these 2 last ones can result in a bigger file [23:01] <JEEB> since if someone is encoding, there's usually a reason for that (the original file can't be used for thing X or whatever) [23:01] <JEEB> otherwise person shouldn't just be re-encoding [23:02] <JEEB> wald0, I recommend you just stop trying to make a "will make the input smaller always" kind of thing :P [23:03] <JEEB> will make your life much easier, and will keep you away from thinking derpy things [23:03] <JEEB> it's much more useful to actually keep your output within some frames of compatibility or whatever [23:03] <JEEB> if you are going to be making an app for the random luser to encode video for a device or whatever [23:03] <JEEB> ChocolateArmpits, -bufsize sets the amount of buffer the client will be buffering, and maxrate sets the maximum average bit rate within this buffer. So if you have maxrate of 2000 and bufsize of 3000, you'd have a 1.5 second buffer with the minimum transfer rate needed to watch the stream being maxrate (+ audio bit rate + some container overhead) [23:03] <average> llogan: so thank you very much sir, basically now I have a solution to make webm screencasts. Unfortunately it involves using two ffmpegs, one that comes with Ubuntu, and a static one to make avi => webm , buuut, I am ok with that. I don't mind [23:04] <average> as long as it works, it works [23:04] <wald0> JEEB: but i can tell you that my old compressor script (using almost static parameters) always saved me a ton of space on lots of videos (specially movies), let's say 3-4 times smaller [23:04] <wald0> average [23:04] <JEEB> well, I'm still pretty sure you didn't poke the CRF value [23:04] <average> I'm sure I could rebuild ffmpeg with x11grab but I am not open to the hassle entailed by that (because I know it'll take more than one try, and each try would cost me around 5-10minutes) [23:05] <JEEB> and actually tried to find the highest that still looks good for you [23:05] <wald0> mmh, let me see [23:05] <JEEB> and if you actually did, then the shit that 'saved you space' actually WAS NOT UP TO THAT QUALITY [23:05] <JEEB> it's simple business, either you do 2pass and set an overall bit rate, or you do single-pass CRF and care about the quality [23:05] <JEEB> both have as good compression ratio [23:06] <ChocolateArmpits> JEEB, that sounds quite strange. I've seen the highest suggested bufsize for dvd compliant video to be 1792kilobits, but for actual video most often 8000kilobits [23:06] <JEEB> just the difference in what is more important [23:06] <wald0> well, actually was made with mencoder and "qp" parameter [23:06] <ChocolateArmpits> Is there something I'm missing? [23:06] <JEEB> wald0, "gg constant quantizer" -- even worse than CRF :P [23:06] <wald0> mencoder -ovc x264 -x264encopts me=umh:subq=9:nr=100:threads=auto:crf=26 -oac mp3lame -lameopts abr:br=80:vol=0:mode=1:aq=0 -sid 42 [23:06] <JEEB> uhh, no -- that has crf [23:06] <wald0> more exactly, good parameters, the minimal amount to see "good" a video [23:06] <JEEB> and crf 26 at that [23:07] <JEEB> anyways, set the highest crf that still looks good and use the slowest preset you can take [23:07] <wald0> oh wait, true... i was reading a comment which said "qp" instead of "crf" :) [23:07] <JEEB> I'm pretty sure I linked it to you [23:07] <JEEB> wald0, NEVER USE QP INSTEAD OF CRF [23:07] <JEEB> seriously [23:07] <JEEB> super fucking seriously, unless you're a developer [23:07] <wald0> why comments doesn't make apps to fail compilation ? lol [23:08] <JEEB> of course this only applies to x264, pretty much everything older only has average bit rate or qp modes [23:08] <wald0> JEEB: i used it in the past but switched to crf let's say 1-2 years ago [23:08] <JEEB> so qp was the "closest" you had to something that didn't limit the bit rate [23:08] <wald0> thats why my comment where it said that is outdated [23:08] <llogan> average: compiling isn't that bad and there are some issues with the libav x11grab [23:08] <JEEB> ChocolateArmpits, as far as I remember the bufsize on DVDs indeed is small like that, and the maxrate much bigger [23:09] <JEEB> welcome to 1990s [23:09] <JEEB> memory was Expensive [23:09] <wald0> btw, what is exactly the -intra parameter? i noted that in a comment too, waiting for a newer version of ffmpeg, seems like this option is needed for NLE ? [23:09] <JEEB> no idea what that is you're talking about, I know of no such parameter in either x264 or ffmpeg [23:09] <JEEB> that said, if you need all-intra you just set a keyint of 1 :P [23:10] <JEEB> (-g in ffmpeg) [23:10] <average> llogan: well well.. I think it's totally doable, but you have to agree it's not trivial [23:10] <ChocolateArmpits> well yeah, but then how is the relationship explained then? dvd buffer works at higher rate than one per second? [23:10] <wald0> and what this does exactly? sorry i didnt understand what is intra for [23:10] <average> especially when you put the word "static" in the mix [23:10] <JEEB> ChocolateArmpits, yes -- there's less than one second's worth of video in the buffer (if maxrate is kept) [23:11] <JEEB> wald0, intra = within, inter = between. [23:11] <llogan> average: copy 'n paste seems fairly trivial to me [23:11] <JEEB> in other words, intra compression is within one picture only, no other pictures needed [23:11] <llogan> average: ...but of course i don't know many details of what you're trying to achieve [23:11] <JEEB> inter compression is basing a picture on previous coded pictures, for example [23:11] <JEEB> in general intra-only is not something you want :P [23:11] <ChocolateArmpits> intra-only is good for editing purposes [23:12] <wald0> i see [23:12] <wald0> that makes sense for NLE, where replay the video frame-to-frame is very needed [23:12] <JEEB> ChocolateArmpits, basically you could enjoy more data transmission speeds back then than having a big buffer. That is also why DVDs suck quality-wise in the end :P [23:12] <wald0> or playing in reverse mode [23:15] <ChocolateArmpits> thanks for the info, JEEB, maybe you'll also know minimum GOP suggested for dvd titles/menus ? I know intra only is not supported, having no B-frames is also ill advised, so that should make IBP the minimum GOP for playback ? [23:15] <ChocolateArmpits> that is a 3-frame gop [23:15] <JEEB> no idea, I don't have the DVD specs [23:17] <ChocolateArmpits> well anyways, thanks for the info [23:17] <JEEB> I think blu-ray was the first digital media where you could actually get good quality within the limits of the format [23:18] <JEEB> DVD was made in a completely different world, and it shows by now :) [23:21] <Zeranoe> Can FFmpeg be used to normalize audio? Doing some research pointed to no, but they were all pretty old posts [23:22] <viric> Zeranoe: do you mean setting the gain flag? [23:22] <viric> or guessing the gain? [23:23] <Zeranoe> viric: Guessing [23:23] <viric> you can use the volumedetect filter [23:23] <viric> that will give you an idea of the gain [23:29] <Zeranoe> viric: So if mean is -34.7 and max is 7.3, i can gain by 7.3. There seems to be a huge gap though, between -34.7 and -7.3, which means it will probably still be quite... [23:29] <Zeranoe> quiet [23:29] <viric> you get stats about how many frames get out, by different proposed gains [23:30] <Zeranoe> hmm [23:30] <viric> well, you maybe not. I do. I don't know what version do you use [23:31] <Zeranoe> viric: So 'histogram_10db: 280' means 280 samples will be clipped if i gain by 10? [23:31] <viric> right [23:32] <viric> not samples; frames, I think [23:32] <viric> ah no. Samples, right. [23:33] <Zeranoe> i wish i knew if that was a lot lol. im not archiving this, but i also dont want to gain by too much so its unplayable [23:34] <Zeranoe> there isnt any sort of way to boost the low samples and keep the loud ones where they are? [23:39] <FunkyELF> Handbrake produces m4v files but I want mp4.... is there an actual difference in container? I know m4a and mp4 are exactly the same and you can just rename the file. [23:40] <FunkyELF> I'm using ffmpeg -i some_file.m4v -c copy some_file.mp4 [23:40] <sacarasc> mv blah.m4v blah.mp4 [23:41] <FunkyELF> sacarasc: okay.... wasn't sure if it was actually a different container [23:42] <FunkyELF> sacarasc: the result is different file sizes when I run it through ffmpeg again. Does that make sense if I used -c a? [23:42] <sacarasc> Very different? Or just slightly? [23:43] <FunkyELF> 3440192 vs. 3440932 [23:43] <sacarasc> If you ran it through 10 more times, you'd probably get 10 different sizes. [23:44] <FunkyELF> sacarasc: even on the same version of ffmpeg? [23:44] <sacarasc> Yes. [23:44] <FunkyELF> how is that? is it not deterministic? isn't the audio and video streams exactly the same when you use -c copy [00:00] --- Fri Jan 3 2014

1 0

ffmpeg-devel.log.20140101
by burek 02 Jan '14

02 Jan '14

[00:06] Action: ubitux hope one debug day will be enough [00:06] <ubitux> but it probably wont :( [00:46] <BBB> Daemon404: that was my suggestion [00:47] <BBB> Daemon404: but I suppose the person is way too special to use something straightforward as threads [00:51] <JEEB> well, you know non-blocking is hip now [00:51] <JEEB> javascript and all that jazz [00:52] <nevcairiel> such terms dont even make much sense in such a concept, how many interrupt points does he want, one check every macroblock? :p [00:52] <llogan> JEEB: jazz...damn it. you got Kenny G's "Song Bird" stuck in my head [00:53] <Daemon404> nevcairiel: the solution is for ffmpeg to write its own userland scheduler [00:53] <Daemon404> and use that [00:53] <Daemon404> duh. [01:27] <wm4> trollol [01:56] <iive> i've missed the discussion, but generally, nobody is born as skilled programmer. If he is ready to learn he would likely stick around. If not, removing stuff is the easiest task. [02:56] <cone-90> ffmpeg.git 03Michael Niedermayer 07master:46b06bd894f0: avcodec/mpegvideo_enc: implement frame skip score normalization [03:05] <cone-90> ffmpeg.git 03Hendrik Leppkes 07master:8abdf46b3805: dxva2_vc1: set refdist value according to spec [03:05] <cone-90> ffmpeg.git 03Hendrik Leppkes 07master:3021d1be9ef1: dxva2_vc1: set bfraction in slice info according to spec [03:05] <cone-90> ffmpeg.git 03Hendrik Leppkes 07master:e1facd3f8198: dxva2_vc1: set PQUANT as described by the 2010 spec update [03:05] <cone-90> ffmpeg.git 03Hendrik Leppkes 07master:719f1ce5fb41: vc1: set chromaformat = 1 for simple/main profile [03:05] <cone-90> ffmpeg.git 03Hendrik Leppkes 07master:c5562890c7c3: dxva2_vc1: fix signaling of intensity compensation values [03:05] <cone-90> ffmpeg.git 03Hendrik Leppkes 07master:3d8eeea62009: dxva2_vc1: signal skipped p frames [03:40] <cone-90> ffmpeg.git 03Michael Niedermayer 07master:07728a111583: avcodec/binkaudio: clear padding area of packet_buffer [13:06] <cone-427> ffmpeg.git 03Werner Robitza 07master:704c217ff32f: ffplay: add documentation for "s" key [13:06] <cone-427> ffmpeg.git 03Stefano Sabatini 07master:8c8c3ca91e8c: doc/protocols/rtsp, lavf/rtsp: fix/extend options documentation [13:06] <cone-427> ffmpeg.git 03Stefano Sabatini 07master:0b1cfc4f28d8: configure: bump year number 2013 -> 2014 [15:42] <cone-427> ffmpeg.git 03Michael Niedermayer 07master:e6364ea26df9: avcodec/vqavideo: use av_mallocz() for decode_buffer() [15:44] <ubitux> i don't get it, isn't "pandn a, b" = "a & ~b"? [15:44] <ubitux> "a &= ~b" even [15:48] <ubitux> i have m6=0xffffffffffffffffffffffffffffffff and m2=0xffffffff00ffff0000ff000000000000; pandn m6,m2 leads to m6=0 [15:50] <ubitux> shouldn't m6 be ~m2? [15:52] <ubitux> aah shit it's "pandn a, b" = "~a & b" [16:44] <nevcairiel> oh boy now he really posted this weird idea to the devel list, in before trollfest [16:48] <wm4> which? [16:48] <wm4> passing AVFrame to muxers? [16:48] <nevcairiel> that is weird too [16:48] <nevcairiel> but i'm referring to "Allow interrupt callback for AVCodecContext" [16:50] <BBB> ubitux: https://github.com/rbultje/ffmpeg/commits/vp9-simd first compiling version of full idct32x32 [16:50] <BBB> ubitux: doesn't work yet (likely?) but for some first fun [16:50] <ubitux> ah! [16:50] <BBB> not too many register shuffles if you ask me [16:50] <ubitux> you're going to finish it before me haha [16:50] <BBB> we'll see, I have to write all the subforms still, plus debug it [16:50] <BBB> I think we'll finish around the same time [16:50] <BBB> then we can blog about it :) [16:51] <BBB> trollfest on ml [16:51] <BBB> where [16:51] <BBB> wanna see [16:52] <nevcairiel> nothing yet [16:52] <nevcairiel> unless you want to start [16:52] <nevcairiel> but i expect it to start soon [16:53] <BBB> oo can't resist [16:53] <wm4> I don't see such a thread [16:53] <nevcairiel> BBB found it and responded [16:55] <wm4> hm what would be more useful than this stuff would be getting correct key frame flags, or the ability to extract timestamps without decoding [16:55] <nevcairiel> how is that even related [16:56] <cone-427> ffmpeg.git 03Michael Niedermayer 07master:06bb1de1c78a: avformat/vqf: check avio_read()s return value more completely [16:56] <wm4> depends on his application; e.g. if he just wants to make sure he's seeking to the right position [16:57] <nevcairiel> he wants to cancel decoding when he needs to seek, so his application is 5ms faster [16:58] <BBB> he wants the wrong thing [16:58] <BBB> he should want "fast seeking" [16:58] <BBB> he wants a particular solution that may or may not be related to the problem or be the best or easiest solution to the problem [17:53] <BBB> ok time to start debugging asm, see you guys in a few days :-p [19:35] <Daemon404> BBB: pretty sure the solution is a bad one regardless [21:29] <ubitux> aah i think i get my bug [21:30] <ubitux> BBB: any trick to "sign unpack"? [21:37] <ubitux> (basically unpack signed byte to signed words) [21:45] <ubitux> ah, maybe i should simply use a cmpgt as second unpack operand [21:45] <BBB> pmovsxbw? [21:46] <BBB> that's sse4 though [21:46] <ubitux> oh [21:46] <ubitux> mmh [21:46] <ubitux> i think i'm sse right now [21:46] <BBB> ssse3 [21:47] <ubitux> ah? [21:47] <Daemon404> not like anything precludes you from having a templated ver for both [21:47] <BBB> you can emulate it using pcmpgtw zero, yourthing and then use that to punpckbw as you said [21:47] <BBB> yeah I think in vp8 we did the templatization and had a ssse3 and a sse4 version or so [21:48] <Daemon404> i know sse2 is pretty much 100% available [21:48] <ubitux> i was thinking punpck*bw(x, pcmpgtb(x, 128)) [21:48] <Daemon404> no idea about ssse3 [21:48] <ubitux> BBB: what ssse3 am i using? [21:48] <BBB> dunno [21:48] <ubitux> i don't have any pshuf* [21:48] <BBB> didn't look at your code, maybe it's sse2 [21:48] <ubitux> i think it's even sse [21:48] <ubitux> but i might be wrong [21:49] <BBB> sse is foat only [21:49] <BBB> float [21:49] <BBB> assuming you use punpcklbw or so, you're sse2 [21:49] <ubitux> ah, then i'm wrong :) [21:49] <ubitux> ok [21:49] <BBB> anyway, punpcklbw(data, pcmpgtb(zero, data)) should work [21:49] <BBB> not sure why you'd want 128 instead of zero? [21:50] <ubitux> isn't the comparison unsigned? [21:53] <iive> in this case adding 128 unpacking and substracting 128 may be faster. [21:53] <iive> and you'd use same constant twice. [21:53] <iive> well, not really. [21:54] <ubitux> (actually pcmpgtb(x, 127)) [21:54] <ubitux> and ~ it [21:54] <ubitux> ah, no, without the ~ [21:59] <BBB> ubitux: no pcmp is signed [22:00] <ubitux> O_O [22:00] <ubitux> how the hell my masks are working.. [22:03] <BBB> see e.g. http://software.intel.com/sites/products/documentation/doclib/iss/2013/comp… [22:03] <BBB> pcmpgtb: "Compares the 16 signed 8-bit integers in a and the 16 signed 8-bit integers in b for greater than." [22:12] <ubitux> ok [22:29] <ubitux> one cool thing with lpf is that i don't have any reg swap [22:29] <ubitux> it simplifies the debug so much... :D [22:53] <cone-427> ffmpeg.git 03Michael Niedermayer 07master:e9a26dc5bf66: avformat/mux: fix "no tag found" check in validate_codec_tag() [23:36] <ubitux> BBB: according to intel man, punpck* are in sse [23:36] <ubitux> but maybe i'm reading it wrong [23:36] <BBB> http://cs.fit.edu/~mmahoney/cse3101/nasmdocb.html [23:36] <BBB> says willamette, sse2 [23:37] <nevcairiel> punpck are in mmx, but you need sse2 to use them on xmm regs [23:37] <ubitux> ah [23:37] <ubitux> ok [00:00] --- Thu Jan 2 2014

1 0

ffmpeg.log.20140101
by burek 02 Jan '14

02 Jan '14

[00:07] <relaxed> there is a volume filter [00:07] <DeadSix27> [AVFilterGraph @ 00000000027203c0] No such filter: 'volume=1.5' [00:09] <relaxed> if `ffmpeg -filters 2>&1|grep volum` doesn't list it your ffmpeg is too old [00:10] <DeadSix27> no linux m8 [00:10] <DeadSix27> T.. volume A->A Change input volume. [00:11] <ubitux> bad escaping [00:11] <DeadSix27> oh was due to me using ' [00:11] <DeadSix27> yep. [00:11] <DeadSix27> Cannot connect audio filter to non audio input [00:11] <DeadSix27> Error opening filters! [00:11] <DeadSix27> mp3 isnt audio? [00:11] <ubitux> -af? [00:11] <DeadSix27> right, tried both and didnt change vf back to af. [00:11] <DeadSix27> well thanks, works. [05:17] <MAF> I use ffmpeg for transcode RTMP to HLS, if my input audio wasn't aac, i can't do this, some devices doesn't support this format. any solution? [09:03] <Dongyancai> hello, I met a question in fullrange and tvrange, the command line and image source is at http://goo.gl/bn7xmZ (require rename), the padded area is light gray, how can I make the padding white or convert the image into a tv-range one ? [12:18] <DelphiWorld> hey people [12:18] <DelphiWorld> happy new year to all [13:01] <DelphiWorld> happy 2014 iive [13:02] <iamtakin1iteasy> hello, is there anyone having experience of streaming to twitch.tv over rtmp? is it possible to lower delay of 60 seconds there? [13:03] <iamtakin1iteasy> if not, could anyone suggest any other comaptible with ffmpeg straming services? [13:04] <iive> Happy New Year [13:06] <iamtakin1iteasy> also, is it possible to make an own straming service somehow? something taking a stream for a single slow source and re-streaming it multitude of clients? [13:07] <iamtakin1iteasy> does ffmpeg/other FOSS tools providing any tools for that? [13:07] <iamtakin1iteasy> err, s/tools/means/2 [13:09] <spaam> iive: happy new year mr ivan :) [13:12] <DelphiWorld> happy new year saste [13:13] <DelphiWorld> guys [13:13] <DelphiWorld> i have a apple tv [13:14] <DelphiWorld> i want to view my ogv video's [13:14] <DelphiWorld> is there anyway to view them without any converting (AKA) On the Fly transcoding? [14:22] <BeWilled> Hello and happy new year. My mind is about to blow off, I'm trying to pseudo-stream an FLV file with no luck, I gave up and my new strat consists in encoding that [14:23] <BeWilled> into mp4, however, does the .flv extension file has to be re-encoded into flv? or its only about chaning the "format" to mp4 [14:23] <BeWilled> into mp4* [14:23] <BeWilled> here is the ffprobe output http://pastebin.com/1QBFP6Lx [14:26] <BeWilled> I ask because transcoding iirc consumes cpu and my result in video quality loss [14:28] <ubitux> -c copy [14:29] <ubitux> 1.0 is very old btw [14:36] <BeWilled> ubitux: thank. guess I'll have to compile it. Wouldn't you know if there is a .deb around with the latest version? [14:36] <ubitux> maybe on deb-multimedia [14:36] <ubitux> but it should work in 1.0 [14:37] <ubitux> maybe 1.0 hasn't -c copy, in which case just -vcodec copy -acodec copy ... [17:48] <nfk> what's the lowest CPU I should be looking for if I want 4k H.265 decoding in software? [17:49] <nfk> 12 bpc obviously [17:50] <JEEB> ... that is sure one of the most derpy ways you could possibly say 8bit, unless you actually meant 12bit [17:50] <JEEB> (which is not supported yet, and is not in any of the current HEVC profiles) [17:50] <nfk> yes, i did mean 36 bit colours [17:51] <JEEB> come back after stuff's implemented then :P [17:51] <JEEB> if they ever will be [17:51] <nfk> i just want to be future-proof [17:51] <nfk> since if i'm going to assemble a new box, then it has to last at least half a decade at least [17:52] Action: nfk yawns [17:52] <nfk> not sure if i have had too much coffee or too little [17:52] <nfk> btw, happy new year, JEEB [17:52] <JEEB> the most useful thing for you then would have been "current performance", although at this point it most definitely would not be too useful since the implementations at this point aren't exactly speed-wise optimized (only asm in HEVC is for the CABAC, which is shared with H.264) [17:53] <JEEB> but since you want 12bit figures you can't really have "current performance, now can you?" [17:53] <JEEB> *"current performance" [17:53] <nfk> any guidelines then? [17:56] <clever> i would say just use hw decoding, but no hw decoder supports 12bit files [17:56] <nfk> not gonna happen [17:57] <nfk> by the time PC hardware decoding gets there, H.265 will be obsolete [17:57] <JEEB> nah, it will just never get there for end-users :P [17:57] <nfk> and i'm sure you people are already preparing to roll out H.265 this year, aren't you? [17:57] <JEEB> why are you trying so hard to get tsukkomi'd by me :P [17:59] <nfk> but i'm not [18:03] <nfk> fun, my usual PC part shop currently has EUR prices that look just like the LVL prices yesterday [18:03] <nfk> i wonder if they cocked up something or those are the actual prices [18:05] <nfk> i7 4770K is currently 80 EUR cheaper than on amazon.de [20:29] <stockholm> hi [20:30] <stockholm> my dvd player is old and a little retarded. it can play files from my usb stick, though [20:30] <stockholm> it plays avi files, for example. [20:30] <stockholm> now i have a file here: .mkv: EBML file, creator matroska [20:30] <stockholm> i think i can get better info on that, moment [20:32] <stockholm> http://paste.debian.net/73669/ [20:32] <sacarasc> ffmpeg -i input.mkv -c:a libmp3lame -b:a 192k -ac 2 -c:v mpeg4 -vtag XVID -b:v 800k output.avi [20:32] <stockholm> can i simply "copy" that into an avi file? [20:32] <sacarasc> You're probably looking for something like that. [20:32] <stockholm> ah, right, i do [20:33] <stockholm> thanks, i try that [21:47] <BoRiS> Seems theres a problem with the latest ffmpeg from git [21:53] <BoRiS> When building xine-lib from git, it errors out with [21:53] <BoRiS> ..libs/xineplug_decode_ff.so: version node not found for symbol av_fast_malloc@LIBAVCODEC_55 [21:54] <BoRiS> (Even though searching through the xine-lib source code, theres *no* reference to av_fast_malloc) [23:49] <hassen_> join #english [00:00] --- Thu Jan 2 2014

1 0