February 2015 - Ffmpeg-devel-irc

ffmpeg-devel.log.20150207
by burek 08 Feb '15

08 Feb '15

[00:32] <cone-888> ffmpeg.git 03Christophe Gisquet 07master:9dc45d1f4280: x86: lavc: share more constants [04:16] <cone-246> ffmpeg.git 03Michael Niedermayer 07master:f906982c9411: avcodec/h264_slice: Do not change frame_num after the first slice [04:16] <cone-246> ffmpeg.git 03Michael Niedermayer 07master:f111831ed611: avcodec/h264_slice: Check picture structure before setting the related fields [04:16] <cone-246> ffmpeg.git 03Michael Niedermayer 07master:38d5241b7f36: avcodec/h264_slice: ignore SAR changes in slices after the first [04:16] <cone-246> ffmpeg.git 03Michael Niedermayer 07master:2fd9ce92af43: avcodec/h264_slice: assert that reinit does not occur after the first slice [05:20] <cone-246> ffmpeg.git 03Alex Converse 07master:9295d10ea9d1: fate: Add a test for AAC ELD480. [05:20] <cone-246> ffmpeg.git 03Michael Niedermayer 07master:a40de9ac07fc: Merge commit '9295d10ea9d138462b7f67d16bf95ae9ca76aca6' [06:01] <cone-246> ffmpeg.git 03Anshul Maheshwari 07master:bf30161a8db1: avcodec/ccaption_dec: Added Roll up functionality [06:01] <cone-246> ffmpeg.git 03Anshul Maheshwari 07master:5647286e6778: avcodec/ccaption_dec: handle error from ass_sub api [06:28] <cone-246> ffmpeg.git 03Anshul Maheshwari 07master:f05efd42af37: avcodec/ccaption_dec: Added Debug logs [13:32] <cone-53> ffmpeg.git 03Michael Niedermayer 07master:c02b8c452786: avutil/pixfmt: Clarify the meaning of the alpha byte in RGB0 and similar formats [13:32] <cone-53> ffmpeg.git 03Michael Niedermayer 07master:088dd0fd4c8b: avutil/pixfmt: Clarify the meaning of the "alpha" bit in rgb555/bgr555 [13:32] <cone-53> ffmpeg.git 03Michael Niedermayer 07master:dbab5f59849a: avutil/pixfmt: Clarify the meaning of the alpha bits in rgb444 and similar formats [13:35] <wm4> michaelni: the RGB555 ones still say "most significant bit to 0", is that intended? [14:06] <cone-53> ffmpeg.git 03Michael Niedermayer 07master:be1cb1e63ae9: avutil/pixfmt: Clarify meaning of the A/X bit in RGB555 / remove wording about significant bit [15:13] <cone-53> ffmpeg.git 03Timothy Gu 07master:1de742145f58: pixdesc: Include more functions in FF_DISABLE_DEPRECATION_WARNINGS [15:35] <cone-53> ffmpeg.git 03Christophe Gisquet 07master:46e2afa4dcb5: wmalossless: reset lms_update [15:35] <cone-53> ffmpeg.git 03Christophe Gisquet 07master:691b7f5e9e29: lavc/lossless_audiodsp: revert various commits [16:19] <cone-53> ffmpeg.git 03Paul B Mahol 07master:ec8ec999f4a3: avcodec/twinvqdec: use init_get_bits8() [16:24] <cone-53> ffmpeg.git 03Carl Eugen Hoyos 07master:ac923ed47002: lavf/mlv: Only add streams that are supposed to contain frames. [17:54] <cone-53> ffmpeg.git 03Christophe Gisquet 07master:ed450d4acfc9: x86: lavc: share more constant through defines [18:12] <cone-53> ffmpeg.git 03Christophe Gisquet 07master:97996eff4f26: hevc/sao: do in-place band filtering when possible [18:36] <cone-53> ffmpeg.git 03Paul B Mahol 07master:134e8c73eaaf: avcodec/faxcompr: use init_get_bits8() [19:31] <michaelni> kurosu, did you see: http://article.gmane.org/gmane.comp.video.ffmpeg.cvs/86347 ? [19:34] <kurosu> no I hadn't [19:35] <kurosu> I would expect a crash if 1) somehow there was an incorrect alignment 2) the buffers were large enough [19:35] <kurosu> for 1), the check "if (((intptr_t)dst | (intptr_t)src | stride_dst | stride_src) & 15)" would seem sufficient [19:35] <kurosu> but maybe setting it to "if (1)" would help if that's the issue [19:36] <kurosu> for 2), dst should be large enough as it is the edge_emu buffer [19:37] <kurosu> unless icc decides to rearrange the buffers? [19:37] <kurosu> src should be the image, which would crash if not properly aligned, but it's weird [19:41] <michaelni> kurosu, carl posted gdb output in http://thread.gmane.org/gmane.comp.video.ffmpeg.cvs/86168/focus=86347 [19:41] <kurosu> at worst the patch can be reverted [19:41] <kurosu> oh ok [19:44] <kurosu> huh, why does it generate movdqa/u for AV_COPY64 whose code clearly uses movq [19:44] <kurosu> over-eager optimization by icc? [19:50] <nevcairiel> icc is evil like that sometimes, personally i would just call it broken and move on, but some people are more stubborn like that [19:51] <michaelni> maybe it generates that from the C code and not the asm, is the asm used for icc ? [19:55] <kurosu> well yeah, but is that code so evil that it decides to just miscompiles that one ? I would expect several such places to fail [20:02] <michaelni> i dont understand why icc is doing that but shouldnt it be AV_COPY64U() ? [20:11] <kurosu> does it change anything for x86 to push icc into generating this? but yes [20:11] <jamrial> that can't be the simd version of AV_COPY64() from x86/intredwrite.h [20:11] <jamrial> x86_64 enables HAVE_FAST_64BIT [20:11] <jamrial> and that implementation is under a "#if !HAVE_FAST_64BIT && defined(__MMX__)" check [20:11] <jamrial> it must be icc optimizing the c version [20:12] <kurosu> what platform distinguishes unaligned and aligned 64bits copies [20:12] <kurosu> bbl [20:52] <cone-53> ffmpeg.git 03Christophe Gisquet 07master:626d6184ce74: x86: lavc/hevc_mc: fix comments [21:00] <cone-53> ffmpeg.git 03Hendrik Leppkes 07master:8029af586fd5: dxva2_hevc: properly fill the scaling list structure [21:00] <cone-53> ffmpeg.git 03Michael Niedermayer 07master:89c7332b3825: Merge commit '8029af586fd5a5f2a0803001f9eff386e5545fe2' [21:07] <cone-53> ffmpeg.git 03Luca Barbato 07master:e352520e3ed7: oma: Report a timestamp [21:07] <cone-53> ffmpeg.git 03Michael Niedermayer 07master:7474ea74952b: Merge commit 'e352520e3ed7f08f19e63cd60e95da6bb6f037c1' [21:38] <cone-53> ffmpeg.git 03Michael Niedermayer 07master:7c2fa13df9a6: avformat/omadec: only compute timestamps based on bitrate if its set [21:38] <cone-53> ffmpeg.git 03Michael Niedermayer 07master:29fd3032cc96: avformat/omadec: Subtract headersize in timestamp calculation [00:00] --- Sun Feb 8 2015

1 0

ffmpeg.log.20150207
by burek 08 Feb '15

08 Feb '15

[00:12] <llogan> hotwings: possibly the concat demuxer [00:12] <llogan> http://ffmpeg.org/ffmpeg-formats.html#concat [00:12] <llogan> https://trac.ffmpeg.org/wiki/Concatenate [00:38] <konbon> Howdy. Is it possible to encode flac to mp3 via ssh using ffmpeg? [00:42] <c_14> Why would that be a problem? [01:00] <konbon> Not sure if it's even possible, tbh. [01:00] <konbon> Was wondering before I install it on my server. [01:00] <konbon> I'll find some tuts and see if it works out. [01:09] <c_14> ffmpeg -i flac out.mp3 [01:10] <c_14> https://trac.ffmpeg.org/wiki/Encode/MP3 [01:10] <c_14> have fun [01:17] <konbon> ty sir [01:29] <hotwings> llogan - it looks like concat works for my needs, thanks [01:31] <hotwings> konbon - ffmpeg can be used as command line so anything you can do from the/a gui should be possible on the command line [01:31] <konbon> oh cool [01:31] <konbon> never really used it, tbh [01:32] <konbon> Just encoded my backed up music from flac to mp3 just to save some space. [01:32] <konbon> 1TB can go by easily these days [01:33] <alyawn> join /#ffmpeg-devel [01:33] <alyawn> lol... long day [01:50] <konbon> pc decided to die for a bit [01:50] <konbon> back now [01:52] <heffer2k02> when writing an h264 stream, should i make a call to av_write_frame for each of the SPS and PPS? Or should they just be included along with the data of the first IDR? [02:10] <Seb_Sob> hello, anyone here? [02:11] <klaxa> no [02:11] <klaxa> (yes) [02:11] <c_14> ((no)) [02:11] <Seb_Sob> i have a problem concating 2 mp4 files [02:11] <Seb_Sob> could anyone help ? [02:12] <Seb_Sob> its in android though [02:13] <klaxa> well can you state your problem more precise? [02:14] <klaxa> "i have a problem" is pretty general [02:14] <klaxa> *precisely even [02:15] <Seb_Sob> http://pastebin.com/s1HqGZ88 [02:16] <Seb_Sob> well the output is just 0bytes so there must be something wrong [02:17] <klaxa> can you get ffmpeg's output? [02:17] <klaxa> (stderr) [02:17] <Seb_Sob> one mp4 is just a cropped mp4 of a video on my phone, the other one is an mp4 made of a frame image and layed with music [02:17] <llogan> scripts are not supported here. we need the actual command and the complete console output [02:17] <Seb_Sob> oh okay [02:18] <Seb_Sob> no i don't have the ffmpeg's output :s [02:18] <Seb_Sob> it doesn't generate an error, it just produces an mp4 with 0 bytes [02:19] <llogan> then we are at an impasse [02:19] <Seb_Sob> but i have another question than: what parameters must be equal for both mp4's in order to concat correctyle? like frame size, etc.. [02:20] <Seb_Sob> because the documentaion says: "As you can infer from this example, multiple types of input are supported, and anything readable by ffmpeg should work. The inputs have to be of the same frame size, and a handful of other attributes have to match." [02:20] <llogan> http://ffmpeg.org/ffmpeg-filters.html#concat [02:21] <Seb_Sob> i could try unsafe mode [02:23] <llogan> you must now be referring to the concat demuxer, although your script was using the concat filter [02:23] <Seb_Sob> yeah im confused [02:23] <Seb_Sob> i have both mp4 files with audio so i should use the demuxer? [02:26] <relaxed> Seb_Sob: https://trac.ffmpeg.org/wiki/Concatenate [02:26] <Seb_Sob> i was using this : http://i.gyazo.com/f3db1db0c6ab5a38b24976c5f640fda9.png [02:26] <llogan> that's the concat filter [02:27] <llogan> it required re-encoding, but sometimes you need to re-encode [02:27] <relaxed> MP4Box -cat 1.mp4 -cat 2.mp4 -new combined.mp4 [02:27] <Seb_Sob> thats mp4Box [02:27] <Seb_Sob> don't have that on android [02:28] <relaxed> oh [02:28] <llogan> give the demuxer a try. if it doesn't work then use the concat filter [02:28] <Seb_Sob> im using the ffmpeg4android_lib [02:28] <Seb_Sob> ok [02:28] <Seb_Sob> but the demuxer, is it possible without first writing to a file? [02:28] <llogan> i don't know what ffmpeg4android_lib is, but 3rd party stuff is not supported here [02:29] <Seb_Sob> oh thought it was official [02:29] <Seb_Sob> its basicly so that i can perform commands on android [02:29] <Seb_Sob> the syntax is the same [02:30] <relaxed> is it just ffmpeg compiled for android? [02:30] <Seb_Sob> yeah kind of [02:30] <Seb_Sob> im writing a mobile app, i need do crop vids and concat them [02:31] <Seb_Sob> the first part works great, but its the concat that drives me crazy :( [02:31] <Seb_Sob> i don't know how to write it to a file.txt [02:31] <Seb_Sob> so im looking for a method ffmpeg -f concat -i mylist.txt -c copy output without the .txt part [02:32] <Seb_Sob> is it possible? [02:32] <llogan> if you need to crop them too, then use the crop and concat filters [02:33] <llogan> about the "without the .txt part", yes, probably. http://trac.ffmpeg.org/wiki/Concatenate#Instructions [02:33] <relaxed> Seb_Sob: my android phone has echo and cat [02:34] <Seb_Sob> In case you wonder: i crop my video, that gives me part1.mp4, then from that part1.mp4 i take the last frame and i create a 7second .mp4 layered with an mp3 audio and a .png watermark. [02:34] <relaxed> and /system/bin/sh [02:34] <Seb_Sob> ah ok [02:34] <relaxed> be sure and avoid bashisms, though [02:35] <Seb_Sob> what exacly is the difference between stream level and file level? [02:36] <Seb_Sob> i have two .mp4 files, is that stream level? [02:38] <Seb_Sob> btw i tried : ffmpeg -i input1.mp4 -c copy -bsf:v h264_mp4toannexb -f mpegts intermediate1.ts [02:38] <Seb_Sob> but then intermediate1.ts gives me a black screen, with only audio... [02:39] <relaxed> what about when you mux back to mp4? [02:40] <Seb_Sob> when i concat the 2 .ts files it gives me a 0bytes .mp4 [02:40] <Seb_Sob> ffmpeg -i "concat:intermediate1.ts|intermediate2.ts" -c copy -bsf:a aac_adtstoasc output.mp4 [02:43] <relaxed> ffmpeg version? [02:44] <Seb_Sob> no idea :s [02:45] <Seb_Sob> this is what i use: http://androidwarzone.blogspot.be/2011/12/ffmpeg4android.html [02:48] <relaxed> I would see if it works on your desktop, and if it does you'll have to seek support from that guy on why his software doesn't [02:49] <Seb_Sob> yeah probably [02:49] <Seb_Sob> i'll try that tomorrow, i need some sleep now, thanks for the help [04:24] <techtopia> ffmpeg -t 60 -i test.mkv -c copy sample.mkv [04:24] <techtopia> i use this command to cut a 60 second segment out of a video [04:24] <techtopia> is there someway i can tell ffmpeg to cut it out after an hour into the video [04:24] <techtopia> instead of the first 60 seconds [04:34] <grepper> techtopia: combine with -ss 3600 [04:35] <grepper> or -ss 00:01:00 [05:04] <techtopia> thanks grepper [05:54] <grepper> techtopia: if you have a recent enough ffmpeg it will do a fast accurate seek with the -ss AFTER the -i FILE [05:54] <grepper> afaik [05:55] <grepper> oh, sorry, maybe that really means that the -ss BEFORE the -i FILE will be accurate [05:56] Action: grepper should have kept quiet about that [05:58] <grepper> yep, that's it, just did a test [05:59] <grepper> time ffmpeg -ss 00:01:00 -i VTS_07_1.VOB -frames 1 -y before-i.jpg [05:59] <grepper> real 0m0.026s [05:59] <grepper> time ffmpeg -i VIDEO_TS/VTS_07_1.VOB -ss 00:01:00 -frames 1 -y after-i.jpg [05:59] <grepper> real 0m2.610s [06:00] <grepper> $ diff -s before-i.jpg after-i.jpg [06:00] <grepper> Files before-i.jpg and after-i.jpg are identical [10:56] <fling> How to mix two files together? [10:56] <fling> Both has video and audio [10:57] <fling> I want to have two video and two audio in the output [10:57] <fling> or atleast a single video and two audios [10:57] Action: fling can't figure out how to do so with -map when two files with multiple streams involved [10:59] <fling> Ok, looks like I figured it out, thanks to this page -> http://ffmpeg.org/ffmpeg.html#Advanced-options [11:03] <fling> How to set language for a stream? [11:06] <fling> &with -metadata [11:14] <fling> subtitles time! [11:14] <fling> How can I add regions?& [12:06] <gdrc> hello. [12:06] <mcnesium> on osx i am trying to use ffmpeg v2.5.3 with the parameter -pass 2 to check out quality improvements of two-pass encoding, but it only replies Additional information: rc_twopass_stats_in requires at least two packets [12:07] <mcnesium> this is when trying to encode .webm video from a set of jpeg files [12:08] <mcnesium> heres the full command qmin=0;qmax=50;ffmpeg -i test/%09d.jpg -r 24 -quality best -cpu-used 0 -threads 2 -qmin $qmin -qmax $qmax -pass 2 -an -vf scale=480:480 -y test_$qmin-$qmax.webm; [12:08] <mcnesium> can i use that pass 2 parameter here at all? [12:12] <Mavrik> um, you do know that you have to first run with -pass 1 to generate the info file right? [13:07] <techtopia> hey guy's [13:08] <techtopia> trying to download some videos off some servers, which both have the same key file [13:08] <techtopia> some of their servers are regular http connections, some use port 443 and are ssl connections [13:09] <techtopia> ffmpeg has no problem with the regular connections but when i tell it to download from the one's using ssl i get 403 errors [13:09] <techtopia> tried throwing the open ssl dll's in the ffmpeg folder but that didn't work [13:09] <techtopia> anyone got any ideas [13:54] <smurdnomad> hi all [13:55] <smurdnomad> i'm trying to put a watermark on a little clip with -filter_complex filter [13:56] <smurdnomad> all works fine with landscape videos, but when I use it with portrait video, the output video is in landscape mode ... someone already has this issue ? [13:57] <smurdnomad> ffmpeg -i bb.mov -i watermark.png -filter_complex overlay aa.mp4 [14:03] <techtopia> use -s resolution [14:03] <techtopia> for what ever your video resolution is [14:04] <smurdnomad> can you explain a little bit with an example, i'm new in ffmpeg :) [14:04] <smurdnomad> (and french ^^) [14:07] <techtopia> like ffmpeg -i bb.mpv -iwatermark.png -filter_complex overlay -s 800x1600 -aa.mp4 [14:07] <techtopia> manually specify the output resolution you want [14:07] <Seb_Sob> hey all, is it possible to create a zoom in animation for a specific duration, start zoom level, end zoom level? [14:09] <smurdnomad> just tried your solution techtopia, but the video is still in landscape but is resolution portrait (I don't know if you see what I mean ...) [14:10] <techtopia> my resolution was just an example [14:10] <techtopia> you need to set whatever the resolution of the output device is going to be [14:10] <techtopia> you also need to be feeding it a portrait video if you want a portrait output [14:11] <smurdnomad> yes I understand, but the video is not really in portrait mode like the input one (it looks like a portrait mode but the image is just flattened from the edge ....) [14:12] <smurdnomad> but still rotated [14:42] <Seb_Sob> zoompan: -vf "zoompan=z='min(zoom+0.0015,1.5)':d=125", are the quotes needed? , does anyone know what the d=125 stand for ? Thanks [14:47] <Seb_Sob> because this doesn't work: [14:47] <Seb_Sob> "ffmpeg -loop 1 -i " + imagePath + " -vf zoompan=z=min(zoom+0.0015,1.5):d=125 -c:v libx264 -r 30 -s 640x360 -t 7 -pix_fmt yuv420p " + outputFileName; [14:48] <Seb_Sob> anyone knows? [15:13] <gdrc> Seb_Sob: are you on android? [18:28] <|some1l> hi [18:28] <|some1l> I use .avi container for output. I noticed ffmpeg doesn't write duplicate frames into file, it uses 0byte framesize. Is there an option to force to write duplicate frames into output? [18:39] <AlmondPotato> Even with -vf select='between(t\,141\,764)', it's still outputting the first bit of video [20:05] <gdrc> I'm having trouble merging two video: https://gist.github.com/gderaco/a429c2c384fb24da6763 - stackoverflow doesn't help [20:32] <gdrc> anyone guys? [21:10] <django> Is there a way I can detect dupe frames in linux by command? [00:00] --- Sun Feb 8 2015

1 0

ffmpeg-devel.log.20150206
by burek 07 Feb '15

07 Feb '15

[00:07] <compn> what [00:08] <compn> lou and me review the ml often [00:11] <compn> as evidenced by there only being 6 spams in the queue right now, that i just cleared :P [00:11] <compn> lou looks at it more than i do :) [00:11] <compn> gmx has some of the shittier servers [00:11] <compn> so its not his fault [00:12] <compn> michaelni has trouble with gmx too [00:18] <michaelni> wm4, jamrial posted a patch with the suggested change [00:20] <wm4> michaelni: yeah, that's exactly what I tried and what worked [00:46] <cone-125> ffmpeg.git 03Michael Niedermayer 07master:3b4ffba3af96: avcodec/x86/lossless_audiodsp: Make scalarproduct_and_madd_int16 prototypes more similar [00:46] <cone-125> ffmpeg.git 03Michael Niedermayer 07master:d41b66a1a216: avcodec/vp3: pass correct context to av_log() [01:47] <jamrial> michaelni: this is not enough. it's still broken on win64 because of xmm6+ being spilled during prologue but then not restored after the jump to the mmxext version [01:49] <jamrial> try running ./ffmpeg -i [CCCP\]_Mega_Weird_Audio_Test_new.mkv -map:a 0:23 -map 0:0 -vn -sn -f null - on a win64 machine with ssse3 (use -cpuflags -sse4.2 if the cpu supports it, to get around the weird init check) [01:56] Action: michaelni did not realize there was a sse4.2 function [01:58] <jamrial> there is none [01:58] <jamrial> but the check in lossless_audiodsp_init.c is weird and checks for sse4.2 as well [01:59] <michaelni> that explains why i missed it when testing my patch ... [02:01] <jamrial> it even checks for 3dnow for whatever reason. no CPU exists with ssse3 and 3dnow... [02:06] <jamrial> i think the best solution at this point is to duplicate the mmxext code in the ssse3 function [02:07] <jamrial> or making c wrappers, but that's going to be slower [02:34] <michaelni> ive implemented C wrapers as that seems alot more robust, also theres another bug in there [02:40] <michaelni> but if you think doing that in C is a speed problem we/i could look into changing it to asm but ATM i want to fix that crash [02:44] <cone-125> ffmpeg.git 03Michael Niedermayer 07master:f1214763af1a: avcodec/x86/lossless_audiodsp: Move order&8 fallback into C code [02:44] <cone-125> ffmpeg.git 03Michael Niedermayer 07master:a6c2c8fe3f07: Revert "avcodec/x86/lossless_audiodsp: Make scalarproduct_and_madd_int16 prototypes more similar" [02:46] <jamrial> i just wonder about the call overhead using c wrappers [02:47] <jamrial> a "test order, 8" "jnz .fallback" (where .fallback is a copy of the relevant mmxext code) is probably faster [03:09] <jamrial> michaelni: is http://pastebin.com/6S5S69EG ok as a simplified version of the tta patch? [03:12] <michaelni> iam not sure about the EXPLODE change being a good idea, rest LGTM [03:13] <jamrial> ok, will push without the check for explode [03:13] <michaelni> do we have a file that has a broken header CRC and otherwise decodes to something useful ? if so it could make sense to accept broken CRCs by default [03:16] <jamrial> i don't have any. but the header only contains channel amount, bitdepth, duration and samplerate [03:16] <jamrial> any of those being wrong (except maybe duration) is going to end up with an unplayable file [03:16] <michaelni> hmm [03:17] <michaelni> if you prefer the explode check then iam sure not against it [03:26] <cone-125> ffmpeg.git 03James Almer 07master:15a88468aecd: avformat/tta: only check for header and seek table crc if requested [04:39] <cone-125> ffmpeg.git 03Michael Niedermayer 07master:69aa79365c1e: avcodec/h264_ps: More completely check the bit depths [05:14] <jamrial> nevcairiel, Daemon404: any idea why http://fate.ffmpeg.org/log.cgi?time=20150206010917&log=compile&slot=x86_64-… is happening? [05:15] <jamrial> i know it's related to PIC, but no idea why it only affects msvc x64 and not mingw64 [07:53] <Timothy_Gu> michaelni: found something fishy in this merge: 7d82020fcb7f81fcbbd30b7546ba62af45f1a33c [07:54] <Timothy_Gu> michaelni: is there a reason why flv_data_packet() is not used at all while it is in 21e2dc9fb76bc34fcdedf32e4ce820d6cdf923fb [08:00] <kurosu> michaelni, regarding the scalar product, I remember discussing with you about that fallback [08:00] <kurosu> the issue was orders multiple of 8, in particular exactly 8 [08:01] <kurosu> the buffers are padded, so we can just do 16 [08:02] <kurosu> maybe the fallback to mmx is faster, but maybe just rounding the order to the nearest multiple of 8/16 depending on the instruction set is sufficient [08:20] <kurosu> http://pastebin.com/rLVyA64c <- something like that, generates the same crc on the cccp sample, and passes fate [08:20] <kurosu> i've modified the mmxext version, but it's really unnecessary [10:47] <cone-142> ffmpeg.git 03Paul B Mahol 07master:dc3c3758ce63: avformat/thp: check av_get_packet() for failure [11:11] <ubitux> AVRational av_guess_frame_rate(AVFormatContext *format, AVStream *st, AVFrame *frame) [11:11] <ubitux> all of these should be const [11:12] <nevcairiel> ew, more evil uglyness for the broken OpenCL support [11:16] <durandal_1707> 2 files changed, 3141 insertions(+) [11:21] <ubitux> lol wtf is that [11:21] <nevcairiel> clew, just like glew [11:21] <nevcairiel> :d [11:22] Action: ubitux doesn't understand [11:23] <ubitux> here we go, opencl autodetect [11:23] <ubitux> dammit [11:25] <nevcairiel> ubitux: i think with this clew thing it doesnt have any link-time deps, so its all runtime-detect [11:25] <nevcairiel> which is one of its reasons for existing [11:28] <nevcairiel> but importing this lib into the code base is just wrong ... glew for example is distributed by distributions and could be an external dep, why couldnt clew be like that as well [11:29] <ubitux> yeah whatever [12:25] <wm4> man this opencl loader thing is full of nightmares [12:25] <wm4> a see of ifdefs [12:25] <wm4> *sea [12:26] <wm4> and even defining reserved identifiers [12:26] <wm4> because the guy who wrote it doesn't understand C [12:42] <michaelni> Timothy_Gu, ive sent a patch to fix that already, see "[FFmpeg-devel] [PATCH 2/2] avformat/flvdec: re enable flv_data_packet() and use AVMEDIA_TYPE_SUBTITLE" but IIRC theres no sample/testcase to test [12:58] <michaelni> kurosu, http://pastebin.com/rLVyA64c breaks cccp here, tested with -i \[CCCP\]_Mega_Weird_Audio_Test.mkv -map 0:23 test.wav [12:58] <michaelni> test.wav is just noise [13:43] <wm4> ubitux: was my mail addressed to ffmpeg-devel only, or did you remove libav-devel from your reply? [13:50] <ubitux> wm4: ah mmh i'm sorry, i just "replied" instead of "group reply" on the ffmpeg-devel version, that wasn't on purpose [13:50] <wm4> ok [13:50] <ubitux> but now that i think of it, i would have done the same on purpose [13:50] <ubitux> since i'm talking about ffmpeg internal [13:50] <wm4> I wasn't sure how this works (I wanted to avoid CCing one because that would be "offensive") [13:51] <ubitux> and mentioning it's done one way in ffmpeg will probably encourage the other side to do it differently [13:51] <ubitux> so better not talk about it [13:51] <wm4> lol [13:51] <wm4> no I don't want to play this kind of game [14:03] <wm4> does anyone have the file posted in this ticket? https://trac.ffmpeg.org/ticket/1730 [14:03] <wm4> the user uploaded it on some file hoster which looks dead now [14:04] <nevcairiel> 2 years is too long for such hosters :p [14:05] <nevcairiel> sometimes carl puts those files onto the sample ftp when he tests them [14:05] <wm4> oh it was here https://samples.mplayerhq.hu/ffmpeg-bugs/trac/ticket1730/FFMpeg_Bug_1730_cr… [14:05] <wm4> yes [14:05] <ubitux> http://ffmpeg.org/bugreports.html "Furthermore movie files uploaded to services like rapidshare or any other similar service will be ignored. We are not willing to spend our time fighting with this ridiculous, bloated and spam-filled crap." [14:05] <ubitux> but yeah, people do that :p [14:05] <wm4> this file doesn't play [14:05] <wm4> and predictably, crashes Libav [14:06] <wm4> because Libav doesn't handle this special case [14:06] <wm4> but seriously, an id3 tag on a mp4 file [14:22] <JEEBsv> > id3 in mp4 [14:22] <JEEBsv> just kick it out [14:26] <wm4> you can't [14:26] <wm4> ffmpeg reads id3v2 for every format [14:26] <mateo`> wm4: do we know which kind of muxer generate such file ? [14:27] <wm4> the file contains this: encoder : iTunes v4.8.0.32, QuickTime 6.5.2 [14:27] <wm4> though whether it added the id3 tag is questionable [14:27] <wm4> it probably was some shitty tagging tool [14:28] <mateo`> iTunes v4.[...] is pretty old, maybe they did a sneaky fix at some point :D [15:36] <cone-888> ffmpeg.git 03Michael Niedermayer 07master:31cc9c04ca38: avcodec/h264: Be more strict on rejecting pps_id changes [15:36] <cone-888> ffmpeg.git 03Michael Niedermayer 07master:6fafc62b0bd0: avcodec/h264: Be more strict on rejecting pps/sps changes [17:08] <cone-888> ffmpeg.git 03Paul B Mahol 07master:f968166439e4: avformat/rpl: check av_get_packet() for failure [17:36] <cone-888> ffmpeg.git 03Pierre Edouard Lepere 07master:a0d1300f7112: x86: hevc_mc: add AVX2 optimizations [17:50] -:#ffmpeg-devel- [freenode-info] if you're at a conference and other people are having trouble connecting, please mention it to staff: http://freenode.net/faq.shtml#gettinghelp [17:59] <kurosu> michaelni, ok and weird [17:59] <kurosu> I'll give it more time tomorrow [18:09] <cone-888> ffmpeg.git 03Christophe Gisquet 07master:5eedd36df108: x86: hevc_mc: use epel_hv 16-wide function [18:09] <cone-888> ffmpeg.git 03Mickaël Raulet 07master:bcb092511525: x86/hevc: use CLIPW macro when possible [18:10] <kurosu> bye [19:07] <cone-888> ffmpeg.git 03James Almer 07master:aea29a891ff1: x86/hevc_sao: fix loading of RIP address [19:13] <cone-888> ffmpeg.git 03Timothy Gu 07master:5081012eb88b: audioconvert: Add missing include for FF_API_AUDIO_CONVERT [19:14] <cone-888> ffmpeg.git 03Timothy Gu 07master:23659fdb812f: generate_wave_table: Add include for AVSampleFormat [19:14] <cone-888> ffmpeg.git 03Michael Niedermayer 07master:9bc7ee8a9874: avcodec/avfft: Add simple self test [20:45] <cone-888> ffmpeg.git 03Timothy Gu 07master:e66a18763841: img2dec: Remove dead code [20:45] <cone-888> ffmpeg.git 03Timothy Gu 07master:510b39c2137b: nutdec: Remove unused variables [21:31] <cone-888> ffmpeg.git 03James Almer 07master:383fddeec65f: x86/lossless_audiodsp: fix compilation with --disable-yasm [21:40] <cone-888> ffmpeg.git 03Mickaël Raulet 07master:6ecc3fd61225: x86/hevc_mc: use aligned loads [22:29] <cone-888> ffmpeg.git 03Michael Niedermayer 07master:a0640e63463e: avutil/opt: Fix types used to access AV_OPT_TYPE_PIXEL_FMT [22:29] <cone-888> ffmpeg.git 03Michael Niedermayer 07master:1750b45cdf74: avutil/opt: Fix type used to access AV_OPT_TYPE_SAMPLE_FMT [22:50] <heffer2k02> Hello. I'm trying to write an MP4 file from a sequence of h264 frames. It seems to write everything out correctly, but clearly I'm missing something. The error I get when doing ffmpeg -i file.mp4 is "Could not find codec parameters for stream 0 (Video: h264 (avc1 / 0x31637661), none, 1280x720, 4941 kb/s): unspecified pixel format". Now in my code I have set pix_fmt=AV_PIX_FMT_YUV420P, although I'm in the dark a bit here. Is the [22:50] <heffer2k02> complaint coming from failure to read stuff from the h264 stream? Or a misconfiguration in the container? [22:51] <heffer2k02> aaaand, I see I should be asking this in the other channel. [22:58] <thardin> . [22:58] <cone-888> ffmpeg.git 03Carl Eugen Hoyos 07master:d45fadb6df48: lavc/tscc: Make 32bit output opaque. [22:58] <cone-888> ffmpeg.git 03Michael Niedermayer 07master:bfb988b1fafd: Merge remote-tracking branch 'cehoyos/master' [00:00] --- Sat Feb 7 2015

1 0

ffmpeg.log.20150206
by burek 07 Feb '15

07 Feb '15

[00:50] <nathanpickles> I'm trying to combine two video files into 1 side by side video file. I'm trying to get them to playback in sync. http://hastebin.com/enovamacen.md [00:52] <c_14> The problem being? [00:53] <nathanpickles> c_14: So, the 2 videos were recorded at the same time. They are each about 40 seconds long. [00:55] <nathanpickles> c_14: I want to delay one video by a time delta given by the difference between the start time of both videos [00:59] <c_14> Either trim the front of the later video, or concat a short color=black in front of the earlier. [01:03] <nathanpickles> c_14: I have the creation times for each video (from ffmpeg -i) 2015-02-04 21:27:07 and 2015-02-04 21:27:11. How would I add color=black to the first 4 seconds to the second video? [01:05] <c_14> What are the dimensions of the second video? [01:05] <nathanpickles> c_14: http://hastebin.com/ukudeneyur.avrasm [01:06] <nathanpickles> c_14: Dimensions... one sec [01:07] <nathanpickles> c_14: 1920x1080 for the second video [01:07] <nathanpickles> the one on the right [01:08] <nathanpickles> c_14: Here are the details on each: http://hastebin.com/gugagafojo [01:08] <nathanpickles> 1280x720 for the first one [01:08] <c_14> `ffmpeg -i 20150204_132628_109.mp4 -i movie=VID_20150204_132628.mp4 -filter_complex "[0:v]scale=iw/2:ih/2,pad=2*iw:ih[left];color=black:duration=4:s=1920x1080[b];[b][1]concat=n=2:v=1:a=0,scale=iw/3:ih/3 [right]; [left][right] overlay=main_w/2:0 [out]" -b:v 768k Output2.mp4' [01:08] <c_14> ^should do it [01:10] <nathanpickles> movie=VID_20150204_132628.mp4: No such file or directory ? [01:10] <c_14> eh, oops [01:10] <c_14> get rid of the movie= [01:11] <nathanpickles> Output pad "default" with type video of the filter instance "Parsed_overlay_5" of overlay not connected to any destination [01:11] <nathanpickles> do i need to pu [left] ? [01:11] <c_14> I should pay more attention, add -map '[out]' somewhere before Output2.mp4 and after the filtergraph [01:12] <c_14> That should be it anyway. [01:12] <nathanpickles> It's good to know this is possible [01:12] <nathanpickles> I am trying to automat the process. [01:12] <nathanpickles> Is this the filter graph? "[0:v]scale=iw/2:ih/2,pad=2*iw:ih[left];color=black:duration=4:s=1920x1080[b];[b][1]concat=n=2:v=1:a=0,scale=iw/3:ih/3 [right]; [left][right] overlay=main_w/2:0 [out]" [01:12] <c_14> yep [01:13] <nathanpickles> got a couple errors: http://hastebin.com/itofaxotot.vbs [01:14] <nathanpickles> c_14: http://hastebin.com/itofaxotot [01:14] <nathanpickles> sorry [01:14] <c_14> 65536/65536 != 1/1 eh [01:15] Action: c_14 smells floats [01:16] <c_14> `ffmpeg -i 20150204_132628_109.mp4 -i VID_20150204_132628.mp4 -filter_complex "[0:v]scale=iw/2:ih/2,pad=2*iw:ih,setsar=1[left];color=black:duration=4:s=1920x1080[b];[1]setsar=1[n];[b][n]concat=n=2:v=1:a=0,scale=iw/3:ih/3 [right]; [left][right] overlay=main_w/2:0 [out]" -map '[out]' -b:v 768k Output2.mp4' [01:16] <c_14> Try that [01:31] <nathanpickles> c_14: pretty close, just not getting audio [01:31] <c_14> You wanted audio? [01:31] <c_14> From which file? [01:31] <c_14> Both? [01:31] <c_14> Mixed? [01:32] <nathanpickles> just the left file [01:32] <c_14> add a -map 0:a somewhere before Output2.mp4 and after the inputs [01:32] <nathanpickles> mixed could be interesting [01:35] <c_14> If you want mixed, add [0:a][1:a]amix[a] to the end of the filtergraph (after the [out], separated with a ';'), and -map '[a]' somewhere after the filtergraph and before the output file [01:39] <nathanpickles> c_14: Thank you very much! You have been super helpful [01:41] <c_14> np [06:45] <SBait> anyone used dshow inputs and any good atdiagnosing input erros avail for help? [06:47] <SBait> [NULL @ 0000000008659180] Unable to find a suitable output format for 'audio=Lin [06:47] <SBait> e in at rear panel (Blue) (R' [06:47] <SBait> audio=Line in at rear panel (Blue) (R: Invalid argument [06:48] <SBait> stream encodes fine if i process without the video, and the video process fine without the audio, but join the 2, and = dramas [06:52] <SBait> fmpeg -f dshow -i video="Timeleak HD Capture": -s pal -aspect 16:9 -vcodec libx264 -b:v 550000: audio="Line in at rear panel (Blue) (R" -acodec libvo_aacenc -af aresample = async = 1 -application lowdelay -ac 2 -f matroska -content_type matroska icecast://source:hackme@internalIP:port/stream.mkv [09:00] <Nopik> hi all [09:02] <Nopik> so I'm piping some audio to the ffserver, and it seems that the audio filters get ignored. This doesn't work (filter is not applied): ffmpeg -i tcp://127.0.0.1:5002 -af "compand=0.3|0.3:0.8|0.8:-90/-90|-70/-70|-60/-20|0/0:0.01:-5:-45:0.4" http://localhost:8090/feed.ffm (I also tried with -filter_complex), but then I change the output to -f mp3 - | ffmpeg -i [09:02] <Nopik> - http://localhost:8090/feed.ffm it works and filter is applied [09:03] <Nopik> so, at worst i can live with extra ffmpeg, but I'd prefer to get rid of it (it is live streaming over multiple channels, so my server is maxing out cpu already, adding dozen of 'empty' ffmpeg instances is definitely not helping [09:04] <Nopik> any idea how to optimize that? [09:24] <__julian> hi [09:25] <__julian> I do run a ffmpeg transcode from an mpeg2 rtsp stream to mpeg2 udp using ffmpeg command line tool. is there a way to dynamically overwrite aspect ratio without restarting ffmpeg? [10:48] <ariscop> are there any container formats that can store metadata with a pts? worst case i could use a subtitle format but would rather not [17:04] <applejack> how do I embed an srt file as a subtitle stream in an mkv container while converting an avi with video and audio to mkv ? [17:06] <applejack> ffmpeg -i video.avi -i sub.srt -c:v copy -c:a ac3 -c:s copy video.mkv <-- thinks the .srt file is an audio track type mp3 ?!?? [17:07] <applejack> however, I can merge it into the mkv with mkvmerge if I omit the srt from ffmpeg [17:08] <applejack> but that makes it a two step operation [17:15] <relaxed> applejack: ffmpeg version? [17:25] <applejack> 2.2.12 [17:37] <applejack> no matter what I try, the srt is treated as an mp3 [17:46] <relaxed> applejack: also, that version is fairly old. You might compile a more recent version or http://johnvansickle.com/ffmpeg/ [17:51] <relaxed> gcl_5cp: edit the h264 wiki and add it [17:51] <applejack> relaxed: http://pastie.org/9892543 [17:52] <relaxed> try again [17:52] <applejack> huh? [17:53] <relaxed> did you look at the link? [17:53] <applejack> yea [17:54] <relaxed> yeah what? It says there's nothing there. [17:54] <relaxed> hmm, now it's there...how odd [17:55] <relaxed> applejack: "file sub.srt" returns? [17:56] <relaxed> did you open sub.srt to see if it was text? [17:57] <applejack> : sub.srt: Little-endian UTF-16 Unicode text, with CRLF, CR line terminators [17:59] <applejack> relaxed : sub.srt is in utf-16, so it looks binary-ish [18:01] <relaxed> try, ffmpeg -i movie.avi -i sub.srt -map 0 -map 1:s -c:v copy -c:a copy -c:s copy out.mkv [18:02] <applejack> Stream map '1:s' matches no streams. [18:03] <applejack> hah [18:03] <relaxed> ok, ffmpeg -i movie.avi -f srt -i sub.srt -map 0 -map 1 -c copy out.mkv [18:04] <applejack> iconv utf16 -> utf8 sub.srt > subs.srt fixed it [18:04] <applejack> so ffmpeg don't like UTF16 [18:05] <relaxed> ah, I wonder if 2.5.3 has that problem? [18:06] <relaxed> ubitux: ^^ [18:06] <ubitux> it should work [18:06] <ubitux> -f srt should be omitted too [18:06] <applejack> even on 2.2.12 [18:06] <ubitux> can i get the sample? [18:07] <ubitux> it's a recent feature [18:07] <ubitux> utf16 is supported [18:07] <relaxed> well, I was trying to forse it since it was detected as an mp3 [18:07] <relaxed> force* [18:07] <applejack> ubitux: did you want the subtitle file? [18:07] <ubitux> yes please [18:09] <applejack> ubitux: http://www.mrhansen.id.au/sub.srt [18:11] <ubitux> works fine here [18:12] <applejack> 2.5.3? [18:12] <ubitux> yes [18:12] <applejack> figures,.... lol [18:13] <ubitux> http://pastie.org/pastes/9892584/text [18:13] <ubitux> 757e4659848ac36c4408bf58bc429f3e sub.srt [18:13] <ubitux> test.avi generated with ffmpeg -f lavfi -i testsrc=d=30 -y test.avi [18:14] <applejack> yea,. and notice: [srt @ 0x7fa535904fa0] UTF16 is automatically converted to UTF8, do not specify a character encoding [18:14] <applejack> i'll just iconv them [18:22] <ubitux> applejack: but... it works? [18:22] <ubitux> it's not an error [18:46] <applejack> ubitux: yea,. but I'm stuck with 2.2.12 [18:47] <ubitux> ah, ok [18:47] <ubitux> too bad for you :) [18:47] <applejack> yea [18:47] <ubitux> suse? [18:47] <applejack> gentoo [18:48] <ubitux> you have a masked one :p [18:48] <applejack> hard masked i think [18:50] <applejack> i put media-video/ffmpeg ** in package.accept_keywords, which got me 2.2.12, up from 1.2.11 [18:51] <JEEB> applejack, they're hard masked because of rare things. I wouldn't be surprised if you were all right if you went all the way to the latest :P [18:51] <ubitux> 2.2 is almost 1yr old too [18:52] <JEEB> basically, the chance of hitting something with 2.2->2.5 is pretty miniscule at this point [20:14] <Mista-D> JEEB: its in NAL, thanks anyways. ( a day later : ) [22:51] <heffer2k02> Hello. I'm trying to write an MP4 file from a sequence of h264 frames. It seems to write everything out correctly, but clearly I'm missing something. The error I get when doing ffmpeg -i file.mp4 is "Could not find codec parameters for stream 0 (Video: h264 (avc1 / 0x31637661), none, 1280x720, 4941 kb/s): unspecified pixel format". Now in my code I have set pix_fmt=AV_PIX_FMT_YUV420P, although I'm in the dark a bit here. Is the compl [22:51] <heffer2k02> aint coming from failure to read stuff from the h264 stream? Or a misconfiguration in the container? [22:52] <llogan> heffer2k02: are you using the libraries or the cli tool? [22:52] <heffer2k02> libraries. [22:52] <llogan> your best bet for an answer is probably the libav-user mailing list. [22:52] <heffer2k02> ok, thanks. [22:52] <llogan> the mailing list for the FFmpeg libraries... [22:53] <llogan> but stick around in case a library user is hanging around [23:12] <hotwings> hey fellas. i have a video that was encoded in 4 segments. seg1.mp4, seg2.mp4, seg3.mp4, seg4.mp4... how would i use ffmpeg to join those files consecutively without any re-encoding? the video in them is h264, the audio is ac3 [00:00] --- Sat Feb 7 2015

1 0

ffmpeg-devel.log.20150205
by burek 06 Feb '15

06 Feb '15

[00:40] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:fd52d2d3d1ee: avcodec/mpegvideo_motion: Fix gmc chroma dimensions [00:40] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:692b22626ec9: swscale/utils: Limit filter shifting so as not to read from prior the array [00:41] <Timothy_Gu> aetasx: it's not practical with projs like FFmpeg [00:44] <Timothy_Gu> aetasx: who knows which codecs or filters the user will use? [00:45] <Timothy_Gu> aetasx: plus well-tested formats like H.264 will be better-optimized than not well-tested formats, simply because it is run for more times [01:18] <aetasx> Timothy_Gu: thats true, but they're given the choice to pick which ones to use or not to use right now so I didn't think of it as being an issue. Mainly something more for power users like the LTO option which I don't think I've seen anyone actually use in their builds (of the ones I've looked at anyway) [01:19] <Timothy_Gu> aetasx: the LTO option which I don't think I've seen anyone actually use in their builds [01:20] <Timothy_Gu> ^ true because LTO is a beast [01:20] <aetasx> yeah but how you guys have it setup works perfectly ;) [01:20] <Timothy_Gu> aetasx: not when you have literally 15 deps [01:21] <Timothy_Gu> each of them has at least 1 other dep [01:21] <Timothy_Gu> and sometimes circular deps too [01:21] <aetasx> those would only fall under LTO if they're compiled with support for it though [01:22] <Timothy_Gu> aetasx: but IIRC don't you need to have all of the libs compiled with LTO to have one resulting LTO program? [01:22] <Timothy_Gu> i.e. you cant mix LTO and non-LTO [01:22] <aetasx> sure ya can [01:23] <Timothy_Gu> or maybe im missing something? [01:23] <aetasx> if it doesnt have LTO information, it links in it just as it normally would [01:23] <aetasx> just doesnt do anything special with it [01:24] <aetasx> that problem often comes up on a number of libraries that you try to compile with LTO support and then link into ffmpeg like you said but if you compile them without LTO support, the issue goes away [01:25] <Timothy_Gu> aetasx: or well i played around with LTO with ICC a long time ago (they called it "IPO" interprocedural optimization). Maybe GCC is different I dunno [01:25] <aetasx> if you have ffmpeg link in just a bunch of normal libraries with no LTO, then you can enable the LTO configureation option and it'll just handle ffmpeg's code alone without worrying about the rest which works well [01:26] <aetasx> I think I saw that in their docs actually, one sec since I have the page opened anyway [01:27] <aetasx> https://gcc.gnu.org/wiki/LightweightIpo this? [01:29] <aetasx> I have no idea what -fripa is, my guess is its dated [01:29] <Timothy_Gu> aetasx: not really the page is about profile-guided optimization, which thye call lipo [01:31] <aetasx> no thats the FDO stuff, theres some other -fripa flag [01:31] <aetasx> never seen it before [01:33] <aetasx> heres the note that they have on LTO in the optimization area "When producing the final binary, GCC only applies link-time optimizations to those files that contain bytecode. Therefore, you can mix and match object files and libraries with GIMPLE bytecodes and final object code. " [01:34] <aetasx> it just skips stuff that wasn't compiled with -flto on so any of those unresolved symbol errors you typically get sometimes simply go away since it treats it as a normal build [01:35] <aetasx> honestly if you were to do an ffmpeg build right now and just turned the flag on, most likely it will just work perfectly like mine unless you're using LTO elsewhere [01:35] <Timothy_Gu> aetasx: OK [01:36] <aetasx> Im not sure who put the feature in but it was done pretty successfully [01:36] <Timothy_Gu> aetasx: in gcc or ffmpeg? [01:36] <aetasx> into ffmpeg [01:37] <aetasx> I havent run across any other libs that have it and have it just work out of the box without fiddling [01:37] <Timothy_Gu> it's literally a 20-line change of adding `-flto` so nothing surprising hard here [01:38] <Timothy_Gu> aetasx: cairo has it IIRC [01:38] <aetasx> yeah but if you pass -flto in through CFLAGS, things go horribly wrong [01:38] <Timothy_Gu> but as cairo is a library downstream complained about cairo enabling LTO BY DEFAULT and then they removed it altogether [01:39] <aetasx> yeah there's too many hiccups when trying to link it into other things to turn it on by default [01:39] <Timothy_Gu> aetasx: you just need to pass -flto into LDFLAGS too [01:40] <Timothy_Gu> aetasx: all the other changes in https://github.com/FFmpeg/FFmpeg/commit/65d12900432ac880d764edbbd3681843148… are either refactoring or checking lto /really/ works [01:40] <aetasx> it doesnt work, I think its applying flto to object files with assembly or something and they end up getting trashed on the way out [01:41] <Timothy_Gu> aetasx: oh yeah that [01:41] <aetasx> I ran into that problem initially before finding the flag [01:41] <Timothy_Gu> that's because gcc is dumb [01:41] <Timothy_Gu> https://github.com/FFmpeg/FFmpeg/commit/a64df19c5bd22f543a254a7b93c82ade500… [01:41] <Timothy_Gu> https://github.com/FFmpeg/FFmpeg/commit/a01fedd9b87f05de45b7e49a62bec4bcbd0… [01:41] <aetasx> lol [01:42] <aetasx> their IRC room, despite having hundreds of people, is like a ghost town [01:42] <Timothy_Gu> aetasx: lol [01:42] <aetasx> I could go there to take naps [01:43] <Timothy_Gu> aetasx: I sent a message there let's see :) [01:44] <aetasx> Timothy_Gu: theres quite literally days in gaps between messages in my logs where the only breaks in between silence are people who ask questions and never get replies [01:44] <Timothy_Gu> aetasx: OK I believe you [01:45] <Timothy_Gu> aetasx: GCC is dying, slowly [01:45] <Timothy_Gu> aetasx: only companies like Intel and Google are still keeping it alive [01:46] <aetasx> Timothy_Gu: whats your opinion on CLANG compared to it? I've heard its still a bit behind in performance but catching up quickly [01:46] <Timothy_Gu> aetasx: see http://trac.ffmpeg.org/wiki/CompileBenchmarks [01:46] <Timothy_Gu> (those are my results) [01:47] <Timothy_Gu> I mean, it works pretty well. I never had any complaints with it. [01:47] <aetasx> that is definitely interesting [01:47] <Timothy_Gu> but it still needs some time to mature, and for big companies to start sending patches to it so that features like LTO are implemented [01:48] <aetasx> I was reading something about it having built-in opencl support for openmp or something. I forget what they're calling that standard [01:48] <Timothy_Gu> omp is semi-automatic parallelization [01:48] <aetasx> here it is http://www.openacc.org/ [01:49] <aetasx> from the gcc docs: https://gcc.gnu.org/wiki/Offloading [01:49] <Timothy_Gu> sounds like a competitor to OpenMP [01:50] <Timothy_Gu> so it's like OpenMP with automatic offloading support [01:51] <aetasx> thats what I took away from it, yeah [01:51] <aetasx> gcc5 apparently has it [01:53] <aetasx> I was looking at it to see if it had any benefit for multimedia uses and encoding [02:00] <aetasx> what libs do you typically build in with your ffmpeg? I may just do a generic build with its most used libraries and just enable the LTO flag to get some stats as far as binary reduction size, speed, etc? [02:17] <compn> Timothy_Gu : should encourage tcc to build ffmpeg ;) [02:17] <compn> i think gcc will be around forever, if only for backwards compat [02:23] <aetasx> I think he meant in leading development. I think its just assumed anything in unix-like systems will be around longer than reasonable [02:36] <cone-616> ffmpeg.git 03Philip Langdale 07master:d20df2601f02: avcodec/nvenc: De-compensate aspect ratio compensation of DVD-like content. [02:48] <Timothy_Gu> compn: gcc will be there forever. But the people maintaining it will decrease over time [03:30] <cone-616> ffmpeg.git 03Lukasz Marek 07master:21051af218d7: lavf/avc: add buffer padding to extradata allocation [04:26] <jamrial> kurosu, michaelni: http://fate.ffmpeg.org/report.cgi?time=20150205015545&slot=x86_64-archlinux… [04:27] <jamrial> very likely to be a regression introduced with da81cc38 or 52f2adc0 (judging by what the last good commit was and which ones touched hevc files since then) [05:23] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:f2579dbb4b31: avformat/thp: Check av_get_packet() for failure not only for partial output [05:47] <Timothy_Gu> TGBot: hi [05:47] <TGBot> Timothy_Gu: Hello! [05:48] <Timothy_Gu> If I find time I'll use TGBot to make a FATE status bot [05:48] <Timothy_Gu> TGBot status [05:48] <TGBot> I'm running quite happily. [06:12] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:7801a54ec3d9: avfilter/vf_fps: update frame drop comment [08:45] <j-b> 'lo [11:02] <cone-125> ffmpeg.git 03Paul B Mahol 07master:57bca64b3f44: avcodec/cllc: use init_get_bits8() [11:02] <cone-125> ffmpeg.git 03Paul B Mahol 07master:dccd648f9fa0: avcodec/alsdec: use init_get_bits8() [13:25] <durandal_1707> wm4: what is broken in ffmpeg? [13:29] <wm4> durandal_1707: returning vp9 timestamps from the parser when parsing webm vp9 packets [13:29] <wm4> they simply get lost [13:30] <durandal_1707> is that regression? [13:30] <wm4> for those "golden frames", which, in webm, are put in the same packet as a normal frame [13:30] <wm4> I don't know [13:30] <wm4> the way Libav handles this (inside of the decoder), it just works [13:30] <wm4> Libav has no vp9 parser [13:31] <nevcairiel> but libav also has no frame threading [13:31] <wm4> then this could be the reason [13:34] <cone-125> ffmpeg.git 03Carl Eugen Hoyos 07master:afada9ac67a4: Fix compilation with --disable-protocol=udp [13:34] <cone-125> ffmpeg.git 03Michael Niedermayer 07master:7dad2f7bb975: Merge remote-tracking branch 'cehoyos/master' [14:51] <rcombs> might be a good time to make sure none of ffmpeg's uses of gethostbyname could be vulnerable to CVE-2015-0235 [14:51] <Daemon404> well [14:51] <Daemon404> the correct solution is to not use gethostbyname [14:52] <Daemon404> its not useful in 2014 [14:52] <Daemon404> er [14:52] <nevcairiel> shouldnt people just update the libc :p [14:52] <Daemon404> 2015\ [14:52] <rcombs> Daemon404: you'd think! [14:53] <Daemon404> you should use getaddrinfo [14:53] <rcombs> Daemon404: but if you're running on a machine old enough not to have getaddrinfo (which AFAIK is the only case where we'd use gethostbyname)& [14:53] <Daemon404> inb4 some stupid AIX system doesnt have it [14:53] <Daemon404> ot something [14:53] <Daemon404> ... getaddrinfo is not new [14:53] <Daemon404> in any sense of the word [14:54] <rcombs> if no supported systems lack getaddrinfo, we should remove the gethostbyname wrapper [14:54] <Daemon404> depends what you define as support [14:54] <Daemon404> e.g.: i dont give ashit about haiku [14:54] <Daemon404> or plan9 [14:55] <Daemon404> or in fact HP-UX or AIX [14:56] <Daemon404> i see the abomination of ffserver also directly uses gethostbyname for some reason [14:56] <Daemon404> that makes no sense [14:56] <rcombs> \o/ [14:58] <Daemon404> well if you are running ffserver in prod, youre probably not smart enough to update your systems either. so. [14:58] <Daemon404> <_< [15:01] <Daemon404> rcombs, the sole author if the wrapper said primarily for win2k... [15:01] <Daemon404> so... perhaps we can nuke it [15:01] <rcombs> \o/ [15:01] <rcombs> sounds good to me [15:01] <JEEBsv> from the orbit [15:02] <Daemon404> we dropped win2k support quite a while back [15:02] <rcombs> if you're running win2k you shouldn't be on a network anyway [15:02] <Daemon404> xp has getaddrinfo. [15:02] <rcombs> (nor should you reproduce) [15:04] <Daemon404> even compn updated.... to xp [15:04] <Daemon404> ... last year [15:04] <nevcairiel> lol [15:04] <nevcairiel> after it went out of support, i'm sure [15:05] <Daemon404> aye [15:05] <rcombs> :| [15:05] <nevcairiel> i saw some idiots on doom9 yesterday advocating for xp64 to be the best OS ever [15:05] <nevcairiel> i was not sure if it was a troll post [15:06] <Daemon404> it most certainly was not [15:06] <rcombs> :| [15:06] <kierank> yes it was [15:06] <kierank> much better than lunix at the time anyway [15:07] <Daemon404> kierank, the key here is: in 2015 [15:07] <nevcairiel> but the post was from yesterday [15:07] <nevcairiel> not from 15 years ago [15:31] <compn> it was a mistake to update to xp too [15:31] <compn> because xp didnt support my 4tb hd :\ [15:31] Action: compn should switch back [15:34] <wm4> so when do we drop xp support? [15:35] <ramiro> compn: so you upgraded from one obsolete OS to another? [15:36] <Daemon404> wm4, the official stance is: we drop it when it gets in the way in an unreasonable way [15:36] <Daemon404> i.e. we're not strictly keeping it if it's a big problem [15:37] <wm4> dropping it would allow easy implementation of pthread_once, and removing stupid things like the lock manager [15:38] <Daemon404> propose it [15:38] <Daemon404> you have my axe^Wsupport [15:41] <wm4> I'm trying to find out how the fuck parsers handle timestamps [15:42] <av500> they fuck with them [15:50] <compn> ramiro : yes. i also installed vista on another machine to handle the 4tb :P [15:51] <ramiro> compn: win2k to vista is a downgrade =P [15:51] <compn> whys everyone hate on obsolete OS anyhow. [15:51] <Daemon404> ... why would you install vista [15:51] <Daemon404> win7 is agreed to be superior in every way [15:51] <compn> i wanted to switch to linux but then debian removed ffmpeg and mplayer so it took me a while to find a replacement distro :\ [15:52] <compn> Daemon404 : except w7 is bloated more than vista [15:52] <ramiro> nothing is bloated more than vista. not even windows me [15:52] <compn> have you used w7 ? [15:52] <Daemon404> what [15:52] <Daemon404> w7 is LESS bloated [15:52] <Daemon404> sys reqs are far less [15:52] <compn> anyways i'm done with windows [15:53] <compn> next upgrade/install is 100% linux. [15:53] <Daemon404> i have used every windows within a month or two of its release. [15:53] <ramiro> compn: this is a great distro: http://www.linuxfromscratch.org/ [15:53] <Daemon404> sicne ~98se [15:53] <ramiro> even though it's technically not a distro =) [15:56] <compn> building from source sounds like a big hassle :P [15:57] Action: compn remembers installing slackware and compiling a kernel [15:58] <compn> ah well [15:58] <compn> whos going to test binary codecs when i ditch the windows boxes ? [15:59] <ramiro> install win2k in a virtualbox =) [15:59] <ramiro> to remember the good ol' times [16:04] <cone-125> ffmpeg.git 03Michael Niedermayer 07master:e09ad5bd0de4: avcodec/h264_refs: set last_pic_for_ec only if it has not been set previously [18:37] <jamrial> kurosu: since you oked 2 and 3, is "[PATCH 1/3] hevcdsp: remove compilation-time-fixed parameter from sao_edge_filter" ok as well? (the former depend on the later after all) [18:38] <kurosu> oh right [18:38] <kurosu> indeed [18:39] <jamrial> ok, thanks [18:40] <kurosu> I have one last patch on sao band filter, but it depends on whether you want to use the fixed parameter to reduce the number of gprs so as to port to x86_32 [18:41] <kurosu> so I'll just send it, up to you afterwards [18:44] <jamrial> sao_band already uses only 6 gprs even if one of the parameters is stride_src. making it 5 by removing stride_src will not make x86_32 any easier since it's already good to go as is [18:44] <jamrial> it's a matter of using stack to get around the xmm reg limit of x86_32 [18:44] <kurosu> ok, I didn't quite check it [18:44] <jamrial> nonetheless i was planning on also removing stride_src from it [18:45] <kurosu> patch sent, basically, we can most often do band filtering in-place [18:45] <jamrial> to keep both band and edge similar [18:45] <kurosu> but the stride is different in that case [18:46] <jamrial> yeah, 2*MAX_PB_SIZE [19:05] <cone-125> ffmpeg.git 03James Almer 07master:1f1c7c8a57e8: hevcdsp: remove compilation-time-fixed parameter from sao_edge_filter [19:05] <cone-125> ffmpeg.git 03James Almer 07master:042c1159fcf8: x86/hevcdsp: add ff_hevc_sao_edge_filter_8_{ssse3,avx2} [19:06] <cone-125> ffmpeg.git 03James Almer 07master:15574c505b81: x86/hevcdsp: add ff_hevc_sao_edge_filter_{10,12}_{sse2,avx2} [19:39] <klmno> q:how can I tell if my reply was received at ffmpeg-devel(a)ffmpeg.org ? [19:39] <klmno> I did add myself on bcc, and the email went out, but there is no update on the email list, and I got no notification from the email list [19:50] <nevcairiel> If you want to get replies you should be signed up for the ML [19:56] <kurosu> michaelni, so yes, you're right, the original reference counting was just a lazy av_freep, and we can readd just that [19:57] <kurosu> it's supposed to be a new sps, but I'm not sure how it would behave if other threads are still using those buffers, though [19:58] <klmno> nevcaririel: I am in the ffmpeg-devel ML, and I also have "Receive acknowledgement mail when you send mail to the list?" and "Mail Delivery" enabled [19:59] <klmno> it works when I send patches using send-mail but it doesn't work when I just reply to emails [20:00] <klmno> I also have "Avoid duplicate copies of messages?" set to NO [20:57] <michaelni> kurosu, if other threads can acess the buffers after free() that would be bad, is that possible ? [21:08] <michaelni> seems it passes hevc fate under valgrind, should i apply it ? [21:17] <kurosu> michaelni, I have mixed feelings on this [21:17] <kurosu> the only sequence behaving that way isn't sufficient to evaluate a risk [21:19] <kurosu> it's an improvement for the fate test, but I would imagine cases where stream changes format with a new sps, and then... [21:19] <kurosu> the problem is we can only do a best guess fixing that issue, which isn't nice [21:23] <cone-125> ffmpeg.git 03wm4 07master:a83b6b077c56: vp9: assign PTS to visible instead of invisible frames [21:29] <kurosu> I'm not so sure other parts of the decoder are resilient to resolution change anyway [21:33] <kurosu> I'm more for applying the fix, overall [22:00] <cone-125> ffmpeg.git 03Christophe Gisquet 07master:9a2f5d825a25: hevc: free sao buffers when receiving a new SPS [22:00] <michaelni> kurosu, applied [22:17] <cone-125> ffmpeg.git 03Seppo Tomperi 07master:0c494114ccbe: hevcdsp: ARM NEON optimized deblocking filter [22:37] <wm4> getting a crash in asm with wma lossless [22:38] <cone-125> ffmpeg.git 03Michael Niedermayer 07master:43b1fd5bb4ef: avcodec/hevc: drop redundant ifs() [22:54] <j-b> michaelni: why does FFmpeg has this very peculiar way of dealing with VP9 ? [22:55] <j-b> Is there a good reasons? [22:55] Action: j-b curious [22:55] <nevcairiel> Frame threaded decoding [22:55] <wm4> and webm being stupid [22:55] <nevcairiel> Its not that peculiar either, it just splits individual frames [22:56] <nevcairiel> The super frames are really the ones you should blame [23:10] <wm4> could it be that ff_scalarproduct_and_madd_int16_ssse3 is broken on 32 bit? [23:14] <wm4> michaelni: ping, it seems your changes to lossless_audiodsp might have caused this, although it's only a theory, I didn't check [23:15] <ubitux> does it exists? [23:16] <wm4> what exists? [23:16] <ubitux> ff_scalarproduct_and_madd_int16_ssse3 [23:16] <ubitux> i see only the definition for mmxext and sse2 [23:16] <wm4> why wouldn't it? I have no diea how it comes into existence, though [23:17] <ubitux> but i indeed see the assignment for ssse3 [23:17] <ubitux> ah, it's below, my bad [23:17] <ubitux> yeah it can't work for 32-bits [23:17] <ubitux> it uses 10 reg [23:17] <wm4> so it tries to use scalarproduct_and_madd_int16_fallback? [23:18] <j-b> nevcairiel: sure, but it works with libav, and with libvpx. Only FFmpeg neeeds the parser to be inserted, as far as I can ee [23:18] <j-b> see [23:18] <nevcairiel> j-b: libav is drastically slower though [23:19] <nevcairiel> and libvpx has a entirely different architecture (and is slower, too) [23:19] <nevcairiel> some codecs just require the parser, vp9 is not the only one, its just a fact of live and doesnt really change anything [23:20] <ubitux> wm4: ff_scalarproduct_and_madd_int16_ssse3 should not be called on 32-bit, xmm reg is 10, it can't work [23:21] <ubitux> you can probably send a fix for libavcodec/x86/lossless_audiodsp_init.c [23:21] <nevcairiel> that file is weird .. the SSSE3 variant is slower when the CPU has SSE 4.2? [23:22] <jamrial> it wouldn't compile with x86_32 if it really needed 10 xmm regs [23:22] <wm4> ubitux: how do I check for 32 bits? [23:22] <wm4> sizeof(void*)? [23:23] <nevcairiel> wah, you check at compile time with a config thing of course [23:23] <jamrial> wm4: no, but still, the function should work on x86_32 [23:23] <ubitux> wm4: !ARCH_X86_64 [23:23] <jamrial> look at the SCALARPRODUCT_LOOP macro [23:23] <jamrial> it has a path for both x86_64 and x86_32 [23:23] <nevcairiel> a definition higher than 8 in the head isnt actually bad, as long as it doesnt use any more [23:24] <ubitux> jamrial: the definition should be conditional then [23:24] <ubitux> btw, can't we check that at compile time too? [23:24] <nevcairiel> ubitux: it doesnt change generated code however, x86inc just lowers the number on 32-bit iirc [23:24] <ubitux> like, a cglobal with more registers than your cpu allow [23:25] <nevcairiel> it would break anyway if you actually used more than allowed [23:25] <ubitux> at build time? [23:25] <ubitux> didn't know that, ok [23:25] <jamrial> why do you want to check for x86_32? [23:25] <nevcairiel> yeah, no opcodes for the extra regs [23:25] <nevcairiel> cant assemble [23:25] <ubitux> wm4: where does it fail more exactly then? [23:25] <jamrial> the function compiles regardless of arch [23:26] <ubitux> sorry for misleading [23:26] <jamrial> It's written to work on both archs. with x86_64 it uses 10 xmm regs, with x86_32 it uses stack [23:30] <wm4> ubitux: I don't know, the debugger stops in the middle of nowher [23:30] <wm4> e [23:36] <michaelni> wm4, what changes? how can i reproduce ? [23:41] <wm4> michaelni: d3512a0e89262a3ab08f30bab6ff3b98ac4c6e01 and the next 2 parent commits looked suspicious to me, especially since this was a lossless wma track that didn't decode before (so probably a 24 bit one) [23:42] <wm4> this is the file I tested with http://www.cccp-project.net/beta/test_files/[CCCP]_Mega_Lossless_Audio_Test… [23:42] <klmno> can I get some help with sensing a reply to ffmpeg-devel(a)ffmpeg.org? [23:42] <wm4> it has 12 audio tracks or so [23:42] <klmno> I did add myself on bcc, and the email went out, but there is no update on the email list, and I got no notification from the email list [23:42] <klmno> I am in the ffmpeg-devel ML, and I also have "Receive acknowledgement mail when you send mail to the list?" and "Mail Delivery" enabled [23:43] <klmno> it works when I send patches using send-mail but it doesn't work when I just reply to emails from my mail-client [23:43] <nevcairiel> sounds like your mail client is broken [23:43] <wm4> michaelni: oh sorry, it's http://www.cccp-project.net/beta/test_files/[CCCP]_Mega_Weird_Audio_Test.mkv [23:43] <klmno> is gmx web interface [23:43] <nevcairiel> or you're using a different address which is not subscribed [23:43] <klmno> I use the proper one, if different then I get a bounced email [23:44] <klmno> that the email is waiting for approval [23:44] <wm4> michaelni: both wmalossless tracks crash if I allow AV_CPU_FLAG_SSSE3; seems to work if that is disabled [23:45] <wm4> I can just send a patch and disable it on 32 bit [23:45] <wm4> if that's ok [23:45] <nevcairiel> that doesnt sound like the reason for the breakage [23:45] <nevcairiel> fix it, dont disable it :p [23:45] <jamrial> wm4: try changing 4,5,10 to 4,4,10 in the ssse3 version [23:46] <wm4> jamrial: appears to work [23:46] <wm4> no crash and output sounds like it should be [23:46] <jamrial> good [23:46] <jamrial> 32 bits takes all arguments from stack, jumping from the ssse3 version to the mmxext version screwed everything [23:47] <jamrial> since it was 4,4 [23:47] <wm4> so part of the signature has to match [23:47] <wm4> seems silly to do the jump in the asm though [23:47] <jamrial> ssse3 did 5 push, mmxext 4 pop, so stack was ruined [23:47] <nevcairiel> oddly enough the ssse3 version even uses mulm to access it, and not a register anyway [23:49] <klmno> nevcairiel: can I forward the email to you? is there a chance that you can post it? [23:50] <nevcairiel> if you want to participate in development, you should figure out why your mail isnt working, not really feeling the need to play mail proxy [23:50] <klmno> make send, ty for y time [23:50] <wm4> wat [23:55] <nevcairiel> apparently patience to find someone who actually admins the ML is a lost virtue [23:55] <nevcairiel> or getting a proper mail client [00:00] --- Fri Feb 6 2015

1 0

ffmpeg.log.20150205
by burek 06 Feb '15

06 Feb '15

[00:48] <techtopia> hey guys, my cable company lets me watch videos and shows on demand, i just tried downloading one of them and was going to re encode it [00:48] <techtopia> but the video is black and there is no audio [00:48] <techtopia> anyone know whats up [00:49] <techtopia> http://pastebin.com/wExEYA5D [00:49] <techtopia> the video and audio are their but they don't play back [01:38] <macmichael01> I have a .m2ts file containing LPCM audio. Is it possible to mux the audio into a MKV container? When I try, I get this message: No wav codec tag found for codec pcm_bluray. [01:40] <macmichael01> http://pastie.org/9887554 [01:44] <c_14> try switching -acodec copy to -c:a pcm_s16le [01:44] <c_14> Or if you want to keep it closer to the original, use pcm_s16be [01:47] <macmichael01> Does that transcode the audo into the new container or mux it? Also if the audio is 24 bit I would need to change that number to 24? [01:47] <c_14> transcode, yep [01:48] <macmichael01> ok Ill try that out. Whats the difference between le and be? [01:48] <c_14> little endian vs big endian [01:48] <macmichael01> duh gatcha thanks! [02:43] <pentanol> hello everyone. [02:43] <pentanol> Im here with question about unsynchronisation audio and video channals [02:44] <pentanol> what I did: [02:44] <pentanol> ffmpeg -i /mnt/xfs/mov_de/HP_undRerSteinDerWeisen.ac3 -ss 3 -i "concat:vts_02_1.avi|vts_02_2.avi|vts_02_3.avi|vts_02_4.avi|vts_02_5.avi|vts_02_6.avi|vts_02_7.avi|vts_02_8.avi" -t 600 -map 1:0 -map 1:1 -map 1:2 -map 0:0 -vcodec copy -acodec copy -y HP_undRerSteinDerWeisen.avi [02:44] <pentanol> I got audio\video unsynch after 2-3 minutes of the movie [02:45] <pentanol> also I did reencoding: [02:45] <pentanol> ffmpeg -i /mnt/xfs/mov_de/HP_undRerSteinDerWeisen.ac3 -ss 3 -i "concat:vts_02_1.avi|vts_02_2.avi|vts_02_3.avi|vts_02_4.avi|vts_02_5.avi|vts_02_6.avi|vts_02_7.avi|vts_02_8.avi" -t 600 -map 1:0 -map 1:1 -map 1:2 -map 0:0 -vcodec copy -acodec ac3 -ac 2 -ar 192kb -y HP_undRerSteinDerWeisen.avi [02:50] <techtopia> you will [02:50] <techtopia> transcode to a transport stream [02:50] <techtopia> then concat [02:50] <techtopia> or use somthing else to merge the files, [02:52] <pentanol> transcode to a transport stream- what do you mean by that? [02:52] <pentanol> ac3 file is already encodded [02:52] <pentanol> I have only vob's for concat [02:53] <techtopia> ignore the audio for now [02:53] <pentanol> In first I should do the concat and then add new audio stream to avi file? [02:53] <pentanol> I did like that [02:53] <pentanol> but anyway I get wrong synch [02:54] <techtopia> ffmpeg -i file_01.avi -c copy -bsf:v h264_mp4toannexb -f mpegts file_out_01.ts [02:54] <techtopia> do that on them all [02:54] <pentanol> i.e. make concat for audio and video separately and then merge third audios and one video? [02:54] <techtopia> then concat the ts's [02:54] <techtopia> and it will remain in sync [02:54] <techtopia> then transcode back [02:54] <techtopia> or just use another app [02:55] <techtopia> ac3 just copy /b 1.ac3 + 2.ac3 + 3.ac3 out.ac3 [02:55] <techtopia> like that then mux back in later [02:56] <pentanol> when it trouble with video stream? [02:56] <pentanol> but with ather audio streams I don;t found this trouble [02:56] <techtopia> im saying deal with the audio and video seperatly [02:56] <techtopia> then mux back together when done [02:56] <pentanol> ok, I will try [02:57] <pentanol> after I made .ts I can do mpeg2video after that? [02:59] <pentanol> techtopia: but why this not works with just mpeg2video? onlt for h264_mp4toannexb ? [02:59] <pentanol> only* [03:25] <voltagex> c_14: are you still around? [03:25] <c_14> ye [03:26] <voltagex> re my compile issue - PKG_CONFIG_PATH is set [03:27] <voltagex> sorry for the delay in response [03:28] <voltagex> original issue was - I'm trying to build ffmpeg as a static binary on Debian, I'm getting "unable to find libass/fontconfig" even with a custom pkgconfig path - http://sprunge.us/AOJI, fontconfig.pc exists. [03:33] <c_14> How did you build/install fontconfig? [03:34] <voltagex> uhh... [03:35] <voltagex> crap, no access to the build machine [03:35] <voltagex> I built it normally with a prefix to put it in the ffmpeg_build path [03:35] <c_14> Did you build it statically? [03:35] <voltagex> yes [03:36] <voltagex> I'll make clean and try again in about 4 hours [03:39] <c_14> Can you then also test `./configure --prefix=/home/voltagex/ffmpeg_build --extra-cflags='-I$HOME/ffmpeg_build/include -static' --extra-ldflags='-L$HOME/ffmpeg_build/lib -static' --enable-static --disable-shared --disable-ffserver --disable-doc --pkg-config-flags=--static --enable-fontconfig' ? [03:39] <c_14> Since it looks like the problem is with fontconfig, lets make sure that works first and then work our way up from there. [03:40] <voltagex> ah right [03:40] <c_14> Oh, and I'll probably be in bed in 4 hours. Just highlight me and I'll look at it when I have time. [03:40] <voltagex> yeah no worries [03:41] <c_14> Also, if the above configure doesn't work, could you paste the output of that config.log as well as how you compiled fontconfig, and the output of `pkg-config --cflags --libs --static fontconfig' [03:44] <relaxed> voltagex: my static builds have ass support http://johnvansickle.com/ffmpeg/ [03:57] <techtopia> should a keyfile.key be like, clientKey=0000000000000000000000000 or just be like this 000000000000000000000 ? [03:57] <techtopia> 0's not being the real key [04:00] <techtopia> which ever way i do it, ffmpeg errors http://pastebin.com/8kCsGhiT [04:03] <techtopia> hmm i found more keys [04:03] <techtopia> serverKey_internal_ASV3=b3091aa3eb501b5b660c8ef2fedf7068-7941729fbce73339b8c18853077d482f-0-32 [04:03] <techtopia> ^ is that an aes key [04:03] <techtopia> looks too long [04:07] <techtopia> hmm now with the other keys it just crashes ffmpeg on this line [04:07] <techtopia> Format lrc detected only with low score of 5, misdetection possible! [04:07] <techtopia> no one ever downloaded aes encrypted video with ffmpeg before ? [04:12] <relaxed> techtopia: I have not. Did you use -cryptokey ? [04:13] <relaxed> or, ffmpeg -h full|less +/^crypto [04:15] <relaxed> or, man ffmpeg-all|less +/^' crypto' [06:22] <applejack> I can't work this out... [06:24] <applejack> take a stereo source, upmix to 5.1, channelsplit, apply highpass/lowpass to CTR/LFE, join... [06:25] <applejack> I've got the upmix/split/join [06:25] <applejack> but [LFE]lowpass says too many inputs to lowpass [06:26] <applejack> in this: http://pastebin.com/JxxzEVHB [06:28] <applejack> bleh, nm [10:16] <RobertNagy> The concat decoder. Shouldn't it sort the input files by start_time, e.g. when the file order loops around back to 000? [10:23] <minimoo> Morning, "Error when opening encoder for output stream [10:23] <minimoo> any idea how to diagnose that? [10:23] <minimoo> it's only happening on a particular type of import file (vcode: tscc(camtasia) -> h264) [10:24] <minimoo> now this would be a lot easier if windows copy/paste hadn't stopped working:P [10:29] <termos> I'm trying to mux smoothstreaming with libffmpeg, it works fine for a stream with just video but once I include audio everything plays at twice the speed of what it should [10:30] <termos> doing the same but just for flv and mp4 works just fine [10:30] <termos> it also works with ffmpeg from the command line, so I must be doing something wrong? Anyone have some ideas? [10:31] <minimoo> relaxed: http://imgur.com/x01TH9F [10:31] <minimoo> (that's the pastebin output in a rather unconventional format due to windows clipboard being crap :P) [10:32] <relaxed> minimoo: add -pix_fmt yuv420p [10:33] <relaxed> "baseline profile doesn't support 4:4:4" [10:33] <minimoo> ok, that obviously works - so erm, 4:4:4 is some weird import format in this case? [10:34] <minimoo> (i.e. how do I detect this type of video [10:34] <minimoo> i'm assuming from the comment I don't always want to use the -pix_fmt yuv420p ? [10:35] <relaxed> most likely you do, but ffmpeg tries to use the input color space if the encoder supports it [10:36] <minimoo> i'vve been encoding videos a in 'internal school you-tube' type thing for 3 years with those arguments now, and this has only just come up - so i'm just wondering if the fix is to always add -pix_fmt yuv420p or only in some cases :) [10:37] <minimoo> or equally, if ffmpeg's moved on and the command line i'm using is a bit dumb, feel free to say [10:39] <relaxed> minimoo: -pix_fmt yuv420p is safe for what it sounds like you're doing. [10:40] <minimoo> right, will do that then : [10:40] <minimoo> Thanks again - pretty sure you was the dude that helped about a 12-18 months ago when I was last yaer :0 [10:41] <relaxed> you're welcome [11:21] <applejack> does this look right? : http://pastebin.com/uD8tnvAC [11:24] <applejack> meant to upmix a stereo audio channel to ac3 5.1 [13:06] <relaxed> applejack: did you try only using -ac 6 [13:09] <applejack> relaxed: yea, that only copies the stereo sound to FL and FR [13:09] <applejack> the rest of them are silent [13:10] <relaxed> ok, I was just curious if it worked [13:12] <applejack> I think my delay was too long tho, and the differential on the back speakers was too high too [17:54] <Mista-D> is there a way to do -vcodec copy but to trim CBR bitrate padding? [17:55] <JEEB> if the padding is within the nal units, nope [17:55] <JEEB> if it's null packets in an mpeg-ts stream, then it should be possible [18:28] <Taylor__> Howdy howdy everyone [18:29] <FML> Can anyone help me with a problem I am having? Ive ran out of resources and I'm on a deadline and I can't figure out the problem... [18:29] <FML> woe is me [18:37] <FML> Sooooo yeah? [18:41] <c_14> If you described the problem someone might be able to help you. [18:41] <FML> Cool [18:52] <FML> okay, one second, I'm checking all my facts so I can pin point the issue [19:30] <FML> Hey I'm trying to combine a .ivf video file with a .opus audio file into an mp4, but all I get is video and a screeching audio for about a second then nothing on the final product, here is what I'm using to combine the two videos "ffmpeg -i outside_50.ivf -i outside_50.opus -r 15 video2.mp4" [19:30] <FML> ideas? [19:31] <JEEB> that's re-encoding your shit to whatever the defaults are in a random ffmpeg version, that's the first thing I want to note [19:31] <JEEB> anyways, post full terminal output into a pastebin as well [19:31] <JEEB> and link here [19:36] <klmno> q:how can I tell if my reply was received at ffmpeg-devel(a)ffmpeg.org ? [19:38] <klmno> I did add myself on bcc, and the email went out, but there is no update on the email list, and I got no notification from the email list [19:38] <c_14> https://ffmpeg.org/pipermail/ffmpeg-devel/ Check the archives? [19:38] <klmno> I did, and it is not there [19:39] <klmno> I se this is for general q, I will move to ffmpeg-devel IRC [19:41] <FML> klmno: here ya go http://pastebin.com/LGApMgtX [19:44] <klmno> what does that have to do with my q? [19:45] <FML> Oh sorry klmno [19:45] <FML> haha [19:45] <FML> I meant JEEB [19:45] <FML> JEEB: here you go JEEB http://pastebin.com/LGApMgtX [19:46] <JEEB> is 15fps the original frame rate of the video stream? [19:48] <FML> no, but if I try to change it then I get stuck in a building phase that can last hours. [19:49] <JEEB> uhh [19:49] <JEEB> so what is the source frame rate :P [19:49] <JEEB> probably not the 90fps seen in the input side if I guess :P [19:50] <FML> one sec JEEB let me check the code in the JS files, we are getting those two files from a server and they are coded to something around 15 fps but let me confirm [19:50] <JEEB> if they are VFR, good luck and have fun :P [19:51] <FML> So when a user uploads a video to the server the fps is 15 [19:55] <JEEB> then move -r 15 to before -i :P [19:58] <FML> still the same problem :( loud screetching of audio for maybe a second, then only video [20:03] <JEEB> does it sound OK with ffmpeg -i hurr.opus out.wav ? [20:07] <FML> So I just tried it again...and it works....wtf hahaha [20:08] <FML> let me try a set of new files to see what happens [20:16] <FML> ffmpeg -threads 8 -r 25 -loop 1 -i offline.png -t 5.00 -pix_fmt yuv420p -vcodec libx264 -b:v 1500k -s -1:240 -y offline.mp4 [20:16] <FML> oops [20:16] <FML> ignore that [20:21] <FML> ffmpeg -threads 8 -r 15 -loop 1 -i offline.png -t 5.00 -pix_fmt yuv420p -vcodec libx264 -b:v 1500k -s 320:240 -aspect 1.3333 -y offline.mp4 [20:22] <FML> damn it sorry guys, I'm getting my terminal windows all mixed up haha last one I swear [20:48] <iksik> hello [21:06] <jjohn> Ahoy, guy. I am having a weird problem with ffmpeg on Linux when I try to record my screen and my audio. [21:06] <jjohn> guys* [21:07] <jjohn> When I am encoding at 30 FPS (output), the audio stutters a lot to the point of not being present, and I get "ALSA buffer xrun" errors all the time. [21:07] <jjohn> But when I am encoding at 60 FPS (output - input is 60 FPS always), everything runs fine. [21:08] <jjohn> And no, I didn't mix them up. [21:09] <jjohn> I am running on linux (Gentoo, kernel version 3.18.5), ffmpeg version was pullsed from git: N-69056-g3cfd07a [21:15] <ChocolateArmpits> hello, can anyone tell if audio passing through a filtergraph is handled in float or it's native sample format ? [21:15] <gsd> i'm trying to stream a video to an html5 video player - is it possible to tell ffmpeg to only process a particular byte-range? [21:31] <klmno> clear [23:07] <micechal> Does ffplay provide decent performance and is recommended for general usage, or should I be better off using some player which uses ffmpeg as an backend, such as mpv? [00:00] --- Fri Feb 6 2015

1 0

ffmpeg-devel.log.20150204
by burek 05 Feb '15

05 Feb '15

[00:05] <cone-276> ffmpeg.git 03Luca Barbato 07master:898276c16b16: r3d: Fix pos type [00:05] <cone-276> ffmpeg.git 03Michael Niedermayer 07master:d304cb49eb09: Merge commit '898276c16b1683ac77723e97574a3bfdb29507fd' [00:29] <cone-276> ffmpeg.git 03Christophe Gisquet 07master:e93d3a22cb53: x86: lavu/x264asm: fix ymm register instantiation [00:48] <cone-276> ffmpeg.git 03wm4 07master:56cc02422088: avformat/mpc8: fix hang with fuzzed file [01:03] <compn> .ram is rm playlist , so maybe is good ? [01:03] <cone-276> ffmpeg.git 03wm4 07master:b737a2c52857: avformat/mpc8: fix broken pointer math [02:07] <cone-276> ffmpeg.git 03Fabrice Bellard 07master:da81cc38e8a9: avcodec/hevc: reduce memory for SAO [02:07] <cone-276> ffmpeg.git 03Christophe Gisquet 07master:00fe77855b2a: avcodec/hevc: Do not zero is_pcm on allocation [02:07] <cone-276> ffmpeg.git 03Christophe Gisquet 07master:52f2adc015ba: avcodec/hevc: Update the USE_SAO_SMALL_BUFFER case for the alignment requirements in FFmpeg [05:12] <cone-652> ffmpeg.git 03Michael Niedermayer 07master:77f326dc36b7: ffmpeg: Make CFR frame drop threshold user adjustable [10:26] <ubitux> http://b.pkh.me/kdtree-try6.png finally. [10:30] <nevcairiel> neat balance [10:32] <ubitux> yes, but still slow :) [10:32] <ubitux> 87.81% ffmpeg_g ffmpeg_g [.] colormap_nearest_node [10:32] <ubitux> 4.35% ffmpeg_g ffmpeg_g [.] set_frame_sierra2_4a [10:32] <ubitux> 2.26% ffmpeg_g ffmpeg_g [.] ff_lzw_encode [10:32] <ubitux> :( [10:33] <ubitux> maybe i should try removing the recursion in the search somehow [10:59] <cone-616> ffmpeg.git 03Paul B Mahol 07master:6e7ac5ee554b: avcodec/webp: use init_get_bits8() [10:59] <cone-616> ffmpeg.git 03Paul B Mahol 07master:e9eb94ac5a66: avcodec/qcelpdec: use init_get_bits8() [10:59] <cone-616> ffmpeg.git 03Paul B Mahol 07master:c93a7b264f76: avcodec/ac3dec: use init_get_bits8() [10:59] <cone-616> ffmpeg.git 03Paul B Mahol 07master:49f825713536: avcodec/tak_parser: use init_get_bits8() [10:59] <cone-616> ffmpeg.git 03Paul B Mahol 07master:1634d2bcd141: avcodec/on2avc: use init_get_bits8() [10:59] <cone-616> ffmpeg.git 03Paul B Mahol 07master:63cb546f0a55: avcodec/wnv1: use init_get_bits8() [10:59] <cone-616> ffmpeg.git 03Paul B Mahol 07master:cecc4ea1fc2f: avcodec/evrcdec: use init_get_bits8() [12:56] <cone-616> ffmpeg.git 03Carl Eugen Hoyos 07master:adfe89bc4813: Fix standalone compilation of the tblend filter. [13:10] <cone-616> ffmpeg.git 03Alex Converse 07master:82bd8a57692d: aacdec: Refactor decode_ics_info. [13:10] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:0afc26d330e6: Merge commit '82bd8a57692d8f3643ad0903c97e4a0f84393f53' [13:18] <cone-616> ffmpeg.git 03Liviu Oniciuc 07master:d2330e584411: doc/muxers/image2: remove the non-negative number requirement for start_number [13:18] <cone-616> ffmpeg.git 03Stefano Sabatini 07master:fd242b468d66: lavf/utils: fix if_( style [13:30] <cone-616> ffmpeg.git 03Alex Converse 07master:edd9ea3ffba1: aacdec: Refactor aac_er_decode_frame. [13:30] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:5348d7b605c0: Merge commit 'edd9ea3ffba1d5873dead1c57c74b1e3cc912167' [13:39] <cone-616> ffmpeg.git 03Niel van der Westhuizen 07master:0ee2573347ec: aacdec: Support for ER AAC in LATM [13:39] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:f53aa3f4f331: Merge commit '0ee2573347ecdb9cb5656001f7201d819eec16d8' [13:48] <cone-616> ffmpeg.git 03Niel van der Westhuizen 07master:7640c4a371a9: aacdec: Tables for length 480 AAC ELD. [13:48] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:dd72e26d0cea: Merge commit '7640c4a371a97899dfb443d980fd0bf8ac587754' [14:03] <cone-616> ffmpeg.git 03Alex Converse 07master:d615187f74dd: aacdec: Support for ER AAC ELD 480. [14:04] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:ba4fba8f4873: Merge commit 'd615187f74ddf3413778a8b5b7ae17255b0df88e' [14:11] <cone-616> ffmpeg.git 03Paul B Mahol 07master:469a17013554: avformat/siff: fix if_( style [14:27] <cone-616> ffmpeg.git 03Reimar Döffinger 07master:8c1ce7787ad7: aacsbr: silence message for SBR extension "padding". [14:27] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:85a8069d23ed: Merge commit '8c1ce7787ad7af8fc321b4a145e9dc965f6d0e86' [14:40] <cone-616> ffmpeg.git 03Paul B Mahol 07master:9f1a5dbc26bf: avcodec/cdxl: use init_get_bits8() [15:07] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:05e161952954: avformat/mpc8: Use uint64_t in *_get_v() to avoid undefined behavior [16:44] <cone-616> ffmpeg.git 03Hendrik Leppkes 07master:b11994593ae4: dxva2_hevc: properly fill the scaling list structure [17:18] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:8a21620c26b8: avformat/mov: Fix memleaks for duplicate STCO/CO64/STSC atoms [17:18] <cone-616> ffmpeg.git 03Stefano Sabatini 07release/1.2:fa8d407c1f8f: lavf/segment: remove duplicated and inconsistent cleanup code in seg_write_packet() [17:18] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:2dd1d0c60b06: avformat/cdxl: Fix integer overflow of image_size [17:18] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:fd4c80177b00: avformat/flvdec: Increase string array size [17:18] <cone-616> ffmpeg.git 03wm4 07release/1.2:931f4313b2ec: avcodec/dvdsubdec: fix out of bounds accesses [17:18] <cone-616> ffmpeg.git 03Dale Curtis 07release/1.2:07f634f94876: mov: Avoid overflow with mov_metadata_raw() [17:18] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:b6351f9978a4: avformat/mov: fix integer overflow in mov_read_udta_string() [17:18] <cone-616> ffmpeg.git 03Dale Curtis 07release/1.2:54a9f64d4296: mov: Fix negative size calculation in mov_read_default(). [17:18] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:e81236d1a5ac: avformat/matroskadec: Use av_freep() to avoid leaving stale pointers in memory [17:18] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:529488afa315: avformat/mov: Fix mixed declaration and statement warning [17:18] <cone-616> ffmpeg.git 03Johan Andersson 07release/1.2:5410d6d26fc6: cmdutils: update copyright year to 2015. [17:18] <cone-616> ffmpeg.git 03wm4 07release/1.2:8a3a3e3154b7: avcodec/dvdsubdec: error on bitmaps with size 0 [17:18] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:08fe58ef3280: ffmpeg: Clear error message array at init. [17:18] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:b00a3bf33206: avformat/movenc: workaround bug in "PathScale EKOPath(tm) Compiler Suite Version 4.0.12.1" [17:18] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:a86cc0cc1bd2: avcodec/flac_parser: fix handling EOF if no headers are found [17:18] <cone-616> ffmpeg.git 03Vittorio Giovara 07release/1.2:71bdcd66e15a: opt: check memory allocation [17:18] <cone-616> ffmpeg.git 03Vittorio Giovara 07release/1.2:051b47dc4e08: swscale: check memory allocations [17:18] <cone-616> ffmpeg.git 03Martin Storsjö 07release/1.2:237cf31a5736: rtpdec_h263_rfc2190: Clear the stored bits if discarding buffered data [17:18] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:abfa41618124: avformat/rmdec: Check for overflow in ff_rm_read_mdpr_codecdata() [17:18] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:9939d98b2c25: avformat/utils: Fix number suffixes in tb_unreliable() [17:18] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:0e79fd1ec925: swresample/dither: Cleanup number suffixes [17:18] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:ba5e66cf65f3: avformat/matroskadec: Fix number suffixes [17:18] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:ef3687998fb4: avformat/smacker: Fix number suffix [17:18] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:5f2902bdc76c: avcodec/h264_cabac: use int instead of long for mbb_xy [17:18] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:7881d3b2a5c3: avcodec/mpegvideo_enc: Fix number suffixes in rc_buffer_size calculation [17:18] <cone-616> ffmpeg.git 03wm4 07release/1.2:5d7320be52c3: avformat/tta: fix crash with corrupted files [17:18] <cone-616> ffmpeg.git 03wm4 07release/1.2:bdc4abd694a3: avformat/mpc8: fix hang with fuzzed file [17:19] <cone-616> ffmpeg.git 03wm4 07release/1.2:49dd89f9027f: avformat/mpc8: fix broken pointer math [17:19] <cone-616> ffmpeg.git 03Michael Niedermayer 07release/1.2:70ff376e8357: avformat/mpc8: Use uint64_t in *_get_v() to avoid undefined behavior [17:32] <kierank> great, a crash in libavfilter [17:41] <JEEB> my condolences [17:50] <wm4> as long as it doesn't crash in the format negotiation code... [17:55] <aetasx> finally got a profiled + LTOed build done with all the libs that don't break when enabling LTO [17:57] <aetasx> shaved about 27% of the encoding time off, pretty decent imo [17:58] <iive> :O [17:58] <iive> 27% total time off is enormous! [17:59] <Daemon404> that's a very vague statement [17:59] <Daemon404> encoding time *with what* [17:59] <Daemon404> and compared to what [17:59] <aetasx> libx264 placebo, non-profiled/lto vs profiled/lto [18:00] <Daemon404> that seems VERY unlikely [18:00] <Daemon404> most of the time is in its asm [18:00] <wm4> yeah [18:00] <Daemon404> to which LTO makes 0 difference [18:00] <Daemon404> unless you did something silly like disable asm. [18:00] <aetasx> lol no [18:00] <wm4> how did you test? [18:01] <Daemon404> if you perf x264, you see almost all its time spent in e.g. avx functions [18:01] <aetasx> they're just encoding times based on what I use it for and I don't have avx sooo [18:01] <aetasx> ;) [18:01] <Daemon404> s/avx/sse4.2/ then [18:01] <Daemon404> or whatever [18:02] <aetasx> wish libspeex and SDL would LTO properly, those two I had problems with [18:02] <Daemon404> it will also depend on what input, and if the input decoder has asm or not [18:02] <Daemon404> in which case youre jus tbenchmarking the decoder. [18:02] <aetasx> hey I didnt declare it scientific, its the speedup I get on my own [18:02] <wm4> aetasx: they're also useless, so you can just skip them [18:03] <aetasx> SDL yeah, yeah cant encode in speex without libspeex though [18:04] <aetasx> its still used in flash crap [18:06] <aetasx> that and I think it was gnutls or gcrypt that crashes the optimizer out when attempting it but I don't really have a need for them [18:08] <aetasx> wm4: I did take your suggestion and just start stripping everything off Ill never need too, cut quite a few megs out of the static build [18:31] <ubitux> so how do we reproduce the speedup? [18:41] <aetasx> the configure script using --enable-lto and tossing in fprofile-generate/use simply works right on its own so that much is basically a handout. configuration in ffmpeg is so much more resilient than the libs I had to deal with with the exception of maybe x264 which you can feed a sample file and it'll automate the entire profiling build [18:45] <aetasx> it might even justify just chucking an option for it in configure considering how easy it was to do with a little alternate environments testing. either way passing the flags is just as easy [18:59] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:7f02fcd91774: avfilter/vf_fps: Do not drop a random subset of frames [20:59] <ubitux> we should write down the politic of auto selection of libs [21:00] <ubitux> because OSX stuff or x11 junk autodetected is evil [21:00] <Daemon404> any autodetect of x11 is evil [21:00] <Daemon404> packagers will hate you [21:00] <Daemon404> SURE LETS DRAG IN X11 DEPS [21:00] <ubitux> yes, i don't understand why we enabled it by default [21:02] <ubitux> i mean, sdl is already a cancer [21:02] <nevcairiel> osx stuff is fine, since every osx system is going to have it anyway, but x11 is stupid [21:02] <ubitux> but we justify its probing for ffplay [21:02] <ubitux> nevcairiel: what about cross compiling from OSX? [21:02] <kepstin-laptop> the sdl stuff is only needed for ffplay, right? [21:03] <ubitux> kepstin-laptop: we have a sdl device output too, which creates a dep to it too [21:04] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:afa92907f3c6: avcodec/mjpegdec: Check escape sequence validity [21:04] <cone-616> ffmpeg.git 03Michael Niedermayer 07master:fabbfaa09566: avcodec/mjpegdec: Check number of components for JPEG-LS [21:06] <nevcairiel> ubitux: why would your cross-compiling env have the headers or libs for configure to succeed on these things [21:07] <ubitux> dunno, just wondering [21:14] <wm4> can we get a ssl lib autodetected? [21:14] <Daemon404> no [21:15] <Daemon404> since they are all lgplv2.1 incompatible [21:15] <wm4> who cares [21:15] <Daemon404> well anyone who distributes anything [21:17] <Stefan_G> Good evening! Anybody listening at the moment? I am experiencing segfaults with a static build of ffmpeg on Ubuntu 14.04 while attempting to transcode Magic Lantern MLV files. [21:17] <Stefan_G> This is no support request. [21:17] <Stefan_G> I suspect a bug and would provide an MLV file for analysis [21:18] <beastd> Stefan_G: https://ffmpeg.org/bugreports.html [21:18] <Stefan_G> Ack [21:19] <Stefan_G> Is there any option without having to register somewhere? I'm starting to lose count of the sites where I had to register just to report one bug... [21:24] <beastd> Stefan_G: You could also report the bug on the ffmpeg-user mailing list. [21:24] <beastd> But I would not really recommend it, though it would be better than not reporting the crash. Also did you try with current git master version of FFmpeg? [21:24] <Stefan_G> Well, ok, that's it then... There is no ffmpeg_g file in the static package I downloaded, so no gdb... [21:24] <Stefan_G> Yeah, I downloaded a staic build package with the latest git [21:25] <Stefan_G> ffmpeg version N-44018-g77f326d- http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2015 the FFmpeg developers [21:25] <Stefan_G> built with gcc 4.9.2 (Debian 4.9.2-10) [21:27] <beastd> Stefan_G: I think building FFmpeg yourself should not be too hard. It would also be helpful as you could test patches should it be necesarry. There are guides in the wiki: https://trac.ffmpeg.org/#CompilingFFmpeg [21:28] <Stefan_G> You're probably right but getting involved in a full debugging of the issue would be way over my time budget at the moment. I have too much other stuff on my plate. [21:29] <Stefan_G> And I just recently switched from SuSE to Ubuntu, so I'm not yet familiar with the distro that I could easily build stuff. Practically all dev-packages have slightly different names... [21:29] <Stefan_G> Total nightmare. [21:30] <beastd> Stefan_G: The packages are named in the wiki guides. and your probably do not need to many. [21:31] <beastd> Stefan_G: Also if you report without a backtrace there is still a chance a developer could reproduce with the help of your sample file. I can understand being short on time. [21:32] <Stefan_G> Unfortunately... I'm way over my head in projects... Everything more than git clone <repo>\n cd <dir>\n make config all install\n would be too much for me at the moment [21:33] <Daemon404> just open a bug with the sample file [21:33] <Daemon404> and the output [21:33] <Daemon404> that should be enough [21:33] <Stefan_G> Where can I put the sample file? It's nearly 2gig. MLV files tend to be huge... [21:34] <Stefan_G> I could put it to my home server but I only have 5mbit, so download might be slow [21:34] <Daemon404> we have an ftp but i dont know what its limits are [21:34] <Daemon404> i you can, say, reproduce with the first MB of the file, thats finee too. [21:35] <Stefan_G> If I would know the internals of MLV files... [21:35] <beastd> description about the ftp is on https://ffmpeg.org/bugreports.html [21:36] <Stefan_G> that file is only 30 secs of video! [21:36] <Daemon404> beastd, good point. [21:38] <beastd> if the crash is not reproducible with the small file, iirc uploading larger files worked fine in the past. [21:48] <Timothy_Gu> Stefan_G: your version looks odd. Isn't FFmpeg over N-65000 now? [21:48] <Stefan_G> I have no idea. But the crash also occurs with the stable releases [21:50] <Timothy_Gu> Stefan_G: See http://ffmpeg.org/bugreports.html#SubmittingSampleMedia [21:51] <Timothy_Gu> Stefan_G: try to dd the first maybe 10MB of the file and test if it crashes, and if it does just uploading the first 10MB of the file is fine [21:52] <ubitux> i'm too tired to try to explain to carl about libxcb, does anyone care to do it? [21:54] <Stefan_G> Yeah, it does... [21:54] <Stefan_G> Somebody can delete my other two attempts of uploading the files... [21:55] <cone-616> ffmpeg.git 03Seppo Tomperi 07master:74d7faf4001e: hevcdsp: separated sao edge filter and pixel restore funcs [21:55] <cone-616> ffmpeg.git 03Seppo Tomperi 07master:4386e1fd94cd: hevcdsp: simplified sao_edge_filter [21:55] <cone-616> ffmpeg.git 03James Almer 07master:121ccc336551: hevcdsp: further simplify sao_edge_filter [21:55] <cone-616> ffmpeg.git 03James Almer 07master:7457afc64d5f: hevcdsp: replace the SAOParams struct parameter from sao_edge_filter [21:55] <Stefan_G> The correct file name is bug_4296_sample_file_segfault_MLV_format_first10MB.mlv [21:57] <Stefan_G> Timothy_Gu: The readme says: [21:57] <Stefan_G> build: ffmpeg-git-20150204-64bit-static.tar.xz [21:57] <Stefan_G> version: N-44018-g77f326d [23:29] <aetasx> fate verifies you have support for something built-in before running a test against it yes? (Im assuming so since theres that randomized build configure option) [23:36] <nevcairiel> yes it should [23:36] <nevcairiel> if not thats a bug [23:45] <aetasx> thanks [23:56] <aetasx> whats your guys thoughts on letting fate run its tests against a fprofile-generate build as like a generic profiling option? I was going to play around with that idea and see what it turns out [00:00] --- Thu Feb 5 2015

1 0

ffmpeg.log.20150204
by burek 05 Feb '15

05 Feb '15

[00:05] <fixxxermet> c_14: Mind helping me with a more complete example? Here is what I'm currently using: ffmpeg -r 5 -i 'rtsp://$url' -ab 64k -vcodec copy -vb 448k -s 704x480 -f mpegts udp://127.0.0.1:5678 [00:05] <fixxxermet> That /works/, but I can't open that stream with vlc (it just hangs) or with ffplay (no frame! errors) [00:06] <c_14> just replace udp://* with tcp://127.0.0.1:5678?listen [00:08] <fixxxermet> I was using udp on purpose - does ffmpeg not support udp sockets? [00:09] <c_14> No, it does. It should work, but I've had lots of weird problems with it for strange reasons, try tcp and see if that works. [00:14] <fixxxermet> c_14: I thikn I'm almost there, but am missing something probably obvious: https://gist.github.com/kylejohnson/845d4c218844e20f0fd1 [00:17] <c_14> What's with the -f mpeg1video? [00:18] <fixxxermet> Ok, so I had it backwards. The receiving side should have ?listen, no? [00:18] <fixxxermet> c_14: Mistake. I thought I had 'mpegts' there [00:24] <fixxxermet> There we go... [00:24] <fixxxermet> -vcodec copy didn't work, -vcodec libx264 did. [00:27] <fixxxermet> So, ffserver might be what I need. If I understand correctly, with just ffmpeg, first I need to have a 'client' listening for a connection. From there I can output ffmpeg to a tcp stream (where the 'client' is listening) [00:28] <c_14> Not entirely, ffmpeg can also listen and wait (for one client per tcp stream) if you add the ?listen flag [00:29] <fixxxermet> ah, indeed it can [00:33] <fixxxermet> ok, now if I do need more than 1 client [00:34] <fixxxermet> ffserver then might be a good option [01:08] <applejack> how would I apply the lowpass/highpass filters to the lfe/center channels in a 5.1 ac3 stream? [01:10] <c_14> channelsplit highpass/lowpass channeljoin/merge/pan [01:10] <c_14> something like that probably [01:11] <c_14> s/channeljoin/join/ [01:11] <applejack> hmm [01:12] <applejack> can it be done in one pass tho? [01:12] <c_14> Sure [01:13] <c_14> Split the channels into separate streams, highpass the ones you want to highpass, lowpass the ones you want to lowpass and join em together again. [01:14] <applejack> do you just specify the filters in that order on the command line, or do you have to create a filter map [01:14] <c_14> Create a filter_complex [01:14] <applejack> ahh [01:14] <c_14> With at least 3 filterchains [02:49] <DX099> #libreoffice [02:53] <pentanol> hello [02:53] <pentanol> someone here? [02:53] <DX099> might not be of great helpo [02:54] <pentanol> I've trouble with synchronisation audio and video streams, after 600 seconds of the movie I get unsynchronisation and audio listen unhit video stream [02:55] <pentanol> I have third audio streams and this happen only with one audio stream [02:56] <DX099> pentanol, complete noob here but did you change framerate ? [02:58] <pentanol> yes, I did 192 and 128kb/s [02:58] <pentanol> that's ac3 audio stream in avi file [02:59] <pentanol> I don't remember all command, I have that at home. [02:59] <pentanol> and I have 3 movie with that trouble [03:00] <relaxed> well, it's hard to troubleshoot if we don't know what you did. [03:00] <pentanol> I just pul third audio track from another movie to that and get stuck after ~600 sec out of the movie [04:00] <_maz> hi I see at https://trac.ffmpeg.org/wiki/Create%20a%20video%20slideshow%20from%20images I can create a video slideshow. I would like to display specific images at specific times(rather than regular intervals) and add them to an uncompressed WAV file of a specific duration. is this possible? [04:48] <c_14> You can try using the duration option in the concat demuxer, and if that doesn't work specify each of the images separately as inputs and loop 1, then trim each to the duration you want and use the concat filter to concat. [05:16] <_maz> thanks c_14 will look into concat and the separate looping idea [05:21] <pzich> anyone know how to pass multiple quality streams to twitch? [07:17] <voltagex> I'm trying to build ffmpeg as a static binary on Debian, I'm getting "unable to find libass/fontconfig" even with a custom pkgconfig path - http://sprunge.us/AOJI, fontconfig.pc exists [07:44] <Tze_> Hi - is it possible for FFPLAY to play out video while graphically displaying the audio volume? something like this: http://imgur.com/ZFI27pL [07:58] Last message repeated 1 time(s). [09:27] <h4ni> Hello [09:27] <h4ni> i have a question about this wiki https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu [09:27] <h4ni> what i must do to install ffmpeg in /usr/local/bin ? [09:31] <h4ni> any help? [11:35] <emilsedgh> guys, I have 3 ffmpeg commands running. however, all 32 cores on my server are busy [11:35] <emilsedgh> is this normal? they all have like 10% CPU usage [11:36] <emilsedgh> and I see many many ffmpeg commands in my htop. I assume they are threads? [14:17] <c_14> emilsedgh: Depending on what you're doing with it, ffmpeg threads pretty well. So yes, probably normal. [14:18] <emilsedgh> thank you c_14. [14:18] <emilsedgh> one of the commands pushes an mp4 file to rtmp server. [14:19] <emilsedgh> then two other commands re-encode the rtmp to lower bitrates and push it to another rtmp streams [14:20] <c_14> voltagex: have you tried setting PKG_CONFIG_PATH ? [14:20] <c_14> emilsedgh: you can limit the number of threads ffmpeg uses, but normally it's nothing you should be worried about. [14:20] <emilsedgh> c_14: thanks a lot man :) [14:56] <RobertNagy> is it possible to capture 4 audio streams from blackmagic using the directshow device in ffmpeg? [15:05] <c_14> If you can capture 1 stream, then probably yes. [15:17] <ramiro> RobertNagy: there should be support for decklink directly, without using dshow [15:17] <RobertNagy> I don't think there is on windows [15:20] <ramiro> RobertNagy: it should. have you tried? [15:20] <RobertNagy> Not in a while. [15:21] <RobertNagy> I will try again [15:24] <ramiro> it was implemented late last year [15:46] <relaxed> RobertNagy: does "ffmpeg -devices" list it? [15:46] <RobertNagy> Indeed it does [15:46] <RobertNagy> wow, I've totally missed it [15:47] <RobertNagy> thx [15:48] <ramiro> RobertNagy: it's quite new =) [16:00] <RobertNagy> hm, it seems to only do output [16:08] <ramiro> RobertNagy: how old is your build of FFmpeg? [16:09] <RobertNagy> Dec 30 [16:18] <Harzilein> hi [16:20] <Harzilein> when i reencode already somewhat lossy h264 material in hd resolution to h263, the result looks a bit worse than when i reencode an sd source (only have xvid here). any idea what might make the decoded xvid smoother compared to the scaled h264? what filter should i try? [16:25] <jojek> Hi guys, I have a following question: [16:25] <jojek> I have some video file in which audio is some unknown to ffmpeg format: g722.1 (incorrectly recognized as aac): [16:25] <jojek> Input #0, mpegts, from 'input.ts': [16:25] <jojek> Duration: 00:00:34.00, start: 71056.228111, bitrate: 179 kb/s [16:25] <jojek> Program 1 [16:25] <jojek> Stream #0:0[0x21]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(tv, bt709), 1280x720, 10 fps, 12.50 tbr, 90k tbn, 20 tbc [16:25] <jojek> Stream #0:1[0x22]: Audio: aac (SSR) ([146][0][0][0] / 0x0092), mono, fltp, 789 kb/s [16:25] <jojek> Is it possible to extract this very track as binary data, without any processing? Then I could obviously try to convert this track with different tool. I am using ffmpeg in version N-69498-g7620d48. Or maybe someone has an idea how to decode g722.1 with ffmpeg? [16:28] <c_14> try -map 0:1 -c copy -f data out [16:31] <jojek> hi c_14, thank you for your attention. Here is the output: [16:31] <jojek> http://pastebin.com/cxnDUqm1 [16:32] <c_14> using -c:a adpcm_g722 before the -i ? [16:33] <c_14> eh, s/^/try/ [16:34] <jojek> You mean something like that? [16:34] <jojek> ffmpeg -i input.ts -map 0:1 -c:a adpcm_g722 -c copy -f data out [16:35] <c_14> before the -i [16:35] <c_14> ffmpeg -c:a adpcm_g722 -i input.ts [16:37] <jojek> http://pastebin.com/7ZbDJrU0 [16:37] <jojek> Still no luck, although you made a difference ;) [16:38] <c_14> Do you know the samplerate of the stream? [16:39] <jojek> I am afraid not, but probably there is a finite number of possibilities to try. [16:39] <c_14> Try supplying -ar 44100 as an input option [16:39] <c_14> (before the -i) [16:41] <jojek> It says that option sample_rate not found. Does it mean that adpcm_g722 doesn't accept that and you can only set bitrate? [16:42] <c_14> Switch the output part to -map 0:1 -c:a adpcm_g722 -ar 44100 out.mkv [16:42] <c_14> maybe forcing it for the output will help [16:44] <jojek> Here is the ouput: http://pastebin.com/y6jFzQxC [16:46] <c_14> -time_base 1/44100 <- not sure if that'll help or if it's even correct, but try adding it as an output option [16:46] <jojek> I got error while splitting the argument list as option not found. [16:47] <jojek> So indeed it is not possible to do that. Interesting [16:49] <c_14> mhm, might be time to open a bug report. I kind of wish ffmpeg had a way to just extract a stream as without checking validity for anything... [16:49] <jojek> I see [16:49] <jojek> would it help you if you had an original file? [16:50] <jojek> Or maybe it's just pointless... [16:50] <c_14> I don't know much about the internals of audio codecs, but if you can attach it to the bug report/updload it to the samples ftp and mention that on the bug report it'll help the developer who'll look at the ticket [16:51] <jojek> I see [16:51] <jojek> Can I ask you one more thing? [16:52] <jojek> Is it possible to attach a text file to, let's say, wav file? I.e. by muxing them together and then retrieve it just like that? [16:55] <c_14> Muxing it is easy, but demuxing it is a problem because of what I mentioned above about extracting invalid streams. [16:55] <c_14> though, there is a codec for text [16:55] <c_14> So it should work [16:55] <jojek> Jeez, I see... So there is no simple way of saying: treat it as binary data? [16:56] <c_14> None I know of, sadly. [16:56] <c_14> Mux it in with `ffmpeg -i file -c:a text -i text -c copy outfile' [16:57] <c_14> If it really is text it should be able to work with it. ffmpeg just can't work with data it doesn't know about (that I know of). [16:59] <jojek> Am I doing again something wrong? http://pastebin.com/zErS9HNd [17:05] <c_14> I think bintext is a video codec of some weeird sort, try using -c:a text---- wait, I just found the bin_data codec [17:05] <c_14> Try the file with the g722 , with -c:a bin_data as input option [17:06] <jojek> It's still hate me ;) http://pastebin.com/XKe7uxmm [17:07] <c_14> Input option, before the -i [17:08] <c_14> And just -c [17:08] <c_14> not -c:a [17:08] <c_14> actuall [17:08] <c_14> do -c:0:1 bin_data [17:08] <c_14> that should work [17:08] <c_14> *actually [17:09] <c_14> ffmpeg -c:0:1 bin_data -i input.ts -c:0:1 copy -f data out [17:09] <c_14> probably [17:10] <c_14> might have to change it a bit [17:10] <c_14> Never tried it before though. [17:10] Action: c_14 hopes it works [17:24] <Phlarp> Is it possible to set up an image overlay filter to scale/animate? [17:35] <FilterComplex> Hello [17:35] <FilterComplex> I have a problem with filtercomplex [17:35] <jojek> I see c_14 [17:36] <FilterComplex> I'm trying to run something like this: ffmpeg -i overlay_1.ts -i raw_1.ts -c:v libx264 -filter_complex_script "filter_complex.txt" -y -r 29.97 -pix_fmt yuv420p out_1.ts [17:37] <FilterComplex> in filter complex file I have: [1:0] format=rgba [1sared]; [0:0] setsar=sar=1, format=rgba [0rgbd]; [0rgbd][1sared]blend=all_mode='overlay':all_opacity=0.8,format=yuva422p10le [17:37] <FilterComplex> I get this error: First input link top parameters (size 846x360, SAR 0:1) do not match the corresponding second input link bottom parameters (846x360, SAR 1:1) [17:38] <FilterComplex> When I change the filter_complex to : [1:0] setsar=sar=1,format=rgba [1sared]; [0:0] format=rgba [0rgbd]; [0rgbd][1sared]blend=all_mode='overlay':all_opacity=0.8,format=yuva422p10le [17:38] <FilterComplex> I get this: First input link top parameters (size 846x360, SAR 1:1) do not match the corresponding second input link bottom parameters (846x360, SAR 0:1) [17:40] <FilterComplex> the funny part is that this was working perfectly 1 day ago... And I need to launch tomorrow :/ [17:41] <FilterComplex> can anyone help? [17:41] <FilterComplex> any opinion? [17:44] <kepstin-laptop> FilterComplex: simplest fix would probably be to put the setsar filter on both inputs [17:51] <FilterComplex> kepstin-laptop so something like this: [1:0] setsar=sar=1, format=rgba [1rgbd]; [0:0] setsar=sar=1, format=rgba [0rgbd]; [0rgbd][1sared]blend=all_mode='overlay':all_opacity=0.8,format=yuva422p10le ? [17:52] <FilterComplex> kepstin-laptop so something like this: [1:0] setsar=sar=1, format=rgba [1rgbd]; [0:0] setsar=sar=1, format=rgba [0rgbd]; [0rgbd][1rgbd]blend=all_mode='overlay':all_opacity=0.8,format=yuva422p10le ? [21:04] <fixxxermet> I'm starting to get the hang off ffserver. I have two feeds, and each feed has two streams (one webm, and one h264). I can open all 4 streams with vlc and ffplay [21:04] <fixxxermet> Does anyone have experience with live streaming in firefox with the <video> tag? It works in chromium, but not in firefox [21:09] <kepstin-laptop> fixxxermet: what format are you using? firefox explicitly does not (and will not) support playing mpeg-ts. I'm a bit surprised about webem not working tho. [21:12] <fixxxermet> kepstin-laptop: flv with libx264, and webm with libvpx (Format and VideoCodec, respectively) [21:13] <kepstin-laptop> well, flv won't work in firefox unless you have a flash player :) The only container it supports for h264 video is mp4 (which cannot be used for live streaming) [21:14] <kepstin-laptop> I'm a bit surprised chromium would allow that [21:15] <fixxxermet> Chromium is using the webm streams, not the flv streams [21:15] <PoeticallyEvil> Hi, I'm trying to concatenate 10+ songs (.flac) into one file, I use this command: ffmpeg -f concat -i mylist.txt -c copy output.flac. mylist.txt looks like so: "file './01 - Alternate World.flac'\nfile './02 - Lost It To Trying.flac' [...]". The ffmpeg output looks fine: "size= 251423kB time=00:41:39.77 bitrate= 823.9kbits/s", however playing the file stops at 2:55 (first file lasts 04:14) ? [21:15] <kepstin-laptop> if you really want to get live streaming working in firefox html5 video, you'd probably have to do dash streaming with a javascript dash library that uses the browser media source extensions. [21:15] <fixxxermet> kepstin-laptop: I'm investigating that as well - just seeing how far I can take straight-up html5 <video> [21:15] <fixxxermet> So for webm... Format = webm, VideoCodec = libvpx. /should/ that work in ff? [21:16] <PoeticallyEvil> It stops with "End of file". [21:17] <kepstin-laptop> fixxxermet: assuming you're using vp8 and not vp9, yeah. But I dunno how firefox handles live webm. [21:17] <kepstin-laptop> in theory, live webm should be indistinguishable from webm that just doesn't have an index, but i've had issues with unindexed webms not working properly in browsers before. [21:18] <kepstin-laptop> although I thought that was in chrome and not firefox... [21:46] <PoeticallyEvil> Huh, the " -c copy " is unnecessary, and actually what made the thing go wrong. "ffmpeg -f concat -i mylist.txt output.flac" works perfectly. [21:47] <PoeticallyEvil> If someone would consider it useful to mention this here: https://trac.ffmpeg.org/wiki/Concatenate. [21:47] <PoeticallyEvil> Then again, I'm concatenating one stream only (and not two as in a .mkv)... ? [21:48] <PoeticallyEvil> Anyhow, peace. [22:09] <fixxxermet> kepstin-laptop: I'ev actually got it working now. Was an issue with the hostname, not with ffmpeg [22:11] <BtbN> As far as i'm aware, live streaming without flash doesn't work in any browser except safari on mac. [22:12] <c_14> It works with webm. At least, I remember it working with webm at one point. [22:13] <BtbN> Twitch would love to get rid of flash. But they can't, because browser support is severely lacking. [22:16] <c_14> Browsers and video has always been lacking. [22:17] <BtbN> static video is mostly fine now [22:17] <BtbN> but live streaming is the next big problem [22:18] <BtbN> It's pretty much allways done with h264 [22:18] <BtbN> So for webm, live streaming services would have to transcode everything [22:19] <c_14> I personally just try and convince the server to give me an hls link and then watch it externally. [22:19] <BtbN> That's quite a complex process on twitch, but it's possible [22:20] <BtbN> It's quite common to watch ads on twitch though, and those only work when you watch via web [22:20] <c_14> It shouldn't really be that hard to splice those into the video though. [22:21] <BtbN> It is. As it's their flash video player which loads and handles them. [22:22] <BtbN> Even their own hls web player(Which only works on Android/iOS and Safari on Mac) doesn't show ads. [22:24] <c_14> Sure, but implementing ads for hls shouldn't be that hard. It'd be easiest if they showed ads to all clients at the same time though, not sure how they handle that. [22:26] <c_14> I mean, you'd just need to point the playlist entries at the ad files instead of the video files. Should even be possible on a per-user basis if each user got their own hls link. [23:24] <gcl_5cp> is posible edit metadata in original file? i mean, without make a new file like "ffmpeg -i org.mp4 ... new.mp4" [23:26] <c_14> No [23:26] <c_14> Unless you like hex editors. [23:39] <gcl_5cp> but in vlc i do manually [23:41] <c_14> What do you consider "metadata". There are some editors that can edit certain types of metadata, but ffmpeg does not support writing to the same input. [23:44] <gcl_5cp> you are right, is not the tool. python-mutagen seem to be perfect [00:00] --- Thu Feb 5 2015

1 0

ffmpeg-devel.log.20150203
by burek 04 Feb '15

04 Feb '15

[00:01] <BBB> it allows you to run strings binary | grep license: [02:36] <cone-436> ffmpeg.git 03Michael Niedermayer 07master:7620d48f2eab: avformat/network: Check for av_malloc* failures in ff_tls_init() [06:17] <cone-982> ffmpeg.git 03Rong Yan 07master:fc35df893175: avcodec/ppc/idctdsp.c: POWER LE support in idct_add_altivec() [08:58] <rcombs> ^ thoughts pls [14:26] <cone-982> ffmpeg.git 03Michael Niedermayer 07master:ed91c8651a87: avformat/mov: simplify pasp value before setting SAR [14:27] <wm4> man that tta stuff is weird [14:28] <wm4> and it interacts with the ffmpeg index stuff, which is even more weird [15:15] <cone-982> ffmpeg.git 03Michael Niedermayer 07master:63186ac5f12e: avutil/mem: replace remaining void **/*** casts by memcpy() [15:34] <cone-982> ffmpeg.git 03Matt Oliver 07master:22504396fe24: avformat/mpegtsenc: Fix build with msvc [15:35] <Daemon404> void ***.... scary [15:37] <nevcairiel> the indirection is strong in this one [16:16] <jfmcarreira> heyy guys [16:16] <jfmcarreira> there is any error concealment support in ffmpeg video decoders? is this a interesting topic for ffmpeg developers? [16:17] <cone-982> ffmpeg.git 03wm4 07master:6a0cd529a351: avformat/tta: fix crash with corrupted files [16:21] <iive> jfmcarreira: check libavcodec/error_resilience.c [16:24] <compn> jfmcarreira : yes, lots of ecc c [16:24] <compn> or whatever its called [16:24] <compn> :) [16:24] <jfmcarreira> iive: compn: i was just looking at the code [16:25] <jfmcarreira> i am doing reasearch in error concealment and error resilience and I was curious if it would any interest in ffmpeg libs [16:28] <michaelni> jfmcarreira, sure, all improvments to error concealment are welcome [16:30] <jfmcarreira> michaelni: the problem is understand how dat is stored in the ffmpeg libs. i noticed that h264 decoder is using the concealment functions. I am quite familiar with the HEVC and H264 decoder but only the reference software [16:55] <jfmcarreira> i was just trying to build ffmpeg and i am getting this error make: *** No rule to make target 'libavcodec/tpel_template.c', needed by 'libavcodec/h264qpel.o'. Stop. [16:56] <nevcairiel> i dont think that file has existed for a long time, did you have a old checkout you just updated or so? if so, do a make distclean or even better a git clean -xdf [16:58] <jfmcarreira> nevcairiel: make distclean might help [16:58] <jfmcarreira> thanks [17:23] <compn> jfmcarreira : like michaelni says, we are always interested in error concealment and contiuning to decode on broken files. just remember on irc we are on at different times, and you can always ask on the ffmpeg-devel mailing list about adding error concealment , you may find more people reply on the ml than on irc. (we are not all on both communication methods) [17:28] <jfmcarreira> compn: yeah thanks. I will take a look into the concealment code. it was interesting for me to add it to HEVC code [17:29] <jfmcarreira> i mean the ffmpeg implementation of the hevc codec [18:03] <ubitux> http://b.pkh.me/kdtree-try1.png http://b.pkh.me/kdtree-try2.png http://b.pkh.me/kdtree-try3.png ... [18:03] <ubitux> tree starting to get reasonable... slowly.. [18:04] <ubitux> still shitty though [18:22] <ubitux> http://b.pkh.me/kdtree-try4.png mmh [18:57] <wm4> I love that there's a ffio_read_varlen() function, which uses a specific variable length encoding [18:57] <wm4> and which is used by nutdec.c and mpc8.c [18:57] <wm4> and nothing else [19:01] <cone-982> ffmpeg.git 03Liviu Oniciuc 07master:b37858eae186: avformat/img2dec: remove the non-negative number requirement for start_number option [19:04] <michaelni> wm4, yes that function should be named differently [19:04] <michaelni> or be made more generic [19:26] <akira4> ubitux: update patch http://pastebin.com/4VQiqsRV. [20:18] <wm4> is AV_PIX_FMT_RGB32 the right pixel format, if the alpha channel seems to contain garbage? [20:18] <wm4> should the decoder zero-out the alpha and use RGB0? [20:18] <wm4> I have a camtasia sample that basically has randomly 0xFF or 0x00 as alpha [20:19] <wm4> as rectangular regions [20:37] <ubitux> akira4: 1. use sub.srt or something instead of video.mkv in the examples [20:37] <ubitux> akira4: 2. use AV_LOG_ERROR instead of 0 for av_log() [20:39] <ubitux> akira4: 3. actually, instead of these av_log, do ret = AVERROR(ENOMEM); goto end [20:41] <ubitux> akira4: 4. fix your style, it's still off (space after if/while, { on the same line) [20:41] <ubitux> akira4: rest LGTM [20:41] <ubitux> you can submit the patch with these modification directly to the mailing list [20:42] <akira4> okay. Thanks. [20:46] <akira4> ubitux: do I need to make an entry in the Changelog? [20:49] <ubitux> akira4: up to you [20:52] <kurosu> jamrial, the avx2 fix is x264's sandbox. Not sure what it implies, but that's probably several steps towards merge in x264 [20:52] <kurosu> as for the indentation, ok [20:54] <kurosu> not sure if it's worth sending a new one, but it looks final (your suggested improvements would be in another patch I guess) [20:54] <mraulet> hi kurosu thanks for fixing it [20:55] <mraulet> jamrial sse4 and avx2 are more or less same performance [20:55] <mraulet> on 8bits [20:55] <mraulet> 10bits gives us huge improvements [20:56] <mraulet> but we dont have those cross lane issues [20:58] <jamrial> is it worth adding the 8bit one, then? [21:00] <mraulet> I havent benchmarked them solely [21:01] <kurosu> the issue is also that the sizes are so much varying that you often need to split them a bit to notice things [21:02] <mraulet> I have improvements on hv (intrinsics version) but the choice that have been chosen by P Lepere mixing hv doesnt allow an easy avx2 implementation [21:02] <kurosu> iirc, mc intrinsics are on par with yasm for larger blocks, slower for smaller ones [21:02] <kurosu> that avx2 implementation seems to have made epel_hv 16-wide possible in sse4 [21:02] <mraulet> seperating h and v would be simpler to implement and give 5% improvements decoding time on basketballdrive qp27 [21:03] <mraulet> it works for e but not for q :-) [21:03] <kurosu> I have integrated the relevant changes but they wait in another patch [21:03] <mraulet> ok [21:03] <mraulet> let me try [21:04] <kurosu> btw is your comment about hv being done in a single step, without an intermediate buffer? if yes, indeed it was a bit unusual [21:05] <kurosu> of course bidir in hevc triples the amount of code, but vp9's mc asm is so much more compact ... [21:05] <mraulet> intrinsics version has this intermediate buffer [21:05] <mraulet> the asm implementation does not have it [21:06] <mraulet> it is computing h and v on the fly [21:07] <kurosu> the actual big stuff to do is port idcts, but that's a huge amount of work even translating it, let alone improving it [21:07] <mraulet> oh yes [21:07] <kurosu> it's an important piece, but I really don't want to touch it [21:08] <mraulet> didnt want to do it again [21:08] <mraulet> I did some refactoring on the intrinsic functions and it is a nightmare [21:08] <mraulet> you have to deal with the size of the code to have something efficient [21:09] <kurosu> the worst is that the most complex ones (16x16 and 32x32) are likely the more useful ones [21:09] <mraulet> it has huge effects on the performance [21:09] <mraulet> not sure they are more used [21:09] <mraulet> but as they are bigger ^^ they use more time [21:10] <kurosu> yeah, my point [21:11] <kurosu> actually, iirc, it is 16x16 above all else, 32x32 and 8x8 frequency depends on content/resolution [21:11] <mraulet> michaelni: if jamrial or kurosu are willing to lead the development of the hevc decoder I dont care& [21:11] <mraulet> I am supposed to not touch him to much now [21:12] <mraulet> *him > it [21:12] <kierank> mraulet: hehe [21:12] <kurosu> I don't want to lead nor really be active [21:12] <mraulet> oops I am not supposed ^^ [21:12] <kurosu> on its development [21:13] <jamrial> same, not interested. i'm just trying to port some of the missing asm [21:13] <mraulet> but you already are [21:13] <jamrial> and i don't know its inner workings enough to lead any kind of development [21:13] <kurosu> but there are like 2 or 3 ffmpeg devs interested in it, so it's easy being among the most active [21:14] <jamrial> why are you not supposed to touch it much, btw? [21:14] <mraulet> working for a company now [21:14] <mraulet> kurosu can mention it :) [21:14] <kurosu> people can look at your linkedin page, too ;) [21:15] <mraulet> they can for sure [21:15] <kierank> not an evil company so it's ok :) [21:15] <mraulet> kierank! :-) [21:18] <mraulet> kierank: before leaving I have some evil work to do :-) I need to fix interlace, I already have something more or less working but I need some feedback when it is a bottom field first > do I need to store this bottom field first with an offset? [21:19] <kurosu> what a nice parting gift [21:19] <nevcairiel> i thought giving the seperate fields out of the decoder was actually a fun effect [21:20] <kierank> I think in ffmpeg the first line is always the top field [21:20] <nevcairiel> hm i wonder how i can get the hwaccel to combine those fields [21:20] <kierank> but i might be wrong [21:20] <nevcairiel> kierank: its correct [21:20] <mraulet> ok [21:20] <nevcairiel> they are always in proper spatial order [21:20] <mraulet> so I do have something then [21:21] <nevcairiel> you would offset it by one line i guess, yeah [21:21] <mraulet> nevcariel I combine them when I output the frame, I copy the second field [21:21] <nevcairiel> ew [21:21] <mraulet> they are not always in the same order [21:21] <nevcairiel> cant you just decode into the same frame, and adjust the stride/linesize to always skip one line? [21:22] <nevcairiel> that way it may not break apart terribly for hwaccels either [21:23] <mraulet> I did want it to intrusive [21:23] <mraulet> and 4K wont deal with interlace so& a copy should be okay [21:24] <nevcairiel> but i cant copy hwaccel frames :( [21:24] <kurosu> nevcairiel, do you mean you can't mimic the behaviour mraulet wants to have for the software decoder ? [21:25] <nevcairiel> right [21:25] <kurosu> (or at a large cost) [21:25] <kurosu> just pointing that out, as I wasn't sure he thought of that [21:25] <kurosu> different behaviours for different decoders of the same format = not good [21:25] <nevcairiel> the only way a hwaccel would potentially deal with this is if it decodes both fields directly into the same frame [21:26] <nevcairiel> which a sw decoder can also do without too much harm with some linesize magic [21:28] <kurosu> how is interlacing indicated? sei messages and something else? because I wonder how h/w decoders deal with this (output separate fields or?) [21:29] <mraulet> sei message only [21:30] <kurosu> gotta go, goodnight [21:32] <nevcairiel> i'm not sure if the hwaccel defines support for this kind of interlaced mode at all, maybe they do expect single fields to be send to the renderer [21:37] <mraulet> what kind of information do you have as output? [21:39] <mraulet> the output is already POC ordered? [21:39] <nevcairiel> its directly integrated in the ffmpeg hevc decoder, it does the re-ordering, the hwaccel just replaces the actual slice decoding [21:41] <mraulet> so you are decoding the SEI and the slice header in sw so you can easily combine the output [21:41] <nevcairiel> but the output is opaque hardware surfaces :) [21:41] <mraulet> ah [21:42] <mraulet> if you dont send sei to the hwaccel then it wont combine anything [21:43] <nevcairiel> cant send that, it only takes slice data, or a struct with parsed out properties from sps/pps/vps and some slice header data [21:43] <nevcairiel> i dont suppose the sps/pps/vps indicate somewhere if its interlaced, just so that I could know to maybe not use hwaccel on this clip? [21:44] <mraulet> vps can have this information but it is optional [21:44] <mraulet> and you dont know which field comes first with the vps [21:44] <nevcairiel> well the individual field output isnt too bad, built-in bob deinterlace right there [21:45] <nevcairiel> :D [21:45] <wm4> except when it's done with progressive video... [21:45] <wm4> like vdpau gl interop does [21:46] <mraulet> I will extract the interlace code from openhevc and lets see what todo with it [22:02] <cone-276> ffmpeg.git 03Diego Biurrun 07master:75439bb31b3a: siff: K&R formatting cosmetics [22:02] <cone-276> ffmpeg.git 03Michael Niedermayer 07master:46a16ca15a68: Merge commit '75439bb31b3ab76f9a4d30c5de4ff9f407ec8128' [22:14] <cone-276> ffmpeg.git 03Diego Biurrun 07master:27b99d929fe3: siff: Return more meaningful error values [22:14] <cone-276> ffmpeg.git 03Michael Niedermayer 07master:deeba777ad0a: Merge commit '27b99d929fe3440b4bba8e5d970c2802f274f328' [22:22] <jamrial> mode change 100644 => 100755 libavformat/img2dec.c [22:22] <jamrial> mode change 100644 => 100755 libavformat/utils.c [22:22] <jamrial> i assume this wasn't intended, right? [22:23] <jamrial> commit b37858eae186ff84afd38cf3e845ba6e037e6397 [22:26] <cone-276> ffmpeg.git 03Hendrik Leppkes 07master:e3528d2a7bf2: mov: Implement parsing of the "HandlerName" from the MP4 HDLR atom [22:26] <cone-276> ffmpeg.git 03Michael Niedermayer 07master:a83e0639c0ce: Merge commit 'e3528d2a7bf29ba148d7ac1678552ce0089cd14f' [22:27] <nevcairiel> why do both the sps and pps in the hevc decoder have a scaling_list member, but only one of them is filled? :( [22:28] <michaelni> jamrial, fixed, thx [22:28] <cone-276> ffmpeg.git 03Michael Niedermayer 07master:e005dc621ab9: avformat: fix mode of img2dec.c and utils.c [22:30] <nevcairiel> hm, am i reading this right? the scaling list can be either in the pps or the sps? what wizardry is this [22:32] <kierank> yes [22:32] <kierank> in some of the fancier profiles it can be easier [22:35] <nevcairiel> weird [22:35] <kierank> not sure why [22:35] <kierank> either* [22:35] <nevcairiel> what is that useful for? do you refresh the pps more often than the sps or something? [22:35] <kierank> some encoders write a pps with every frame [22:35] <kierank> so i guess it allows per frame scaling list [22:36] <cone-276> ffmpeg.git 03Carl Eugen Hoyos 07master:67e9f3907d6d: isom: Decode Prores 4444 XQ with the existing Prores decoder [22:36] <cone-276> ffmpeg.git 03Michael Niedermayer 07master:94337f8c2dd9: Merge commit '67e9f3907d6d438a413df72d72c1339a0c11a290' [22:40] <fixxxermet> Could anyone fill me in on the status of ffserver? Is it still being maintained? [22:41] <JEEB> yes, barely'ish [22:41] <nevcairiel> well there we go, more complete patch [22:43] <fixxxermet> heh. [22:43] <fixxxermet> Any details JEEB? I've read a efw forum posts where people recommend not using it. [22:43] <fixxxermet> I'm considering integrating ffserver as a core component of zoneminder and I guess I'm wondering if that is a good idea at this point [22:43] <fixxxermet> I'd rip most of our own streaming code out and use what you guys have [22:44] <JEEB> fixxxermet, it gets a patch every now and then but generally it's only usable if you know how it works [22:44] <JEEB> a lot of stuff is supported by ffmpeg itself [22:44] <JEEB> the command line tool that is [22:44] <JEEB> so the need to actually use ffserver is limited to very specific scenarios [22:44] <fixxxermet> The 1 feed to 2 streams feature is really, really fantastic... But you're right, ffmpeg can do that [22:45] <fixxxermet> Well, I think it can anyway [22:45] <JEEB> yes, as far as I know it can :P [22:45] <fixxxermet> So in which cases do you see ffserver making sense? [22:46] <JEEB> a) if you know how to use it and b) something that ffmpeg itself can't do [22:46] <JEEB> I haven't used ffserver enough to know anything regarding the latter :P [22:46] <fixxxermet> heh, good answers. [22:50] <cone-276> ffmpeg.git 03Michael Niedermayer 07master:6f7a32839d1b: svq1enc: correctly handle memory error and allocations [22:50] <cone-276> ffmpeg.git 03Michael Niedermayer 07master:acfb4ede2f6b: Merge commit '6f7a32839d1b913be8170e91c3ac9816b314da21' [23:10] <cone-276> ffmpeg.git 03Carl Eugen Hoyos 07release/2.5:763e6ecf8336: lavc/aarch64: Do not use the neon horizontal chroma loop filter for H.264 4:2:2. (cherry picked from commit 4faea46bd906b3897018736208123aa36c3f45d5) [23:21] <cone-276> ffmpeg.git 03Luca Barbato 07master:9b8c8a9395c8: svq1dec: Validate the stages value strictly [23:21] <cone-276> ffmpeg.git 03Michael Niedermayer 07master:cba3a46e936c: Merge commit '9b8c8a9395c849639aea0f6b5300e991e93c3a73' [23:21] <cone-276> ffmpeg.git 03Michael Niedermayer 07master:741b56fceece: avcodec/svq1dec: Add assert to ensure "stages >= 0" [23:46] <cone-276> ffmpeg.git 03Luca Barbato 07master:a6653787a4b1: libopencv: Check kernel_str life cycle [23:46] <cone-276> ffmpeg.git 03Michael Niedermayer 07master:1dc1c4e7a82e: Merge commit 'a6653787a4b1e89170ff7962312dd110769c83a3' [23:56] <cone-276> ffmpeg.git 03Vittorio Giovara 07master:ec524ed12aa1: tta: Fix framepos and start_offset types [23:56] <cone-276> ffmpeg.git 03Michael Niedermayer 07master:bd12634d6b83: Merge commit 'ec524ed12aa1aeb37125203f1adf5aa10dfcb0de' [00:00] --- Wed Feb 4 2015

1 0

ffmpeg.log.20150203
by burek 04 Feb '15

04 Feb '15

[00:04] <hetii> hmm [00:04] <hetii> ha no windows:( [00:04] <hetii> *have [00:05] <klaxa> like i said, you could try wine [02:04] <fixxxermet> Is ffserver still maintained? [06:29] <teckris> hi guys [06:29] <teckris> can someone help me with ffmpeg libraries [06:30] <teckris> i am using FFMPEG with QtAV library [06:30] <teckris> in which i use streaming [06:30] <teckris> I need to have buffer used in streaming [06:32] <teckris> in functions AVFORMAT_CONTEXT is used by player [06:32] <teckris> so i need to control how the structure is filled for which i would like to get byte buffer [06:50] <Diaz> hi all, [06:50] <Diaz> I use libav 0.8.6 [06:50] <Diaz> for ape demux and ape decode [06:50] <Diaz> I got a big trouble [06:51] <Diaz> since ape decode error when some ape frame input [06:51] <Diaz> does any one have ideas? [06:51] <Diaz> about the layout of ape format? [06:51] <Diaz> HELP! [07:15] <Diaz> does anyone know where I can find the doc that describe ape header format? [07:15] <Diaz> HELP? [07:19] <Diaz> does anyone know where I can find the doc that describes monkey's audio format? [07:48] <pentanol> what's ape? [07:48] <pentanol> Diaz: [07:51] <Diaz> monkey's audio [07:51] <Diaz> do you ever use that? [07:51] <pentanol> no [07:52] <pentanol> I use audacity [07:52] <Diaz> i have big trouble about ape decode error for a long time when I use libav 0.8.16 [07:53] <Diaz> I do not find any doc that describe monkey's audio format [07:54] <pentanol> try ffprobe .ape file [07:54] <pentanol> I'm not sure this format is supported [07:55] <Diaz> ffprobe for what? [08:00] <pentanol> that tool from ffmpeg [08:02] <pentanol> yes that supported by libav D.A..S ape Monkey's Audio [08:03] <pentanol> only for reading those .ape files [08:08] <Diaz> just parse the ape file , right? [08:08] <Diaz> pentanol [08:13] <pentanol> I mean you can use ffmpeg for that in that programm or ffmpeg directly in cli ffmpeg -i file.ape -acodec mp2 -ac 2 -ar 192kb file.mp2 [08:17] <Diaz> ok, you mean I can use ffmpeg for compare [08:27] <rcombs> Diaz: this isn't #libav [08:28] <rcombs> topic [08:28] <rcombs> erm [08:28] <rcombs> well the topic doesn't actually mention that [08:35] <Diaz> OOOO, I can not write the question here? [16:14] <jfmcarreira> Heyy guys [16:15] <jfmcarreira> there is any error concealment support in ffmpeg video decoders? is this a interesting topic for ffmpeg developers? [17:02] <shevy> anyone happens to know some kind of audio filter, perhaps even in ffmpeg, that can filter away background noise? it is a .mp3 file recording of a speaker/lecture and there is some constant background noise (always the same noise) that I'd love to somehow get rid off [17:02] <fixxxermet> Could anyone comment on the state of ffserver? Is it maintained? [17:03] <kepstin-laptop> shevy: you'd probably want to look into using audacity to do that, it's a dedicated audio editor that, among other things, has filters designed to remove background noise. [17:07] <shevy> but I like ffmpeg more :( [17:39] <RobertNagy> When seeking is it possible to set the seek direction? i.e. I have a huge flv file 100GB and I need to seek to a position near the end, which currently takes a very long time. Alternatively is there some way to speed up seeking by e.g. seeking 99GB into the file before doing a slow seek? [17:39] <RobertNagy> note, I am using the -ss before -i option [17:55] <BtbN> RobertNagy, the problem is, ffmpeg has no information where in the file a specific timestamp is. So all it can do is read through the whole thing. [17:56] <RobertNagy> hm, and there is no way to give it a hint? [17:56] <RobertNagy> would creating a segmented file help? [17:56] <RobertNagy> i.e. is it possible to seek segments? [17:57] <BtbN> a container with a global header might be faster, but i have no idea how much information for example the mp4 header contains, and if it helps with seeking. [17:57] <RobertNagy> I'm seeking in the file while it is being written to [17:59] <BtbN> What ffmpeg version are you using? [17:59] <RobertNagy> latest [17:59] <RobertNagy> 2.5.3 [17:59] <BtbN> The -ss input option(in front of -i) is the fastest you can get then, unless someone knows some magic. [18:00] <BtbN> You can of course just cut a random section of of the flv file [18:00] <BtbN> Not entirely sure if that works with flv, but it certainly does with mpegts [18:00] <RobertNagy> well, not while it's being written to... [18:00] <BtbN> Why not? [18:00] <RobertNagy> how would I do that? the file is locked [18:00] <BtbN> locked? [18:01] <RobertNagy> the process writing to it... [18:01] <BtbN> Why would that stop another process from opening it? [18:01] <RobertNagy> nothing [18:01] <RobertNagy> but from modifying it [18:01] <RobertNagy> or maybe I'm misunderstanding something [18:01] <BtbN> Don't modify it then [18:02] <RobertNagy> what do you mean by "cut a random section" [18:02] <RobertNagy> btw, does any format support seeking from the back? [18:02] <BtbN> tail -c +50G somefile.flv > someotherfile.flv [18:02] <RobertNagy> oh, I see [18:03] <RobertNagy> hm, I don't think that will work in either flv or mpegs [18:03] <RobertNagy> will probably complain about invalid data [18:03] <BtbN> It certainly will for mpegts [18:03] <BtbN> not too sure about flv, but might work [18:03] <BtbN> If the media repeats the SPS/PPS sometimes [18:03] <BtbN> If there is only one IDR frame at the very beginning, you have to parse the entire thing [18:04] <BtbN> Just try it and see [18:04] <RobertNagy> yea, even if it works it won't work for what I need it [18:05] <RobertNagy> since I want to start reading the file before the final section [18:05] <BtbN> ...? [18:05] <RobertNagy> I'm doing a hack with -re + a small initial delay [18:05] <RobertNagy> to read from a growing file [18:05] <BtbN> i don't think ffmpeg will read past the end of the while from the moment it opened it [18:06] <RobertNagy> it does [18:06] <RobertNagy> atleast with flv [18:06] <RobertNagy> and mpegts, however I had problems with seek accuracy with mpegts [18:07] <BtbN> since ffmpeg 2.1 the -ss input option should be frame-accurate. [18:07] <BtbN> Also, i don't see your problem. [18:07] <RobertNagy> what part of it don't you see? [18:07] <RobertNagy> I can't cut a section that doesn't yet exist [18:07] <BtbN> tail -f -c +50G somefile.flv | ffmpeg -f flv -i - ..... [18:07] <RobertNagy> aah [18:07] <RobertNagy> I see [18:08] <RobertNagy> if tail works with a growing file that is [18:08] <BtbN> That's what -f is for. [18:08] <BtbN> seeking to an exact timestamp might get complicated with that method though [18:08] <RobertNagy> hmm [18:08] <RobertNagy> ok, what if I instead of recording to a huge file, recording into segments using the -f segment [18:08] <RobertNagy> would that help? [18:09] <RobertNagy> i.e. can I do frame accurate seek and playback of segments? [18:09] <RobertNagy> I guess I could figure out the frame accurate part by checking the segment start times and picking the right segment [18:09] <BtbN> sure, if you know the start timestamp of a segment, just seek the time-offset into the segment [18:09] <RobertNagy> however, will ffmpeg play segments? [18:10] <RobertNagy> however, will ffmpeg play segments flv? [18:10] <BtbN> If you use mpegts, you can just concat the files [18:11] <RobertNagy> this is getting complicated [18:11] <BtbN> mpegts is a very primitive format [18:11] <RobertNagy> would seeking in a large file possibly be faster with mpegts? [18:12] <BtbN> it's just a long line of 188 byte segments. [18:12] <BtbN> no, it has to parse the entire thing anyway [18:12] <RobertNagy> hm [18:12] <BtbN> but you can freely cut and concat mpegts files [18:12] <BtbN> it has no headers at all [18:12] <BtbN> so if you have a dir of 1h segments, you just concat the ones you want to play, and pipe them into ffmpeg [18:17] <RobertNagy> and what about the timestamps [18:17] <RobertNagy> will they always be 0 from the first segment? [18:17] <BtbN> They will be 0 from the segment you start at. [18:17] <RobertNagy> ic [19:55] <Schrostfutz> can I use ffmpeg to rename audiostreams in files? [20:13] <llogan> Schrostfutz: can you explain in more detail? [20:14] <Schrostfutz> llogan: I have video files with two languages, both lack labels so my playback device does not automatically select the correct one [20:15] <llogan> ffmpeg -i input -c copy -map 0 -metadata:s:a:0 language=sme -metadata:s:a:1 language=nor output [20:29] <Schrostfutz> llogan: thank you! [21:12] <Schrostfutz> llogan: can I savely use the same in and output file? [21:15] <llogan> Schrostfutz: you can't use the same file as input and output [21:16] <Schrostfutz> llogan: too bad [22:02] <vlt> Schrostfutz: Why would you want to do this?! Just remove your input file afterwards and rename the new one. [22:48] <courtarro> Hi all. I'm playing with ffserver for the first time and I'm unable to use the PixelFormat config variable. Am I doing something wrong? I simply have "PixelFormat yuv420p" in my <Feed> tag. The ffserver is version 1.2.6-7:1.2.6-1~trusty1 from an Ubuntu PPA [22:49] <courtarro> I get the error "Incorrect keyword: 'PixelFormat' " [23:52] <fixxxermet> Is it required that ffmpeg stream to a server, or can ffmpeg itself act as the server (from which a client, such as vlc, can directly connect)? [23:53] <c_14> As long as you only have 1 client you can open a listening TCP socket. [23:53] <fixxxermet> What if I have more than one client? [23:53] <c_14> Fixed number or dynamic? [23:54] <fixxxermet> dynamic, but I know the number ahead of time [23:55] <c_14> You _can_ use the tee pseudo muxer to create n tcp sockets, but it'll be easier to just use a dedicated server for that. [23:55] <fixxxermet> OK, so what about the actual syntax? [23:55] <fixxxermet> I seem to be missing it [23:55] <fixxxermet> (assume 1 client here) [23:56] <c_14> tcp://{iporhostname}:{port}?listen [00:00] --- Wed Feb 4 2015

1 0