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June 2013

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ffmpeg-devel.log.20130624
by burek 25 Jun '13

25 Jun '13

[00:02] <durandal_1707> yes [00:18] <BBB> why is ThreadFrame->progress an AvBufferRef? [00:19] <durandal_1707> huh? [00:19] <BBB> thread.h [00:19] <BBB> AVBufferRef *progress; [00:19] <BBB> why? [00:24] <durandal_1707> ask anton? [00:30] <durandal_1707> ask anton3 [03:11] <cone-758> ffmpeg.git 03Michael Niedermayer 07master:46312fc2a95a: cmdutils: dont change the log level for -report [03:50] <cone-758> ffmpeg.git 03Michael Niedermayer 07master:5dba888dd564: msmpeg4: Fix ff_msmpeg4_pred_dc() so it works with lowres>0 [07:45] <j-b> Good morning [08:52] <cone-758> ffmpeg.git 03Hendrik Leppkes 07release/0.10:5e6135f68d77: mathops/x86: work around inline asm miscompilation with GCC 4.8.1 [08:52] <cone-758> ffmpeg.git 03Hendrik Leppkes 07release/0.11:274ec187dcc6: mathops/x86: work around inline asm miscompilation with GCC 4.8.1 [08:52] <cone-758> ffmpeg.git 03Hendrik Leppkes 07release/1.0:491f2e517d92: mathops/x86: work around inline asm miscompilation with GCC 4.8.1 [08:52] <cone-758> ffmpeg.git 03Hendrik Leppkes 07release/1.1:24dc6b1a06d6: mathops/x86: work around inline asm miscompilation with GCC 4.8.1 [08:52] <cone-758> ffmpeg.git 03Hendrik Leppkes 07release/1.2:1065d4197e97: mathops/x86: work around inline asm miscompilation with GCC 4.8.1 [10:25] <cone-758> ffmpeg.git 03Diego Biurrun 07master:6dc6598692da: configure: Simplify an expression with enabled_all. [10:25] <cone-758> ffmpeg.git 03Michael Niedermayer 07master:1152863a3f1b: Merge commit '6dc6598692da3b0ebda2d768681786343f26a4f4' [10:48] <cone-758> ffmpeg.git 03Diego Biurrun 07master:ace87c19ed4c: configure: whitespace cosmetics [10:48] <cone-758> ffmpeg.git 03Michael Niedermayer 07master:14a61bc3f9ad: Merge commit 'ace87c19ed4c4882d7b9b3ce512c4c195e39a203' [11:11] <cone-758> ffmpeg.git 03Kieran Kunhya 07master:95d52464542f: lavc: Add option to encode MPEG-2 AAC with libfdk-aac [11:11] <cone-758> ffmpeg.git 03Michael Niedermayer 07master:af5f9c08763b: Merge commit '95d52464542f532061290192518d5fe1c1930e8d' [11:32] <cone-758> ffmpeg.git 03Rafaël Carré 07master:c3e58f8fb75d: matroskaenc: restore compatibility with non referenced AVPacket [11:32] <cone-758> ffmpeg.git 03Michael Niedermayer 07master:a620c8321efc: Merge commit 'c3e58f8fb75d8467161a65b85eb88281547ebab1' [11:48] <cone-758> ffmpeg.git 03Rafaël Carré 07master:e21307a2b024: lavf: don't abort if both encoder and muxer aspect ratios are not set [11:49] <cone-758> ffmpeg.git 03Michael Niedermayer 07master:f48366c7040a: Merge remote-tracking branch 'qatar/master' [12:52] <ubitux> wm4: not sure i'll update the smi demuxer/decoder soon, but sure feel free to share [13:04] <ubitux> haha the jpeg2k vs jpeg2000 thread is getting seriously ridiculous :)) [13:07] <durandal_1707> really? drama ended yesterday [13:08] <ubitux> durandal_1707: i'm a bit out of time :) [14:29] <cone-758> ffmpeg.git 03Michael Niedermayer 07master:a46e578ddacd: avformat/avio: Fix EOF handiling of ffurl_read_complete() [16:11] <cone-758> ffmpeg.git 03Michael Niedermayer 07master:bbe26eff2235: h264: Fix null pointer dereference with disabled error concealment [17:36] <khali> ah, one more bug found in the delogo filter.. [21:40] <wm4> so it seems libavformat's mp3 demuxer doesn't actually real embedded id3v2 tags [21:41] <wm4> s/real/read/ [21:42] <wm4> there's only code to skip id3v1 tags (maybe because these are typically at the end of the file and can't be skipped as header?) [21:44] <Daemon404> doesnt everybody use a separate lib for mp3 stuff [21:44] <Daemon404> id3lib or w/e it's called [21:45] <wm4> libavformat appears to have its own id3 code [21:46] <Daemon404> ive gotten lots of bug reports (via CCCP) about lavf's mp3 demuxer [21:46] <Daemon404> so much i've just deemed it a crapshoot [21:46] <JEEB> lol, so far it's one of the least worst @ DirectShow [21:46] <wm4> it's always wonderful when I try to replace rotten mplayer code with ffmpeg functionality, only to find that mplayer's code worked better [21:46] <JEEB> it's really the MS splitter that is derp [21:47] <JEEB> which MS for whatever reason still keeps around [21:47] <JEEB> even though even WMP hasn't used it for years [21:47] <JEEB> and thus people complain "but this works in WMP!!!!one-eleven" [21:48] <wm4> also av_dict_get() sure is a weird API (you iterate all keys by passing key=""?) [21:48] <durandal11707> wm4: id3 reading thing is not in demuxer itself [21:48] <wm4> durandal11707: then where is it? [21:48] <wm4> oh, utils.c? [21:51] <wm4> I only see code that reads id3 on the start of the file as part of probing and so on [21:58] <durandal11707> yes, so what you try to acomplish? [22:00] <wm4> durandal11707: well, first it would be nice if it could recognize and parse embedded id3v2 tags... id3v2 tags can be anywhere, not just the start of the stream, and e.g. web streams appear to use them to update info about newly played song [22:00] <wm4> +s [22:00] <wm4> durandal11707: second, I'd like to add shoutcast support... while shoutcast metadata is really simple to read, transferring the info from stream layer to demuxer is not [22:01] <wm4> I thought about injecting artificial id3 tags, but even then I can't align them on packet boundaries [22:02] <durandal11707> perhaps you could add to shoutcast metadata some special header so you could make use of it in demuxer.... [22:05] <wm4> handling shoutcast directly in the demuxer would probably be easier [22:06] <durandal11707> but how would you recognise it? [22:07] <wm4> well, the shoutcast protocol inserts a variable sized metadata header each N bytes [22:08] <wm4> what would your "special header" do instead? [22:11] <durandal11707> and can you recognise such metadata header? [22:12] <wm4> no, it's implicit (you have to count the bytes you receive) [22:13] <wm4> if you request shoutcast metadata, it will reply with a Icy-MetaInt header in the http response [22:13] <wm4> Icy-MetaInt is an integer that sets the data payload size [22:13] <wm4> so all Icy-MetaInt bytes you get a metadata packet [22:14] <wm4> with consists of a single byte for the length of the header [22:14] <wm4> then you can read that header, and after that you get Icy-MetaInt data bytes again [22:14] <wm4> that's it [22:27] <wm4> also if I concat two mp3 files, libavformat returns the same length as for the first original file [22:27] <wm4> weird [22:39] <durandal11707> wm4: concat with what? [22:40] <wm4> cat [22:40] <durandal11707> your own fault than.... [22:40] <wm4> really? I thought mp3 contains no timestamps [22:46] <wm4> well, maybe it's just the best to let the user read the icy metadata as some sort of out of band data [22:47] <wm4> so you could get av_opt_get() it [22:49] <wm4> yeah, probably much better than making up weird hacks [22:49] <durandal11707> that is ugly, av_opt_get is for options and not for metadata [22:54] <wm4> durandal11707: mime type is already "returned" via av_opt_get [22:57] <wm4> and other ideas, like id3v2 injection, don't sound like they'll work any time soon [23:55] <cone-409> ffmpeg.git 03Michael Niedermayer 07master:8a7aabe80b9b: avfilter/vsrc_testsrc: fix artifacts with odd height [00:00] --- Tue Jun 25 2013

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ffmpeg.log.20130624
by burek 25 Jun '13

25 Jun '13

[00:00] <bencc> I don't need to record audio with html5 [00:00] <bencc> all I need is to transcode a live rtmp stream to mp3 and broadcast it [00:00] <klaxa> it might make things easier though [00:00] <bencc> how come? [00:00] <klaxa> no flash, no rtmp [00:01] <klaxa> hmm... although, it might not be as convenient, since i'm not sure how to move the audio from the recording browser to a server [00:01] <klaxa> send it over a socket i guess [00:01] <bencc> there is no echo cancelation [00:02] <klaxa> mmh [00:02] <bencc> webrtc might work but it's not widly supported yet [00:02] <bencc> and it's hard to work with some clients using webrtc and other flash [00:02] <bencc> transcoding to mp3 shouldn't be that hard. that's what ffmpeg is for... [00:05] <klaxa> well right now the only hard problem seems to be to get the audio from the client to ffmpeg [00:06] <bencc> maybe [00:06] <bencc> but on my server it seems to work [00:07] <bencc> I just can't get it to stdout [00:09] <klaxa> you can't even record it! [00:09] <klaxa> ffmpeg obviously does not start encoding, which in turn means it's not even receiving anything properly [00:11] <bencc> for you [00:11] <bencc> I managed to get an mp3 file but not to stream it [00:13] <klaxa> so you got an mp3 and you could listen to it? [00:13] <klaxa> try to reproduce [00:14] <bencc> yes [00:14] <klaxa> can you reproduce and pastebin all the logs [00:14] <klaxa> ? [00:15] <bencc> I'll try now [00:20] <bencc> klaxa: http://dpaste.com/1265946/ [00:20] <bencc> there is test.mp3 file but I'm not able to play it [00:20] <klaxa> and you ca-- [00:20] <klaxa> yes just what i though [00:20] <klaxa> it's not actually recording anything [00:21] <klaxa> *thought [00:25] <bencc> klaxa: I used a wrong url [00:26] <bencc> now I can save test.mp3 and I see the log of ffmpeg [00:26] <bencc> but I can't play the mp3 file [00:26] <bencc> klaxa: http://dpaste.com/1265950/ [00:28] <bencc> klaxa: now I'm getting packets in the shell [00:28] <bencc> weird [00:32] <bencc> klaxa: something is weird with the pipes [00:33] <bencc> it buffers the output and when I'm closing the media server suddenlty I see many asci messages? [00:33] <bencc> without the ? [00:33] <klaxa> huh? that's weird indeed [00:35] <klaxa> try running it with "stdbuf -o0 -e0 -i0" i.e. stdbuf -o0 -e0 -i0 ffmpeg -i <all the other stuff> [00:35] <bencc> ? [00:35] <bencc> what is it? [00:35] <klaxa> sets buffer values for pipes to 0 [00:36] <bencc> the same [00:37] <klaxa> ah hmm... i just saw, you might try your luck with increasing analyze duration and stuff [00:38] <bencc> ? [00:38] <bencc> how? [00:38] <klaxa> add "-analyzeduration 10000000" or something [00:40] <bencc> same [00:41] <bencc> something is wrong with ffmpeg [00:42] <bencc> I'm sure [00:42] <klaxa> file a bugreport then [00:43] <bencc> ok [01:10] <bencc> klaxa: the same happens with avconv. when I'm stopping the media server I see output in the terminal [03:25] <lonestarr> A question regarding RTSP client (connecting to an IP camera reading H.264 directly via RTSP) ... is there a way to limit the bandwidth to say 64 or 128 kbps in the client? Spent quite some time googling but did not find any information on this. [03:28] <lonestarr> ffmpeg -maxrate 64k -b:v 64k -sn -an -i "rtsp://admin:admin@x.x.x.x:80/videoSub" does not work unfortunately - looking for limiting network bandwith somehow, any ideas? [03:35] <sacarasc> lonestarr: Only if the server supports that. [04:06] <ten1> I have a quick and perhaps stupid question: what happens if you try to convert a video to the same format it is already in? [04:07] <llogan> ten1: depends on the format. [04:07] <sacarasc> ten1: You'll lose quality, in most cases. [04:07] <sacarasc> Unless you're going from lossless to lossless. [04:07] <ten1> sacarasc: kay. I'll try and avoid that then [04:08] <ten1> llogan, sacarasc, thanks [04:09] <llogan> what are you trying to do? [04:10] <ten1> llogan: video upload/transcode for web players [04:10] <sacarasc> YouTube transcodes all videos uploaded to it, and people seem to not care about that. [04:11] <ten1> using gearman, gearman-php, ffmpeg, ffmpeg-php [04:11] <sacarasc> ffmpeg-php is years out of date and doesn't work with any recent ffmpeg. [04:11] <ten1> then it should work great with what's in the debian repos [04:12] <ten1> but I suppose I can just use shell commands instead [04:13] <llogan> debian doesn't use ffmpeg [04:14] <ten1> llogan: come again? [04:18] <ten1> llogan: the package aptitude lists has Maintainer: Debian Multimedia Maintainers <pkg-multimedia-maintainers(a)lists.alioth.debian.org> [04:20] <ten1> but I'll have to look into this more carefully [04:21] <llogan> i guess ffmpeg package in squeeze may still be FFmpeg upstream [04:21] <llogan> not squeeze backports though [04:24] <ten1> llogan: this sounds like a minefield. But, so the php wrapper is discouraged? I should be invoking it directly? [04:25] <sacarasc> Yes. [04:26] <ten1> all right. I very much appreciate the information and advice, and I am very pleased that I stopped by here [04:27] <llogan> good luck with the project [04:27] <ten1> thanks :) [04:38] <knIOO> I'm getting [Parsed_concat_0 @ 0245e740] Failed to configure output pad on Parsed_concat_0 - full output is here http://pastebin.com/Tzqpinjf [04:39] <knIOO> I'm trying to join two videos of the same framerate and the same resolution [04:39] <knIOO> Can someone let me know what's going on? [04:39] <knIOO> I'm getting [Parsed_concat_0 @ 0245e740] Failed to configure output pad on Parsed_concat_0 - full output is here http://pastebin.com/Tzqpinjf [04:39] <knIOO> apologies, still getting used to this new irc client heh [04:39] <klaxa> unknown paste id [04:39] <knIOO> argh, apologies, one moment [04:41] <knIOO> klaxa: sorry about that, here is my current output http://pastebin.com/M1B7YSix [04:43] <klaxa> hmm you're using quite some parameters i'm not too familiar with :X [04:44] <knIOO> no problem heh - my goal is to join two videos of unknown codec together basically [04:44] <knIOO> but I thiiink there's a mismatched aspect ratio somehow even though they're both 1280x720 [04:44] <knIOO> I just don't know how to set an aspect ratio for the streams, something to use with setsar/setdar but I have no idea how to use it [04:45] <knIOO> can't find much documentation on the net about it either [04:45] <klaxa> well i guess you've already come across this: http://ffmpeg.org/trac/ffmpeg/wiki/FilteringGuide but this might be the best guide you'll find [04:46] <llogan> knIOO: you have a filter in your input link label: [0:1 -setsar=1:1] [04:46] <llogan> change it to [0:1] [04:46] <knIOO> llogan: same error - just noticed that is an old version of the output, get the exact same error without it tho [04:47] <knIOO> that was just me experimenting on my own [04:47] <llogan> ah, i see. sar 0:1. i've seen another user who experienced this but i forgot how it was resolved. [04:48] <knIOO> llogan: it might have been me heh, I was here asking the same question a few weeks ago [04:49] <knIOO> I just have no idea how to set the sar value for either stream [04:49] <llogan> https://ffmpeg.org/trac/ffmpeg/ticket/2456 [04:49] <llogan> youre on the right track, but you're implementing incorrectly [04:50] <knIOO> yep - also heh I actually added some info to that bug back in may [04:51] <knIOO> I just can't find anything on how to set filters to the individual streams (if I'm understanding the problem correctly) [04:51] <llogan> ffmpeg -i tojoin.mkv -i perftest1.avi -filter_complex "[1:0]setsar=1/1[sarfix];[0:0] [0:1] [sarfix] [1:1] concat=n=2:v=1:a=1 [v] [a]" -map "[v] -" -map "[a]" -vcodec libx264 output.mkv [04:52] <llogan> theres a typo in that [04:52] <llogan> -map "[v]" -map "[a]" [04:53] <knIOO> that..seems to have worked! [04:54] <lonestarr> very sorry everyone, asked the question regarding limiting network bandwidth for ffmpeg as RTSP client about 90 minutes ago but had a power failure and can't see any history. Was there a solution for this? Is there a way to throttle the traffic somehow? [04:55] <knIOO> llogan: you are a hero - that's fixed the problem completely. [04:55] <knIOO> thank you so much [04:56] <llogan> so the filterchain translates as "take the first stream from the second input, and apply setsar and name output link label [sarfix]" [04:57] <llogan> read the following 5 times and then you'll get the hang of it: http://ffmpeg.org/ffmpeg-filters.html#Filtergraph-syntax-1 [04:57] <knIOO> this looks interesting [04:58] <llogan> also http://ffmpeg.org/ffmpeg-filters.html#concat [04:59] <llogan> lonestarr: "<sacarasc> lonestarr: Only if the server supports that" [05:00] <lonestarr> okay, thanks for letting me know. So if the server does support it, how do I request a stream at say 64 kbps? [05:02] <lonestarr> -maxrate 64k or -b:v 64k or something else? I tried this, but it was still downloading the stream at 500 kbps regardless. [05:05] <sacarasc> It's usually a slightly different URL, like rtmp://blah/feed/64 or something. [05:08] <knIOO> llogan: so looking at -filter_complex, am I correct if I am understanding that [0:0] for example represents 1st input, video stream, and [1:1] represents second input, audio stream - if the containers just contain one video and one audio stream [05:09] <llogan> yes. usually. unless the first stream is audio. [0:0] just represents first input, first stream of that input. [05:10] <knIOO> that makes sense, for most cases is it video first stream, audio second or is it just completely random? [05:12] <llogan> most cases, but i've seen some inputs were audio is listed first. or you could probably just use [0:v:0] to be sure: first input:video:first stream that is video [05:12] <llogan> http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1 [05:13] <llogan> http://ffmpeg.org/ffmpeg.html#Advanced-options (see -map for examples) [05:13] <knIOO> you've taught me more in 10 minutes than I've learnt in the few months I've been playing with ffmpeg [05:14] <llogan> it's a good distraction from a boring task that i am postposing. [05:16] <knIOO> Hahah dare I ask [05:17] <llogan> fixing someone elses ancient PHP code. [05:19] <lonestarr> okay, thanks for your help / time guys [05:19] <llogan> knIOO: apparently that user that experienced sar 0:1 was you, and the sarfix example worked for you last time. [05:21] <knIOO> llogan: I worry about my short term memory sometimes [05:21] <knIOO> looking at my own bug report i even listed the fix..facepalm [05:27] <knIOO> and oh God I feel your pain llogan, I find php a language that is particularly irritating to edit/add to after the fact [05:27] <knIOO> curly braces can go to hell [08:14] <ilove11ven> lonestarr: I think you have throttled the traffic on the wrong side. [08:41] <enelar> guys, i am trying to emerge ffmpeg in ARM gentoo. but i get compilation error [08:41] <enelar> i am already tried to install older version, the same [09:47] <ink3> Hello [09:48] <ink3> Can you help with 'burn' subtitles? http://pastebin.com/U7JxeJDC [11:22] <LordDoskias> when i'm muxing stuff with libav but i have already the encoded data (audio/video) do i still need to set up codecs in the output context etc? [11:24] <JEEB> libavformat would be the one you'd be needing in that case [11:24] <JEEB> and I think that nowadays takes in AVPackets or so? not sure [11:32] <LordDoskias> i've been reading the muxing example [11:32] <LordDoskias> but there they setup codecs and whatnot [12:03] <LordDoskias> i have 2 questions: 1. how exactly does av_rescale_q rescale numbers e.g. what is the end result? 2. where can i set the framerate of the resulting container after muxing? [12:43] <LordDoskias> is avformat_alloc_output_context2 a new functions, since in my distribution of avlib i do not have it ? [12:50] <LordDoskias> which is the current stable version of libav 0.9 or 0.8 branch? [12:54] <sacarasc> #libav [12:54] <sacarasc> #ffmpeg deals with ffmpeg, not libav. [12:54] <LordDoskias> okay, thanks [12:55] <LordDoskias> i thought they were practically identical since ffmpeg uses avlib underneath, no? [13:00] <sacarasc> Read that. [13:12] <LordDoskias> but why is that that when i read: https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/muxing.c the includes point to libavformat/libavcodec etC? [13:19] <LordDoskias> on the other hand, how can i use ffmpeg's distribution of libav where for example the function avformat_alloc_output_context2 is implemented? [13:33] <saste> LordDoskias, you're confusing libav* libraries and Libav project [13:33] <saste> libavcodec, libavformat, etc. are included in FFmpeg (the project) as well [13:33] <LordDoskias> and you apparently do some other stuff with it [13:33] <LordDoskias> e.g. add functinos [13:34] <saste> and the API is mostly equivalent, since FFmpeg merges all changes from Libav [13:34] <LordDoskias> but you also add your own funcs [13:34] <saste> on the other hand Libav doesn't do the same with FFmpeg, so in general it has less features, including less functions [13:35] <LordDoskias> so what do i have to do if i want to use ffmpeg in my project programatically ? [13:35] <LordDoskias> e.g. use its apis not the command line tool [13:36] <bencoh> use the api described in ffmpeg doxygen [13:39] <LordDoskias> i do not have a function described there [15:29] <Onkeltem> Hi. I forgot the name of GUI utility for editing contents of MP4 containers. Any thoughts? [15:33] <Mavrik> mp4box? [16:00] <setkeh> hey guys im starting ffmpeg with "ffmpeg -f pulse -i default -f x11grab -acodec flac -r 60 -s 1920x1080 -i :0.0+1360x768 -vcodec libx264 -crf 0 -preset ultrafast out.mkv" but i cannot figure out why the sound is not in sync with the video any idea's ?? [16:27] <smoke_> hi im running ubuntu 13.04 and im having trouble with dvdstyler and ac3, i read on the ubuntu forums ot get the newer version of ffmpeg is there a specific ppa i should grab or will the jons one do? [16:27] <Mavrik> get a static build [16:27] <Mavrik> ppas are also mostly obsolete [16:28] <smoke_> Mavrik, sorry whats a static build? [16:29] <smoke_> Mavrik, in the official repo for ubuntu? [16:29] <Mavrik> no. [16:29] <Mavrik> forget about official repos. [16:29] <Mavrik> http://dl.dropboxusercontent.com/u/24633983/ffmpeg/index.html [16:30] <smoke_> k thx [16:31] <setkeh> hey guys im starting ffmpeg with "ffmpeg -f pulse -i default -f x11grab -acodec flac -r 60 -s 1920x1080 -i :0.0+1360x768 -vcodec libx264 -crf 0 -preset ultrafast out.mkv" but i cannot figure out why the sound is not in sync with the video any idea's ?? [16:39] <setkeh> any one ?? [16:39] <klaxa> it's a common issue, blame pulse maybe? [16:39] <klaxa> just shift the audio after recording [16:39] <klaxa> shouldn't be that hard [16:39] <setkeh> its worked for me before :/ [16:40] <setkeh> i would rather a fix for the problem then some cheap hack [16:41] <klaxa> i'm amazed you can write 60 fps at fullhd even [16:43] <setkeh> the fps isent causing the issue i have tried 10 fps 20 40 and 60 same problem [16:43] <setkeh> and i get around 100-130MB/s iop out of my drives [16:46] <klaxa> have you tried alsa instead of pulse? [16:47] <setkeh> yeah same issue [16:48] <klaxa> well just post-process the audio delay, it's not a big deal [16:48] <klaxa> i doubt you will publish your recording right away anyways [16:49] <setkeh> well thats the idea of fixing the issue and not some cheap hack [17:20] <jure> hullo [17:21] <jure> I've managed to transcode some MLP audio to FLAC, but it ended with a message "lossless check failed" [17:21] <jure> is this significant? [17:25] <jure> ok never mind [17:25] <jure> probably because I didn't use concat [18:10] <Mavrik> BigWhale, asm ja, bi alo :) [18:30] <sacarasc> sdl240: You probably wanted: [18:31] <sdl240> sacarasc, thats exactly what i want! [18:31] <sdl240> thanks [18:35] <bencc> klaxa: installing red5 to see if the problem is on ffmpeg side or the media server side [18:35] <klaxa> alright [18:35] <bencc> klaxa: if red5 won't work I'll try FMS free starter version [18:35] <bencc> it must work with FMS :) [19:13] <bencc> klaxa: now I'm having truble with red5. this is redicolus... [19:13] <bencc> I mean I can't get red5 to work. not related to ffmpeg [19:34] <bencc> klaxa: I have red running. trying to transcode to a file [19:34] <klaxa> ok [19:48] <bencc> klaxa: I'm getting Operation not permitted error when trying to connect to red5 rtmp stream [19:49] <bencc> klaxa: others see this error too but I havn't found a solution yet [19:50] <bencc> http://ffmpeg.org/trac/ffmpeg/ticket/138 [20:08] <sisco12> Hi all [20:09] <sisco12> im facing a little problem here with ffmpeg any expert here can help so i can post my commands ? [20:10] <sisco12> ....??? [20:10] <josemanuel_> just ask [20:11] <sisco12> ok here we go [20:11] <sisco12> when i use ffplay with this command :ffplay "rtmp://stream.marociptv.com/livepkgr/livestreamAlAoulaInter?9700d88cc84a668… swfUrl=http://www.marocstream.com//e mbed/player.swf pageUrl=http://www.marociptv.com/tvaccess.cfm?tv=5E4B50B9959C61EA4AC18FD3_7… live=1 swfVfy=1" [20:11] <sisco12> it works [20:12] <sisco12> but when i try to use ffmpeg to get it as input is not working [20:13] <sisco12> ffmpeg -i "rtmp://stream.marociptv.com/livepkgr/livestreamAlAoulaInter?9700d88cc84a668… swfUrl=http://www.marocstream.com //embed/player.swf pageUrl=http://www.marociptv.com/tvaccess.cfm?tv=5E4B50B9959C61EA4AC18FD3_7… live=1 swfVfy=1"-vcodec libx264 -acodec mp3 -f flv 22.mp4 [20:13] <sisco12> i got this error : HandleCtrl: Ignoring SWFVerification request, use --swfVfy! rtmp server sent error rtmp server requested close [20:14] <sisco12> do u have any idea where it stuck? [20:16] <sisco12> josemanuel_ have any idea? [20:16] <sisco12> have u* [20:17] <josemanuel_> have you used --swfVfy? [20:17] <josemanuel_> as the error message suggests [20:17] <sisco12> well the command is sfwVfy=0 or 1 [20:18] <josemanuel_> have you tried both to see if any works? [20:18] <sisco12> yes i did [20:18] <sisco12> ffplay works and ffmpeg no thats weird [20:18] <sisco12> i mean the same link [20:21] <sisco12> i got ty [20:21] <sisco12> its ok :D [20:37] <mykul> hi folks, i'm using amix to multiplex several audio files, and I've noticed that the final mixdown has significantly reduced volume [20:37] <mykul> is the gain getting normalized somehow? [20:37] <mykul> is there a way to prevent this or counteract it? [21:47] <RSDRSDRSD> I am having some huge problems of making screenshots with ffmpeg from a wowza media server [21:47] <RSDRSDRSD> problem is listed on http://pastebin.com/3KbGyurd [21:47] <RSDRSDRSD> can someone please help me? [22:35] <RSDRSDRSD> What is the best methog to create thumbnails from rtmp streams [22:35] <RSDRSDRSD> if I want a thumbnail of the current live stream [22:36] <bencc> anyone is able to get rtmp stream from red5 [22:41] <blink> Hi, I'm running into an issue using 20130611-git-0f88a98-win64-static, output codec is prores quality 2. When I create a file using an image sequence and define a framerate of 23.976, it not only rounds up to 28.98 (not critical) but drops frames in so doing. I see the debug saying 25fps; is this a known issue or have I done something incorrectly? [22:42] <blink> pardon, input is an image sequence. output is mov container, prores 422 essence [22:56] <JEEBsv> blink: image sequence might need -r before -i, and you should be able to set a proper X/Y frame rate, 24000/1001 [22:56] <JEEBsv> which is what "23.976" is [23:09] <JustinInTheFlesh> I have a problem that I believe has to do with directshow. I changed libavformat\rtp.c to have a few custom payload types due to a device I am trying to stream to. Whenever I capture from directshow to a file, the quality is perfect. From a file, I can stream to multicast and it is played without an issue. If I directly stream from dshow, I get a lot of dropped packets. However, if I stream to a multicast, use a second [23:09] <JustinInTheFlesh> version of ffmpeg to catct the stream and rebroadcast without re-encoding, the device plays the stream without issue. [23:09] <JustinInTheFlesh> And suggestions? [23:12] <JustinInTheFlesh> Anyone that can assist? [23:26] <blink> yes, I know that I can pass it a fraction; is that the preferred way to express it for ffmpeg? [23:27] <blink> thanks JEEBsv, i'll try -r before i [00:00] --- Tue Jun 25 2013

1 0

ffmpeg-devel.log.20130623
by burek 24 Jun '13

24 Jun '13

[00:03] <BBB> positions in yuv file of first mismatch for a variety of cases I've just looked at: 3365648, 3386080, 3370736 - this is all frame 29.something, so it's always the same frame that fails. so now I just pick any of these, make sure the first $thatnumber+a few bytes are identical between the file and the one I'm tracking, and I'm good to go for printf debugging (cmp -n number file.yuv ref.yuv) [00:04] <ubitux> BBB: that stuff sounds relevant for your blog btw [00:05] <matthewjheaney_> @ubitix: I just pushed up the 1st of 2 patches [00:06] <BBB> I could blog about it I guess... I don't blog very much though [00:06] <BBB> let me try to fix it and then blog about the fixc [00:09] <BBB> darn it it refuses to fail now :) 4.4k runs [00:10] <BBB> maybe I shouldn't play youtube [00:39] <Compn> j-b : i got a report from a user that mplayer doesnt run on mavericks either. [00:40] <Compn> but he wouldnt tell me what version of mplayer he was using [01:03] <BBB> ubitux: so the faulty mb for the case I'm tracking (3370736 first mismatch) is frame 29 (counting from zero as first, not counting invisible frames - ARFs), mb_x=11, mb_y=6 [01:04] <BBB> then print data pre and post loopfilter at that mb until you get the mismatch we're tracking, then you know if it's b/c of loopfilter or not [01:04] <BBB> then backtrace from there [01:04] <BBB> I'm betting it's not loopfilter but better be sure [01:44] <cone-319> ffmpeg.git 03Michael Niedermayer 07master:bcb42fb6dbef: sonic: use av_calloc() [01:44] <cone-319> ffmpeg.git 03Michael Niedermayer 07master:4ec7ef56bd03: sonic: simplify quant cliping [01:44] <cone-319> ffmpeg.git 03Michael Niedermayer 07master:730e07f10b97: sonic: avoid float sqrt() for integer input & output [05:36] <BBB> ubitux: ah found it [05:36] <BBB> somebody broke segmentation [05:36] <BBB> that was relatively easy [05:36] <BBB> let me do like 100k runs or so and make sure it works now [12:20] <khali> saste: thanks for reviewing my delogo patch, I'll benchmark it in the afternoon and reply as soon as I have the numbers [12:57] <cone-758> ffmpeg.git 03Luca Barbato 07master:1e340af8d6a9: avconv: drop additional strerror fallback [12:57] <cone-758> ffmpeg.git 03Michael Niedermayer 07master:d894e64acca9: Merge commit '1e340af8d6a97cc013a2ad8ba77c77129625a34f' [13:01] <cone-758> ffmpeg.git 03Luca Barbato 07master:42cc6cefd315: avconv: report the error for codec open failure [13:01] <cone-758> ffmpeg.git 03Michael Niedermayer 07master:cc61ef0479f0: Merge commit '42cc6cefd315c1556e2a52f7ebe2f766ec82b790' [13:46] <cone-758> ffmpeg.git 03Luca Barbato 07master:f963f701d90b: ogg: relax demuxer conformance checks [13:46] <cone-758> ffmpeg.git 03Michael Niedermayer 07master:780b1aa1efa8: Merge commit 'f963f701d90bd7bb03e39aab4e59bd137084e082' [13:51] <cone-758> ffmpeg.git 03Alex Smith 07master:f11e4045b9c4: configure: More msvc/icl combining [13:51] <cone-758> ffmpeg.git 03Michael Niedermayer 07master:fb0df5c113e0: Merge remote-tracking branch 'qatar/master' [14:29] <ubitux> matthewjheaney_: can you send the second one so i can push them? [14:29] <ubitux> BBB: cool, thx :) [16:49] <BBB> nobody review my patch :( [16:57] <BBB> ubitux: go review my patch! [17:02] <ubitux> BBB: you realize i never contributed to anything vpx related, right? :P [17:02] <ubitux> my biggest implications in it was encoding a video in webm, and sending you an email [17:17] <BBB> ubitux: I don't think anyone else here is particularly better suited to review any patch at all [17:17] <BBB> ubitux: I could review it, but I already wrote it so it defeats the 4eye rule [17:18] Action: ubitux feels like naming random people to run away [17:18] <ubitux> well, i can have a look, but that will likely be a superficial review just to make you feel better [17:19] <ubitux> OTOH i can spend some quite time on it, which i might not have right now [17:19] <BBB> if only you worked for a company that cared about opensource software to such an extend that you could spend reasonable time on making it better, right? [17:20] <ubitux> i have no job right now, no company is taking my time [17:20] <BBB> how do you not have time then? [17:20] <ubitux> i'm just busy with my own stuff [17:20] <BBB> ok :) [17:20] <michaelni> BBB the patch looks ok [17:20] <BBB> ah review :) [17:20] <BBB> thanks [17:21] <michaelni> should i apply or do you want to apply it ? [17:21] <BBB> you apply is easier [17:21] <BBB> I'm mostly used to gerrit nowadays [17:21] <wm4> ubitux: so you remember that libavcodec subtitle converters throw away the "decoded" text if it's not valid utf-8, right? [17:21] <cone-758> ffmpeg.git 03Ronald S. Bultje 07master:c329713de7c6: vp8: wait for prev_frame to parse segment_map before reading it. [17:22] <wm4> ubitux: I think it's ridiculous... it leads to the situation subtitle events that happen to contain badly encoded stuff are ignored, while others are displayed normally [17:22] <ubitux> wm4: i don't like it either, as i said in the thread [17:22] <wm4> ubitux: and considering how often this kind of stuff happens, like english subs that contain a badly encoded accent in _some_ events [17:23] <ubitux> i just couldn't come with a relevant argument at that time, so i let Nicolas do what he prefered [17:23] <wm4> ubitux: I'd say it's really damn offensive to the user [17:23] <wm4> ubitux: however printing a warning would be perfectly fine [17:23] <ubitux> i agree [17:23] <ubitux> a patch should be trivial; can you send one (as a RFC eventually) explaining the issue you have? [17:24] <wm4> ok [17:24] <ubitux> thank you [17:24] <wm4> I have some more question [17:24] <ubitux> wm4: keep in mind the main argument was to prevent creating invalid files [17:24] <wm4> so some subtitle formats are framebased - but the user can't know, the libavformat chooses a random FPS? [17:24] <wm4> well, even video formats have error recovery [17:25] <ubitux> yes, and i think you can just force a frame rate in the demuxer [17:25] <wm4> I noticed that the aqtitles demuxer has a subfps option, but microdvd has not [17:25] <wm4> and it'd be nicer if the application could take care of rescaling the times [17:25] <ubitux> does it work if you set the input frame rate? [17:26] <wm4> picking a random timebase is nice too, as long as the application knows it's a frame based format and the timebase is arbitrary [17:26] <ubitux> also, with microdvd you can make it contain the fps in the first even or something like that [17:26] <wm4> how do I set the input frame rate? [17:26] <ubitux> ("{0}{0}25.000 FPS" or sth like that) [17:26] <wm4> oh yeah, like in MicroDVD_capability_tester.sub [17:26] <ubitux> wm4: with the API, dunno [17:26] <wm4> but it seems it's displaying that text for 40 seconds [17:26] <ubitux> how does ffmpeg does it when you -r as input option? [17:27] <ubitux> check if ffmpeg -r 25 -i in.sub out.sub works [17:27] <ubitux> (i never tried) [17:27] <wm4> "{0}{}25.000 FPS" is converted to "Dialogue: 0,0:00:00.00,0:00:40.00,Default,25.000 FPS" [17:28] <ubitux> yes, {} means "up to the next event" [17:28] <ubitux> maybe we could add a special case [17:28] <wm4> also ffmpeg just overwrote my test file [17:28] <wm4> what [17:28] <ubitux> nasty ffmpeg is nasty [17:28] <wm4> ffmpeg -r -i file [17:29] <wm4> oops [17:29] <ubitux> hehe [17:29] <wm4> for microdvd -r doesn't change anything [17:29] <wm4> tried with MicroDVD_capability_tester.sub [17:30] <ubitux> mmh [17:32] <wm4> what's needed is a flag that says that the format is frame based and no FPS was specified in the file header [17:32] <ubitux> there is clearly something to do here [17:32] <wm4> and then it either needs to use frames as timebase, or a random fall back FPS that is known to the application [17:33] <ubitux> yes well [17:33] <ubitux> i need to know how the -r as input can me transmitted to the demuxer, if that's possible [17:34] <wm4> -r is not good [17:34] <wm4> it affects normal video as well right? [17:34] <ubitux> there is no video in a standalone .sub [17:35] <ubitux> or any frame count subtitles format [17:35] <wm4> it would require the application to set -r conditionally based on whether the format is a frame based subtitle, or anything else [17:35] <ubitux> (and that sounds better than the -subfps option...) [17:35] <wm4> not all your users are ffmpeg... [17:35] <ubitux> we can *also* add a flag [17:36] <ubitux> but imo the first step is to make the demuxer access that frame rate information [17:36] <ubitux> and the way -r seems to work looks appropriate [17:40] <wm4> ubitux: btw. what I want to do is setting sub fps to video fps, like MPlayer does [17:40] <ubitux> yes i understood [17:40] <wm4> and doing that after loading the files is most convenient [17:41] <ubitux> after? how could we allow you to do that easily? [17:42] <wm4> basically, it's subtitle_time = frame_number * video_fps / fallback_sub_fps; [17:43] <wm4> which is done if the video fps is known, and fallback_sub_fps is known to be an actual fallback [17:44] <wm4> and where frame_number is the raw subtitle PTS [18:14] <BBB> maybe I'll start writing the initial portions of a vp9 bitstream parser this week, that would be useful [18:15] <BBB> and can't really be done in parallel anyway, so from there we can all take different directions [19:35] <wm4> ubitux: also I have many failures and glitches with a real world .smi sample... should I send it to you? [19:39] <durandal_1707> no, remove it from internet [20:27] <cone-758> ffmpeg.git 03Michael Niedermayer 07master:db27dadcb05a: sonic: replace divide() by ROUNDED_DIV() [20:27] <cone-758> ffmpeg.git 03Michael Niedermayer 07master:69d0a2922f76: sonic: cleanup/simplify num_taps check [20:27] <cone-758> ffmpeg.git 03Michael Niedermayer 07master:9375f5003d02: sonic: use av_freep() as its safer than av_free() [20:31] <durandal_1707> michaelni: sonic actually compress/decompress losslessly? [20:38] <michaelni> durandal_1707, sonicls should [20:43] <durandal_1707> in what container it can be saved? [20:44] <durandal_1707> anyway its doesn't encode losslessly here, is slower than flac and produce 2x times bigger files [20:46] <durandal_1707> also its based on bonk, which is hilarious [21:18] <michaelni> durandal_1707, if its not lossless for you then open a ticket [21:26] <durandal_1707> no point [21:27] <durandal_1707> if you do not plan to rewrite it completely that ignore it and do not waste time on touching it [22:04] <durandal_1707> does anyone have 16/24bit float pcm wavs? [22:59] <nevcairiel> 24 bit float? never heard of such a contraption [23:40] <wm4> hm id3v2 can't be written without seeking? [23:41] <durandal_1707> it could... [23:42] <wm4> I ha the idea that shoutcast metadata could be convert to id3v2 on the fly (though that still sounds like a hack) [23:43] <durandal_1707> indeed [23:53] <BBB> I don't think 24bit floats exist [23:55] <durandal_1707> i can't find half float sample but i know they exist [23:57] <Skyler_> half floats are 16-bit, aren't they? [00:00] --- Mon Jun 24 2013

1 0

ffmpeg.log.20130623
by burek 24 Jun '13

24 Jun '13

[00:22] <vulture> amoxibos: have you checked out coded_frame ? [00:22] <vulture> http://ffmpeg.org/doxygen/trunk/structAVFrame.html [00:23] <vulture> attribute_deprecated short * dct_coeff [00:23] <vulture> says deprecated tho idk [00:23] <vulture> you'd get that from the encoder context ->coded_frame [00:24] <vulture> (not sure if decoder context fills it too but I'd guess it would) [02:32] <xreal> is there a channel for rtmpdump/lib ? [02:33] <durandal_1707> why? native rtmp* support is not working? [02:37] <dagerik> guys. help me extract this video here; http://www.vgtv.no/#!/video/65687/adelen-bombo [02:37] <dagerik> i need to play it using omxplayer [02:37] <dagerik> my raspberry pi cant handle flash [02:51] <klaxa> dagerik: this is not a general "my problem is video related"-help channel [04:47] <apapousek> Hello all, I have question about audio and ffmpeg. [04:48] <apapousek> I'm dumping my BluRay to mkv, then using ffmpeg to change the container to mp4 [04:48] <apapousek> I'm using: ffmpeg -i input.mkv -acodec copy -vcodec copy output.mp4 [04:48] <apapousek> Is it possible to add another audio stream (a stereo downmix) within the same file? [06:41] <silv3r_m00n> what is the difference between the extension .avi, .mpg and the codec and format ? [06:42] <vulture> .avi and .mpg are containers [06:42] <vulture> within them are multiple/different streams of data [06:43] <vulture> for mpg there are different "layers" or codec schemes [06:43] <vulture> mpeg layer 3 for example, is for audio [06:43] <silv3r_m00n> container means ? if the data inside is in different format, then why name all of them as .mpg ? [06:43] <vulture> .mpg is pretty limited in which codecs it can contain, whereas .avi supports any/all codecs [06:43] <silv3r_m00n> it should be something like .codec [06:44] <vulture> because .mpg is a specific container format [06:44] <vulture> and .avi is a different, but specific container format [06:44] <vulture> each codec has their own custom format [06:44] <silv3r_m00n> so what does it mean to have codec A inside .avi ? [06:45] <silv3r_m00n> the video is in codec A format i guess [06:45] <silv3r_m00n> and cannot be played unless the system has the decoder for it [06:46] <vulture> right [06:47] <apapousek> Question: Does the MPEG-4 support Dolby TrueHD audio? [06:47] <apapousek> MPEG-4 container* [06:48] <vulture> no idea [06:48] <apapousek> Alrighty. Any idea is it supports DTS? [06:49] <vulture> as an audiophile I only care about one format, 2 channel stereo :D [06:49] <vulture> dont know anything about what it supports [06:50] <silv3r_m00n> vulture: i see this, http://www.pitivi.org/manual/codecscontainers.html [06:50] <silv3r_m00n> means there are 2 things, container format and codec.... can any format be used with any codec ? [06:50] <apapousek> Alright, thanks. As a pretend audiophile, I care about quality as well. That's why I was wondering about TrueHD [06:50] <silv3r_m00n> or each format works with only specific codecs [06:52] <vulture> pretty much any audio format beyond 2 channel stereo is just flashy selling points. it can be 'useful' if you're trying to emulate some kind of 3D environment in a large room but in the end it gets downsamples to 2 channel stereo in your ears. and I always use headphones anyway :D [06:52] <vulture> so yeah sorry that's why I dont know, because I just havent cared enough to learn about that [06:53] <vulture> silv3r_m00n: containers might only support certain codecs [06:53] <vulture> for example .mpg containers as far as I know only support mpeg-based codecs [06:53] <vulture> but .avi can contain any mpeg codec, as well as a whole lot more [06:56] <sacarasc> silv3r_m00n: A container is a bit like a box. You put stuff in it. [07:53] <knIOO> can someone give me a brief rundown on setsar/setdar and how they are actually used in the command line? I'm getting mismatched sar values when concating two videos together and am attempting to correct it [07:55] <knIOO> I'm getting [Parsed_concat_0 @ 0245e740] Failed to configure output pad on Parsed_concat_0 - full output is here http://pastebin.com/Tzqpinjf [07:56] <ilove11ven> knIOO: [07:57] <knIOO> ilove11ven: hi [14:07] <illusion> how to stream a playlist with ffmpeg ? [14:49] <illusion_> help me please [14:59] <spaam> what? stream a playlist? [15:04] <KuroiTsuki> hello, I compiled latest mplayer/meconder on mingw64 and when I try convert anything with scale (which as far as i understand uses libswscale) mencoder gives me next Assertion firstChrSrcY >= lastInChrBuf - vChrBufSize + 1 failed at libswscale/swscale.c:484. Any ideas what can be causing this? [15:56] <illusion_> spaam, yes to stream a playlist with ffmpeg [17:20] <bencc> burek: now I'm using real ffmpeg and I still can't get live mp3 stream from a rtmp stream [18:11] <bencc> burek: ok. [18:12] <burek> illusion_, im not sure if ffmpeg can handle playlists (yet), but i know vlc can [18:12] <burek> so try with vlc [18:14] <illusion_> can you give me an example to stream a playlist from cvlc to an rtmp server please :) [18:14] <illusion_> burek, [18:15] <burek> cvlc playlist.m3u --sout ... [18:15] <burek> something like that [18:18] <burek> cvlc --sout='#transcode{vcodec=h264,acodec=mp4a,ab=32,samplerate=44100,channels=2}:std{access=rtmp,mux=rtp,dst=1.2.3.4/something}' [18:18] <burek> try reading their wiki, there are some examples how to do that [18:20] <burek> http://www.jpsaman.org/vlc/rtmp [18:26] <bencc> burek: this is the command and output I'm getting http://dpaste.com/1265189/ [18:27] <bencc> I'm trying to get live stream as mp3 to stdout [18:29] <burek> well, everything works ok [18:29] <burek> no errors [18:29] <bencc> burek: I have a live rtmp stream which I want to convert to live mp3 stream [18:29] <klaxa> bencc, you should use pipe: instead of - [18:29] <bencc> and serve it to web clients [18:29] <bencc> testing with pipe: [18:30] <bencc> klaxa: I don't see any output in the terminal with pip:1 (except for log messages) [18:30] <burek> that's also normal [18:31] <burek> what exactly is your issue bencc? [18:31] <bencc> I don't understand how to get a live stream of mp3 packets from ffmpeg [18:31] <burek> did you try ./ffmpeg -loglevel debug -i rtmp://127.0.0.1/audio/test -f mp3 out.mp3 [18:31] <bencc> yes. it works [18:32] <burek> well, what's the problem then? [18:32] <bencc> I can play the output [18:32] <bencc> but I want to stream it live to clients [18:32] <bencc> not create a file [18:32] <burek> do you know what "-" means in your ffmpeg cmd line (or pipe:1) ? [18:32] <klaxa> if i use "ffmpeg -i /home/klaxa/Musik/\[ASL\]\ Various\ Artists\ -\ Ore\ no\ Kanojo\ to\ Osananajimi\ ga\ Shuraba\ Sugiru\ OP\ -\ Girlish\ Lover\ \[FLAC\]/01\ Girlish\ Lover.flac -c:a libmp3lame -b:a 320k pipe:" my terminal gets spammed with mp3 binary [18:32] <klaxa> you must be doing something wrong [18:33] <bencc> I'll try finding a public rtmp stream and show you the result [18:35] <burek> bencc, what exactly do you do with your stream, after you pipe it? [18:36] <bencc> burek: currently nothing [18:36] <bencc> I first want to see it in the terminal [18:37] <bencc> my aim is to pass it to a web server that will serve it to web clients (html5 audio tag) [18:38] <klaxa> to warn you, that will not work with firefox on linux [18:38] <klaxa> firefox on linux doesn't support mp3 in html5 audio tags [18:38] <bencc> for firefox on linux I can use ogg [18:38] <bencc> once I'll manage to make it work with ffmpeg, using other codecs will be easy [18:51] <klaxa> bencc, i took this stream: http://www.twitch.tv/theemero [18:51] <klaxa> got the stream with livestreamer [18:51] <klaxa> and you can listen to it as mp3 now here: http://klaxa.eu/live_streaming.html [18:52] <klaxa> er http://klaxa.eu/live_stream.html [18:53] <bencc> klaxa: is it a live stream or did you just save it to a file? [18:53] <klaxa> it's live [18:53] <bencc> how do you stream it? [18:53] <klaxa> if you open the stream on twich it should be a few seconds ahead [18:54] <klaxa> livestreamer --fifo -o /tmp/stream http://www.twitch.tv/theemero best [18:54] <klaxa> ffmpeg -i /tmp/stream -c:a libmp3lame -b:a 320k pipe: | nc klaxa.eu 8080 [18:55] <klaxa> on klaxa.eu my mp3 server is running and on live_stream.html there is a simple html5 audio tag [18:57] <LordDoskias> is there a tutorial how to use libav to do muxing? the documentation for demuxing was particularly helpful but it is missing the tutorial style presentation for muxing ? [18:57] <bencc> can you see the result on the terminal without sending it to klaxa.eu 8080 ? [18:57] <klaxa> yes [18:57] <klaxa> if i remove that pipe that is [18:59] <bencc> what is the url of this public rtmp stream? [18:59] <bencc> can I test it in my terminal? [19:00] <klaxa> like i said, i'm using livestreamer [19:00] <klaxa> it's some python code that does all the rtmp stuff [19:00] <klaxa> it shouldn't be much different with rtmp directly though [19:01] <bencc> I'm using rtmplite - a python rtmp server [19:01] <klaxa> what software are you using to get your microphone to rtmp? [19:01] <bencc> with a simple swf [19:01] <klaxa> ok [19:01] <klaxa> also, stop reading my mind [19:01] <bencc> I can send you this swf if you like [19:02] <bencc> not getting output in the terminal :( [19:02] <klaxa> would be much appreciated [19:02] <bencc> ok. a second [19:02] <bencc> do you want the mxml source or the compiled swf? [19:02] <klaxa> i don't know, what do i need to reconstruct your setup? [19:03] <bencc> if I'll use 'rtmp://localhost/audio' and later ns.publish('test') will it be fine? [19:04] <bencc> this is the source http://dpaste.com/1265271/ [19:06] <bencc> klaxa: I'm uploading all the files to google drive. a sec [19:06] <klaxa> alright thanks [19:06] <klaxa> also, again, stop reading my mind D: i was about to ask how to use it [19:09] <bencc> https://docs.google.com/file/d/0B12AhxvnYHrAZDlmZ0k5SUI0Nk0/edit?usp=sharing [19:09] <bencc> you can see html file, swf file and the mxml source [19:10] <bencc> I'm using rtmplite with the command python rtmp.py -d [19:10] <bencc> http://code.google.com/p/rtmplite/ [19:10] <bencc> I've tested with a subscriber and I can hear the rtmp stream [19:10] <bencc> test with ffmpeg outputing to a file and it works [19:10] <Yagger> I have an mp4 with 2 audio channels, english and german. I would like to "swap" them so the german audio is "first" and english second. [19:10] <Yagger> Can this be done with ffmpeg? [19:11] <klaxa> bencc: okay i started rtmp.py -d what now? [19:13] <klaxa> Yagger: yes, you can use -map for that: http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20use%20-map%20option [19:14] <jure> hello [19:14] <Yagger> Thank you for the hint, I will try that. [19:14] <klaxa> assuming that your audio streams are at positions 1 and 2 and there are no other streams apart from the video stream: ffmpeg -i my_file.mp4 -map 0:0 -map 0:2 -map 0:1 my_new_file.mp4 [19:14] <jure> is it possible to rip dvd-a to wav with ffmpeg? [19:15] <klaxa> bencc: i have rtmplite and your subscriber directory, what do i do now? [19:15] <klaxa> i started rtmp.py -d [19:16] <bencc> you load the publisher.html file in your browser [19:16] <bencc> allow mic access [19:16] <bencc> and it starts broadcasting [19:16] <klaxa> ah awesome [19:16] <bencc> now you can use this stream from ffmpeg [19:18] <klaxa> i get the stream via rtmp://localhost ? [19:18] <bencc> rtmp://127.0.0.1/audio/test [19:19] <bencc> klaxa: that's the input I'm using for ffmpeg [19:19] <klaxa> mplayer2 returns: HandShake: client signature does not match! [19:20] <bencc> klaxa: that's not good :) [19:20] <bencc> can you try with a different rtmp server? [19:20] <klaxa> ffmpeg fails too [19:21] <klaxa> i don't know of any other rtmp servers [19:21] <bencc> for me the handshake works. weird [19:22] <klaxa> my publisher page is pretty blank too [19:22] <klaxa> is that normal? [19:22] <bencc> do you see a title Publisher? [19:23] <bencc> and below it a text publisher [19:23] <klaxa> http://klaxa.eu/publisher.png [19:23] <bencc> yes. that's good [19:23] <bencc> not very informative, I know [19:23] <klaxa> i didn't have to confirm something about my microphone [19:23] <bencc> weird [19:23] <bencc> let me test [19:24] <klaxa> ah because i already allowed it sometime earlier [19:24] <bencc> ok [19:24] <bencc> you sould see in your browser console [19:24] <bencc> NetConnection.Connect.Success [19:24] <bencc> NetStream.Publish.Start [19:26] <klaxa> where do i find that in chromium? sorry, i'm a webdevelopment stupid [19:26] <klaxa> i barely know how to operate my browser [19:28] <bencc> Ctrl+Shift+J [19:28] <klaxa> nothing there [19:29] <bencc> you need to reload the page to see it [19:29] <klaxa> already did [19:30] <bencc> by the way, chrome have a new mic permission [19:30] <bencc> you need to allow it both in the browser and in the flash app [19:30] <bencc> you should see something in the console [19:30] <bencc> either success or failure [19:31] <klaxa> well the console is empty [19:31] <klaxa> i allowed it in the browser and the flash app [19:31] <klaxa> at least as far as i know [19:32] <bencc> I don't understand why the console is empty [19:33] <bencc> do you use a local web server or did you just dragged the html to the browser? [19:33] <LordDoskias> where can i find information how can I mux audio video in a TS container in memory, not in a file ? so that i can then stream this data over the network? [19:33] <bencc> maybe it's a permission issue [19:34] <klaxa> i opened the file directly, no httpd [19:34] <bencc> let me try it too [19:34] <bencc> maybe that's the issue [19:35] <bencc> can you serve the files with python? [19:35] <bencc> python -m SimpleHTTPServer [19:35] <bencc> than you can see it in http://localhost:8000/publisher.html [19:36] <klaxa> yeah tried that already [19:36] <klaxa> same results [19:37] <bencc> trying [19:38] <Mavrik> LordDoskias, add the right format and url as an output? [19:38] <bencc> klaxa: can you try with FF? [19:38] <LordDoskias> Mavrik, i want to write in a memory buffer [19:38] <LordDoskias> Mavrik, where can i find a tutorial how can i do the muxing? [19:39] <bencc> klaxa: the console doesn't work for me either [19:39] <klaxa> https://gist.github.com/klaxa/5845844 [19:39] <klaxa> stops there [19:40] <bencc> klaxa: right [19:40] <bencc> that's what I'm getting [19:40] <klaxa> why are you using flash inbetween anyways? :X [19:40] <Mavrik> LordDoskias, so you want to use the libav libraries as an API not ffmpeg executable right? [19:40] <LordDoskias> yes [19:40] <xreal> Are there any free RTMP server as online service, where I can stream to using ffmpeg? I don't mean RTMP server software! [19:40] <LordDoskias> Mavrik, yues [19:41] <bencc> klaxa: maybe you are not getting the stream [19:41] <bencc> I'll send you subsriber.html too so you can test if the stream works [19:41] <klaxa> i would think it's the same error as with mplayer2 [19:41] <bencc> klaxa: do you see packets in the rtmplite terminal? [19:42] <klaxa> bencc: https://gist.github.com/klaxa/5845855 [19:42] <klaxa> the complete log during testing [19:42] <Mavrik> LordDoskias, well the best resource you have is doc/examples for muxing and demuxing [19:42] <Mavrik> LordDoskias, and the ffmpeg doxygen [19:43] <bencc> klaxa: ok so publisher.html works. you are sending rtmp audio stream [19:44] <bencc> klaxa: in ffmpeg my output is a bit different than yours [19:45] <bencc> http://dpaste.com/1265350/ [19:45] <klaxa> >playpath=z010002.stream [19:45] <bencc> it manage to do the handshake and play the stream [19:45] <klaxa> where does that come from? [19:45] <bencc> sorry. wrong command [19:45] <bencc> ./ffmpeg -loglevel debug -i rtmp://localhost/audio/test playpath=z010002.stream -f mp3 pipe: [19:46] <bencc> it's probably from testing [19:46] <LordDoskias> Mavrik, i will check the muxing example but i think it deals with files not raw memory, hope it produces muxed packets and then those packets are written into memory [19:46] <Mavrik> um [19:47] <Mavrik> you define your own AVIOContext on the output format [19:47] <Mavrik> and just grab encoded data when it comes to you [19:49] <klaxa> bencc how would i go about to "compile" the mxml file? i uploaded everything to my server, but it fails to connect to localhost (client-side) obviously [19:50] <bencc> klaxa: do you have flex installed? [19:50] <bencc> maybe simpler. there is a demo app in rtmplite [19:50] <bencc> let me see [19:51] <bencc> klaxa: rtmplite/testClient/bin-debug/testClient.swf [19:51] <bencc> and testClient.html [19:51] <bencc> better than my publisher because you can publish and watch the stream for testing [19:56] <klaxa> bencc: http://klaxa.eu/bencc/rtmplite/testClient/bin-debug/testClient.html [19:57] <klaxa> if something should be streaming try accessing it and tell me how... i have no clue how to use that thing [19:57] <bencc> 1. set the URL of your server in NetConnection [19:57] <bencc> 2. click Connect [19:57] <bencc> 3. click Publish [19:59] <klaxa> if i click publish nothing happens [19:59] <bencc> what is the url of your rtmp server? [19:59] <bencc> I'll try to publish form here [19:59] <klaxa> klaxa.eu [19:59] <klaxa> rtmp://klaxa.eu right? [20:00] <bencc> rtmp://klaxa.eu/myapp [20:00] <klaxa> yeah tried that too [20:01] <klaxa> things are happening but it's still not working [20:01] <bencc> trying locally [20:02] <bencc> klaxa: locally it works [20:02] <klaxa> lolweird [20:03] <bencc> maybe a flash permission issue [20:03] <bencc> flash is ugly [20:03] <klaxa> can you get the stream with ffmpeg though? [20:03] <bencc> klaxa: trying [20:03] <bencc> I can publish on my local web server [20:04] <bencc> maybe your rtmplite server is not running or not publically accesible? [20:05] <klaxa> it should have been though... can you forward your rtmp port or something? [20:05] <klaxa> arghs nvm that, but can you grab the stream with ffmpeg? [20:05] <klaxa> can you play it back with ffplay? [20:05] <bencc> trying [20:07] <bencc> klaxa: getting the same output [20:07] <bencc> starts to play the stream but doesn't output packets to the terminal [20:08] <bencc> http://dpaste.com/1265404/ [20:08] <klaxa> can you play it back with ffplay though? [20:08] <klaxa> i mean can you hear it? or anything? [20:10] <bencc> testing [20:11] <bencc> klaxa: I don't have ffplay installed [20:11] <klaxa> try any mediaplayer that supports rtmp then [20:11] <klaxa> mplayer/mplayer2, vlc, anything [20:12] <bencc> I can use testClient in another browser and play the stream [20:12] <bencc> that works [20:12] <klaxa> that is really weir [20:12] <klaxa> *weird [20:13] <bencc> what's weird? [20:13] <bencc> streaming works [20:13] <bencc> playing works [20:13] <bencc> transcoding to mp3 file with ffmpeg works [20:13] <bencc> it just doesn't work in real time [20:13] <bencc> maybe it buffers everything and write to the file when you stop the stream [20:14] <bencc> probably a bug [20:14] <klaxa> stdin should not be buffered [20:14] <klaxa> stdout, yes, but not stdin :X [20:16] <bencc> it should "just work" [20:16] <klaxa> if it would not be too much to ask, i would like to take a look first hand, can you forward an ssh port and add a guest user? (info via query) [21:33] <LordDoskias> so before starting encoding I have to use avformat_alloc_output_context2, however this function implies that the data will be written to a file, how can i do it so that data is written to a memory buffer ? [21:35] <Mavrik> LordDoskias, even though the signature implies filenames, that's not strictly true [21:35] <LordDoskias> i'm reading https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/muxing.c [21:35] <Mavrik> LordDoskias, pass your own AVOutputFormat to that function and AVOutputFormat has callbacks [21:36] <Mavrik> there you have callbacks write_packet, write_header, etc. which will be called by the muxer when it has data ready for you [21:37] <LordDoskias> i see [21:38] <Mavrik> LordDoskias, the question is why don't you just pass something like "udp://192.168.1.1" as an output filename and just let ffmpeg handle that ;) [21:38] <Mavrik> when you said you want to stream over network [21:38] <LordDoskias> because i'm ecndoing with raspberrypi [21:38] <LordDoskias> and i waneed hardware acceleration so i have written custom piece of software that does that [21:39] <LordDoskias> and i get raw encoded data, now i want to feed this data to avlib in order to perform software muxing [21:41] <Mavrik> and what of that stops you from using ffmpeg's streaming code to do the muxing and streaming part for you? [21:44] <LordDoskias> i don't know or rather haven't looked at how i can integrate ffmpeg into this [21:44] <LordDoskias> is there a tutorial ? [21:45] <LordDoskias> how can i hook up ffmpeg to a memory buffer or something? [21:45] <LordDoskias> you are the second person from this channel which hints at this idea [21:45] <LordDoskias> s/which/who [21:45] <Mavrik> just like you started. [21:45] <Mavrik> you have your encoded packets [21:45] <Mavrik> you initialize output context (thatis - the muxer) and as a filename you give it the address of where to stream the output [21:46] <Mavrik> then you just use av_write_packet to it and it'll mux it and send it over network [21:46] <Mavrik> these are the libav supported protocols: http://ffmpeg.org/ffmpeg-protocols.html [21:47] <LordDoskias> is there a raw udp multicast? [21:47] <LordDoskias> Mavrik, but then again this is exactly the same thing i want to do [21:47] <Mavrik> that would be udp://<target-address> :) [21:47] <LordDoskias> oh wait, so you are saying [21:47] <LordDoskias> it will be exatly like in the muxing example [21:47] <LordDoskias> but the file name is going to be the udp://mcast address [21:47] <Mavrik> yup. [21:47] <Mavrik> :)( [21:47] <LordDoskias> yeah, [21:47] <LordDoskias> okay, that's good [21:48] <LordDoskias> saves shitload of work :D [21:48] <Mavrik> that's what I'm telling you :P [21:48] <LordDoskias> but in the demo they are doing: write_video_frame [21:48] <Mavrik> that's a function in the same file. [21:48] <LordDoskias> ah, this is a custom functions [21:48] <LordDoskias> thanks [21:48] <LordDoskias> cool, cool, cool [21:50] <LordDoskias> i don't think ther eis av_write_packet isntead avwrite interleaved frame [21:50] <LordDoskias> which should be ok [21:57] <Mavrik> mhm. [22:56] <bencc> klaxa: not sure how to do it [22:56] <bencc> klaxa: I'm inside a VM behind a router [22:56] <klaxa> oh jeez [22:57] <bencc> anyway, it shouldn't be needed. using rtmplite with the testClient should reproduce the issue [22:57] <klaxa> it still doesn't for me :S [22:57] <bencc> I can help you fix it if you want [22:57] <bencc> I can help you help me :) [22:58] <klaxa> sounds good enough [22:58] <klaxa> do you need shell access to fix it? [22:58] <bencc> I don't think so [22:58] <klaxa> okay [22:58] <bencc> let's start from the beginning [22:58] <bencc> we want to be able to stream with one browser and play it in the other [22:59] <bencc> did you start rtmplite on your local server? [22:59] <klaxa> yes [23:00] <bencc> did you start a local python http server inside the testClient/bin-debug folder? [23:00] <klaxa> yep [23:00] <AnAnt> Hello, I have an .MPG file from a camera, I want to compress it to mp4, I tried the command: avconv -i file.MPG -acodec ac2 file.mp4, but the resulting file had a bad resolution [23:01] <AnAnt> can't I do the compression without losing quality [23:01] <AnAnt> ? [23:01] <klaxa> AnAnt: this channel is fo ffmpeg not for avconv, try #libav [23:01] <bencc> klaxa: now, load http://localhost:8000/testClient.html in your browser [23:01] <AnAnt> ah [23:01] <bencc> klaxa: click on the connect button. you should see NetConnection.Connect.Success in the log [23:01] <klaxa> yep [23:01] <bencc> good. you are able to connect [23:02] <bencc> now click on publish and test the rtmplite terminal [23:02] <bencc> you should see hex messages of packets there [23:02] <klaxa> yes, now i see them [23:02] <bencc> good [23:02] <bencc> you can open testClient with another browser, click connect and than play to test if you can see/hear the stream [23:03] <klaxa> that works too [23:03] <bencc> now you have all that needed to try and create an mp3 stream from the rtmp stream with ffmpeg [23:03] <AnAnt> klaxa: how would I do it using ffmpeg ? [23:03] <llogan> AnAnt: https://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide [23:04] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [23:04] <llogan> all the links you'll ever need. [23:05] <klaxa> bencc: nope won't work [23:05] <bencc> klaxa: what doesn't work? [23:05] <llogan> AnAnt: now you have three choices: simply use a static build of real ffmpeg, compile it, or go to #libav [23:06] <klaxa> can't play it with mplayer2, can't open it with ffmpeg, i'll get a more recent version one sec [23:07] <bencc> klaxa: I wonder if the problem is with rtmplite or with ffmpeg [23:07] <bencc> klaxa: I'll might need to ask on the rtmplite mailing list [23:07] <klaxa> https://gist.github.com/klaxa/5846547 [23:07] <klaxa> i'll build the latest ffmpeg from source, may take some minutes [23:07] <AnAnt> llogan: thanks [23:08] <bencc> klaxa: aren't you missing the stream name? rtmp://localhost/myapp [23:08] <klaxa> hm? [23:08] <bencc> klaxa: I think it should be: rtmp://localhost/myapp/user1 [23:09] <bencc> in testClient you have a textbox which let you enter the stream name [23:09] <bencc> the default is user1 [23:09] <klaxa> looks better, still won't start encoding [23:10] <bencc> you see a message that says it started playing ? [23:10] <klaxa> https://gist.github.com/klaxa/5846555 [23:11] <bencc> klaxa: it should say it is playing something [23:13] <bencc> klaxa: http://dpaste.com/1265775/ [23:13] <klaxa> nope, not happening for me [23:14] <bencc> rtmplite is still running? [23:14] <bencc> are you still publishing audio/video? [23:14] <klaxa> yes [23:15] <bencc> so it should work [23:15] <klaxa> weird shit, we're even using the same binary [23:15] <bencc> no idea why it doesn't [23:15] <klaxa> well not exactly [23:34] <klaxa> haha wow bencc, rmtplite wasn't allowed to bind port 1935 apparently [23:34] <klaxa> which is why things failed [23:36] <klaxa> hmm... but ffmpeg is still not reading the stream [23:36] <klaxa> neither is ffplay [23:37] <klaxa> neither is vlc [23:38] <klaxa> exiting with the same error as mplayer2 [23:38] <klaxa> neither is rtmpdump [23:38] <bencc> klaxa: even on localhost? [23:38] <klaxa> i start to think that rtmplite is at fault [23:39] <bencc> what error do you get? [23:39] <klaxa> https://gist.github.com/klaxa/5846639 [23:40] <bencc> klaxa: are you sure you are publishing? [23:40] <klaxa> yes i'm seeing the stream in the browser [23:41] <bencc> I also got this error with gstreamer. not sure what to say [23:41] <klaxa> with the flash embedd it works [23:42] <bencc> klaxa: I don't know what the problem is [23:42] <bencc> :( [23:43] <klaxa> i would *think* that rtmplite is not 100% compliant to rtmp [23:45] <klaxa> i still don't understand exactly why you need flash :X [23:45] <bencc> klaxa: I can try with red5 [23:45] <klaxa> and why you need rtmp [23:45] <bencc> klaxa: I need users to publish audio stream from the browser and broadcast it as mp3 to browsers without flash [23:46] <klaxa> so you want to make an application to stream audio from browser to browser, one with flash, one without? [23:47] <bencc> yes [23:47] <bencc> one browser to many browsers [23:54] <bencc> klaxa: I'm able to connect with ffmpeg to rtmplite [23:54] <klaxa> so it starts encoding audio? [23:54] <bencc> and sending the play 'test' command [23:54] <bencc> but than it just hang [23:54] <bencc> I've already sent you the log [23:55] <bencc> http://dpaste.com/1265775/ [23:55] <klaxa> yes that's about as far as i get with any player [23:55] <klaxa> also, looking into recording audio with html5, seems possible [00:00] --- Mon Jun 24 2013

1 0

ffmpeg-devel.log.20130622
by burek 23 Jun '13

23 Jun '13

[00:14] <ubitux> ok, tsan report is 131M [00:14] <ubitux> but output completely useless.. [00:14] <ubitux> why is there no symbols.. [00:15] <ubitux> i guess i'll have to --enable-debug [00:15] <ubitux> i thought it was the default [00:15] <ubitux> 'forget disable-stripping [00:16] <ubitux> i'm stupid [00:16] <wm4> I think the common idiom is not stripping by default, and having an install-strip target [00:17] <ubitux> we produce a _g not stripped binary [00:17] <ubitux> so we have both by default, which is nice [00:17] <wm4> and the libraries? [00:17] <ubitux> though, through fate i forgot. [00:17] <ubitux> well, !stripped by default [00:17] <ubitux> afaik [01:23] <ubitux> matthewjheaney_: i'll have a last look to you patch tomorrow, and will likely apply it [01:23] <ubitux> sorry for the delay [01:26] <matthewjheaney_> OK, no hurry. I just wanted to make sure that it didn't fall into /dev/nul. [02:09] <kierank> did webvtt ever get rollup captions [09:38] <ubitux> http://fate.ffmpeg.org/report.cgi?time=20130622012319&slot=x86_64-archlinux… [09:38] <ubitux> a bit more useful output with debug :) [09:39] <ubitux> BBB: btw, did you get a chance to look at the vp8 race? :P [12:07] <cone-319> ffmpeg.git 03Hendrik Leppkes 07master:d76fff7df72d: smvjpeg: use refcounted frames to avoid mem leaks [14:56] <cone-319> ffmpeg.git 03Michael Niedermayer 07master:b6ce50a2d4ba: avutil/ripemd: adjust loop condition to silence CID1035716 [14:59] <Compn> michaelni : are you thinking about merging some of rockboxes' fixed point decoders? :) [19:50] <nevcairiel> \o/ valgrind is green again [19:51] <ubitux> hf with helgrind/drd/tsan now ;) [19:51] <nevcairiel> no thanks [19:51] <ubitux> :( [19:51] <nevcairiel> opening the helgrind report crashes my chrome [19:51] <ubitux> haha [19:51] <nevcairiel> or rather, that tab of the browser [19:52] <ubitux> that's because it tries to put 800M of text in memory maybe [19:52] <ubitux> tsan is only 200M [19:52] <ubitux> so it's ok [19:52] <nevcairiel> anyhow debugging race conditions gives me headaches [19:53] <nevcairiel> especially because many of those are probably false positives [19:55] <ubitux> all the sanity threading tools may be using the same methods then [19:56] <nevcairiel> similar concepts at least i would wager [20:12] <ubitux> michaelni: any objection to the API changes (disposition addition) in "[PATCH] Use private option to specify WebVTT kind"? [20:13] <ubitux> otherwise i'll likely apply soon [20:38] <michaelni> ubitux, no objections but maybe should be a seperate commit [20:38] <ubitux> ok [20:38] <ubitux> matthewjheaney_: care to split the patch? [21:09] <ubitux> saste: are you going to review the delogo patches? [21:39] <Daemon404> wtf [21:39] <Daemon404> ffmpeg fails to build (mingw) for me [21:39] <Daemon404> undefined ref to av_buffersrc_add_ref [21:40] <Daemon404> saste / ubitux -- known? [21:40] <Daemon404> in buffersrc.c i nlavfi [21:43] <ubitux> with lavfi disabled? [21:43] <Daemon404> no [21:43] <Daemon404> straight up plain ./configure [21:43] <ubitux> no idea [21:43] <Daemon404> :| [21:45] <nevcairiel> works for me [21:45] <Daemon404> dafuq [21:46] <nevcairiel> obviously the mingw fate boxes also work [21:46] <Daemon404> oh [21:46] <Daemon404> i radd it wrong [21:46] <Daemon404> udnefined ref to avfilter_copy_buf_props [21:48] <nevcairiel> did you accidentally bump the version? :p [21:48] <Daemon404> no [21:49] <nevcairiel> or build without avcodec? [21:49] <nevcairiel> :) [21:49] <cone-319> ffmpeg.git 03Michael Niedermayer 07master:8689ee0eefb6: sonic: simplify shift_down() [21:49] <cone-319> ffmpeg.git 03Michael Niedermayer 07master:12de2933da6e: sonic: Improve error codes [21:49] <nevcairiel> only reason this function wouldnt exist [21:50] <Daemon404> :| [21:50] <Daemon404> man wtf [22:03] <Daemon404> wut [22:03] <Daemon404> cannot repro on linux with mingw [22:03] <Daemon404> only native [22:03] <Daemon404> dafuq is going on in here [22:03] <nevcairiel> your toolchain is broken [22:03] <Daemon404> its the same toolchain [22:03] <Daemon404> :D [22:04] <nevcairiel> cant run a windows .exe on linux! [22:04] <Daemon404> well you can [22:04] <nevcairiel> try mine if you want, runs fine http://files.1f0.de/mingw/ [22:04] <Daemon404> i think i might have mucked up my ffmpeg git dir [22:04] <Daemon404> in unknown ways [22:04] <Daemon404> $5 says a reclone will fix it [22:05] <nevcairiel> also a possibility [22:06] <ubitux> or just git clean -fdx [22:06] <nevcairiel> i usually prefer that over make distclean, sometimes distclean is a bit silly when files get removed/added/renamed [22:27] <Daemon404> nevcairiel / ubitux -- reclone fixed it [22:27] <Daemon404> herp derp [22:27] <Daemon404> magic. [22:31] <cone-319> ffmpeg.git 03Michael Niedermayer 07master:081a7f3ed065: sonic: replace some float by integers to improve platform independance [22:31] <cone-319> ffmpeg.git 03Michael Niedermayer 07master:37c7a8be19e2: sonic: replace literal numbers by sizeof() [22:51] <cone-319> ffmpeg.git 03Derek Buitenhuis 07master:8bdbabfaa469: configure: Enable MinGW-w64's implementation of vsnprintf and pals [22:51] <cone-319> ffmpeg.git 03Oka Motofumi 07master:81c5ceeda211: avisynth: Make sure the filename passed to Avisynth is in the right code page [22:54] <ubitux> MAX_PATH * 4f [22:54] <ubitux> fear [22:55] <Daemon404> ? [22:55] <ubitux> isn't this gonna be a bit large? [22:55] <Daemon404> it is 260 [22:55] <ubitux> oh [22:55] <Daemon404> always. [22:55] <ubitux> ok haha [22:55] <JEEB> yup [22:55] <ubitux> i guess that's ok then :) [22:55] <Daemon404> i just prefered to use it instead of a magic # [22:56] <ubitux> i was worried because it's something like 4096 on linux [22:56] <Daemon404> ah [22:56] <Daemon404> nah its 260 on windows forever [22:56] <Daemon404> because legacy [22:56] <ubitux> and thus "wchar_t filename_wc[MAX_PATH * 4];" was starting to get a bit large :p [22:56] <ubitux> (especially on the stack) [22:56] <ubitux> yeah sure right ok :) [22:56] <Daemon404> :P [22:56] <ubitux> is 260 enough for absolute path then? [22:56] <Daemon404> finally pushed that mingw fix too [22:57] <Daemon404> [16:56] <@ubitux> is 260 enough for absolute path then? <-- it is windows own path limit [22:57] <Daemon404> so yes [22:57] <Daemon404> for ansi paths anyway [22:57] <Daemon404> which is what avs takes [22:57] <ubitux> dos compat, heh [22:57] <Daemon404> :D [22:58] <ubitux> let's add a freedos fate instance now [22:58] <Daemon404> hey at least we dont need to use 8.1 filenames [22:58] <JEEB> 8.3 you mean? [22:58] <Daemon404> whatever it was [22:58] <JEEB> EIGHTFAG.COM [22:59] <JEEB> 8.3 [23:30] <BBB> ubitux: no... let me try now, it was test-vector-010 right? [23:30] <ubitux> not only [23:31] <ubitux> fate-vp8-sign-bias-emu-edge [23:31] <ubitux> and fate-vp8-test-vector-010 yes [23:31] <ubitux> and fate-vp8-test-vector-emu-edge-010 [23:40] <BBB> let me try to build current git and see how hard it is to reproduce [23:40] <BBB> 2 threads right? [23:40] <BBB> do you know how to debug this? [23:40] <BBB> it's a useful skill to learn [23:41] <matthewjheaney_> Which files go in which patch? [23:43] <BBB> hey matt, working on a weekend? [23:43] <matthewjheaney_> Yup [23:44] <ubitux> BBB: yes, 2 threads [23:44] <ubitux> BBB: and no, i have absolutely no idea [23:46] <BBB> hm, ok, so you start by running the test indefinitely until it fails [23:47] <BBB> you try to get an idea of how often it fails [23:47] <BBB> for me it fails after 500 runs [23:47] <ubitux> that part is ok, i did it for the h264 race(s) [23:47] <BBB> n=0; while true; do out=`./ffmpeg -nostats -cpuflags all -threads 2 -thread_type frame -i ~/Movies/fate-suite-ff/vp8-test-vectors-r1/vp80-00-comprehensive-010.ivf -f md5 -nostats -v 0 - | cut -d= -f2`; let n=n+1; echo "$n: $out"; if test $out != 61d05919a9883d9f215eb3f2db63eb13; then break; fi; done [23:47] <BBB> 495: 61d05919a9883d9f215eb3f2db63eb13 [23:47] <BBB> 496: 61d05919a9883d9f215eb3f2db63eb13 [23:47] <BBB> 497: 1ef41b204e71b74f5ff7ea3b65dc4a6a [23:47] <BBB> ok, then do that a couple of times to be sure it's always the same race [23:48] <BBB> you don't want to study a variety of races [23:48] <BBB> e.g. now I get a different one [23:48] <BBB> 73: 61d05919a9883d9f215eb3f2db63eb13 [23:48] <BBB> 74: 22dd9d372aa487d2ab9978b0854058ea [23:48] <BBB> (that could be uninitialized memory, but it's easier when you keep studying the same race, so I tend to focus on just one) [23:49] <BBB> once you have one that is reproducible, extend the test and write the output yuv file to disk in addition to printing the md5 [23:49] <BBB> if all races have the same initial yuv file mismatch, it's fine [23:50] <BBB> then look at the file of the yuv mismatch and see what goes wrong, in what macroblock [23:50] <BBB> use printf debugging [23:50] <BBB> and always do it in the loop where you break only if the mismatch occurred [23:50] <BBB> so the line above is serves as a control [23:52] <BBB> or just parse a good one to ref.yuv and the bad one to mismatch.yuv [23:52] <BBB> a [23:52] <BBB> nd [23:52] <BBB> [23:52] <BBB> com [23:52] <BBB> pa [23:52] <BBB> re [23:52] <BBB> oops, sorry, stupid chat client [23:53] <ubitux> ok [23:54] <ubitux> the main issue is that i haven't much clue how threading in ffmpeg actually works, but well yeah ok [23:54] <BBB> printf debugging requires no prior knowledge [23:54] <ubitux> sure :) [00:00] --- Sun Jun 23 2013

1 0

ffmpeg.log.20130622
by burek 23 Jun '13

23 Jun '13

[00:59] <norbert_> hi all, I'm transcoding a 25fps .mod to 60fps .mp4 [01:00] <norbert_> this works fine but I need to deinterlace the video and when I add "-deinterlace" for some reason the video speeds up [01:00] <norbert_> here is the command I'm using: ./ffmpeg -r 25 -deinterlace -i in.mod -b 4000k -ab 160k -r 60 -acodec libvo_aacenc out.mp4 [01:01] <norbert_> I've also tried it with "-deinterlace" before "out.mp4" but it still speeds up the video [01:01] <norbert_> without "-deinterlace" it works fine, but I need to deinterlace the video, because I want to get rid of the lines [01:01] <norbert_> any ideas what I could try; how to fix this? [01:03] <norbert_> ok, I thought the input would be enough [01:03] <norbert_> I'll also paste the console output somewhere [01:06] <norbert_> http://pastebin.com/9ynStRUg [01:15] <mootsadog> norbert_, i'm not experienced but what i've been doing today might sort of help... i've been using yadif to deinterlace, and succeeded in creating frame-doubled deinterlaced video at the proper speed [01:17] <mootsadog> i use ffmpeg -i input.mov -filter:v yadif=1 -r [FRAMERATE] output.mov -- where [framerate] hopefully it'd conform? [01:18] <mootsadog> you know what, actually i dunno if that helps at all. but using yadif instead of -deinterlace might yield different results. [01:18] <llogan> you shouldn't need to declare -r with yadif [01:18] <llogan> and doing so may dro por duplicate frames [01:18] <llogan> *drop or [01:23] <norbert_> mootsadog: ok, I'll try: ./ffmpeg -r 25 -i in.mod -filter:v yadif=1 -b 4000k -ab 160k -r 60 -acodec libvo_aacenc out.mp4 [01:24] <norbert_> I'll check if it worked tomorrow; mp4 doesn't play until the whole thing is finished and it's several gigs [01:24] <mootsadog> llogan is right because the yadif=1 makes one frame per field [01:24] <mootsadog> i was derping out [01:25] <norbert_> my input video is 25 fps and I want the output to be 60 fps [01:25] <norbert_> I shouldn't specify -r 25 when I use -filter:v yadif=1 ? [01:29] <llogan> norbert_: you can use the -t option if you don't want to wait to encode the whole thing [01:32] <llogan> why do you want 60? [01:33] <norbert_> the rest of the footage is 60 fps and the kdenlive project is also 60 fps [01:34] <norbert_> do I need to drop the -r 25, you think? [01:34] <llogan> are you sure it's not 60000/1001? [01:34] <norbert_> well, it's 59.something [01:34] <norbert_> or whatever [01:34] <norbert_> what I'm doing now always works fine, I just need to deinterlace this particular video [01:38] <llogan> ffmpeg -i input -vf yadif,fps=60000/1001 -codec:a copy output.mkv [01:38] <norbert_> I need 60 fps [01:38] <llogan> i don't know if the audio will be in sync or not [01:39] <llogan> then change the fps value to 60. now i'm just giving you want you think you need. [01:40] <norbert_> I think I need to specify -r 25 for the input [01:40] <norbert_> or ffmpeg doesn't understand I want to go from 25 to something else [01:40] <norbert_> also, I will stick with -b 4000k -ab 160k [01:41] <norbert_> and I have good experiences with -acodec libvo_aacenc and kdenlive [01:41] <norbert_> :) [01:41] <llogan> unfortunately libvo_aacenc is probably the worst AAC encoder that ffmpeg can use [01:44] <sacarasc> What's the best, llogan? [01:46] <mootsadog> is anyone familiar with tfields? [01:47] <llogan> sacarasc: https://ffmpeg.org/trac/ffmpeg/wiki/AACEncodingGuide [01:48] <sacarasc> Thanks. [01:51] <pyBlob> llogan: I finally catched a nice cloud-free night to use my ffmpeg setup on something more interesting than street-lamps: [01:51] <pyBlob> http://www.pasteall.org/pic/53941 http://www.pasteall.org/pic/53944 [01:53] <pyBlob> the moon is still a bit distorted, as my program doesn't fine-tune the image-positions and it doesn't correct for air-turbulence yet [01:57] <llogan> pyBlob: looks like progress to me. thanks for sharing. [01:58] <pyBlob> thanks ^^ [02:08] <bencc> I want to stream rtmp as mp3 [02:08] <bencc> ffmpeg -loglevel debug -i rtmp://127.0.0.1/audio/test -f mp3 test.mp3 [02:09] <bencc> how can I make client access test.mp3? [02:09] <bencc> or maybe I need to call a different command? [02:20] <klaxa> bencc: what you are doing now is saving the stream as mp3 [02:21] <bencc> klaxa: how can I broadcast the live stream? [02:21] <bencc> as a progressive mp3 so I can load it in html5 audio clients [02:21] <klaxa> i don't understand exactly [02:22] <klaxa> you have a file called test.mp3 and you want to stream it to html5 live? [02:22] <bencc> I have a flash client that broadcasts mic as rtmp stream to a media server [02:22] <klaxa> and you want to restream that as mp3 to html5? [02:23] <bencc> now I want to broadcast this audio rtmp stream as mp3 to html5 clients [02:23] <bencc> yes [02:24] <klaxa> hmm.. interesting usecase [02:24] <klaxa> can you save the file for now though? [02:24] <klaxa> i mean you are saving it, right? [02:24] <bencc> I need it in real time [02:25] <bencc> you are right. my command is creating an mp3 file. when I stop the rtmp server I see the file [02:25] <bencc> is it possible to stream the mp3 in real time? [02:26] <klaxa> hmm... well i don't want to advertise exactly, but i wrote an mp3 streaming server quite a while back [02:26] <klaxa> i don't know if that will work for sure though, it will enable http streaming, i don't know in how far the html5 audio-tag is compatible with that [02:26] <bencc> how can I pass the stream from ffmpeg to a streaming server? [02:27] <klaxa> ffmpeg -i rtmp://whatever/address/ -f mp3 pipe: > ./streaming_server [02:27] <bencc> this is html5 audio palyer playing live mp3 stream http://jsbin.com/ibawut/2 [02:28] <bencc> what is "pipe: > ./streaming_server" ? [02:28] <bencc> ./streaming_server is a file descriptor? [02:28] <klaxa> well ./streaming_server takes a stream via stdin [02:28] <klaxa> it *would* be a streaming server that takes a stream via stdin [02:29] <klaxa> pipe: > redirects the output to stdout of ffmpeg [02:29] <bencc> what if I'm calling ffmpeg from a port driver? [02:29] <bencc> can I just use "pipe: > " ? [02:29] <klaxa> what is a port driver exactly? [02:29] <bencc> it's a way in erlang to call a program [02:31] <klaxa> i'm not sure, if it is interpreted by a shell, then probably yes [02:32] <bencc> so if I have a server that gets mp3 packets from ffmpeg, I just broadcast them to clients? [02:33] <bencc> or do I need to send mp3 header first or similar? [02:34] <klaxa> mp3 is headerless [02:34] <klaxa> well more or less [02:35] <klaxa> each mp3 frame has its own header [02:35] <bencc> so I just broadcast the packets to all clients [02:35] <klaxa> not exactly [02:35] <klaxa> you need to start broadcasting at the beginning of a frame [02:35] <klaxa> https://github.com/klaxa/mp3server [02:36] <bencc> how do I know that I'm at a beginning of a frame? [02:36] <klaxa> if you want to take a look, i think it's horrible code, if you have questions about it, feel free to ask, i should still know about most of the stuff i wrote [02:36] <klaxa> you can determine the length of a frame by reading its header [02:36] <klaxa> the next frame starts at the end of the current frame [02:36] <klaxa> so if you get the first frame, you can calculate where the next frames starts and so on [02:36] <bencc> but if my server just gets stream of bytes, how do I know when a frame starts? [02:36] <bencc> ok [02:37] <bencc> so I have to parse the frame header [02:37] <klaxa> pretty much [02:37] <bencc> I need to find docs showing the mp3 frame header [02:37] <bencc> erlang is very good for handeling protocols [02:37] <bencc> thanks for your server. I'll have a look [02:38] <klaxa> http://mpgedit.org/mpgedit/mpeg_format/mpeghdr.htm [02:38] <klaxa> i used this spec to implement the header parsing [02:38] <bencc> mp3 is the same as mpeg? [02:39] <klaxa> from wikipedia: >MPEG-1 or MPEG-2 Audio Layer III [02:40] <bencc> thanks [02:40] <bencc> I have some work now :) [02:40] <klaxa> :) [02:43] <maldous> Q. Is there an HEVC/H265 patch available, that allows building of a cygwin library (chenm03's code on google only includes the library, not the source) ? [02:54] <blue_misfit> Hello! How might I go about using ffmpeg to take a live HLS stream (from an m3u8 playlist), and remux it into an mpeg-ts being streamed over udp? [02:56] <blue_misfit> I've done something simple like ffmpeg -i http://foo/bar.m3u8 -vcodec copy -acodec copy -f mpegts udp://127.0.0.1:1234 [02:56] <blue_misfit> but when I try to open udp://127.0.0.1234 in VLC, nothing happens [03:17] <klaxa> bencc: i've been playing a bit with html5's audio tag: http://klaxa.eu/live_stream.html [03:17] <klaxa> so at least it's possible to stream mp3 with the audio tag, however, for some reason this doesn't work in firefox [03:19] <sacarasc> Because Firefox doesn't support MP3. [03:20] <klaxa> ah... well... [03:22] <klaxa> ah it does support mp3 on windows 7 and higher apparently [03:22] <klaxa> and windows vista [03:23] <sacarasc> Ah, since Firefox 21, apparently. [03:24] <klaxa> but my phone can play it in chrome, fancy [04:52] <seba-> hm [04:52] <seba-> are these builds http://ffmpeg.gusari.org/static/ [04:52] <seba-> without alsa? [05:03] <braincracker> how do you remove watermark with ffmpeg ? [05:03] <braincracker> logo, or whatever that is stationary [05:16] <klaxa> braincracker: http://libav.org/libavfilter.html#delogo [05:16] <klaxa> it's in the libavfilter part of ffmpeg [06:15] <Gary13579> Does anyone know how to screencast with ffmpeg without audio/video desyncing? I'm trying to stream my desktop to twitch.tv, but the a/v desyncing makes it impossible. [06:15] <Gary13579> I've read the problem may be related to pulseaudio, but I need to use pulse for other various reasons. Is there any way around it? [06:16] <Gary13579> Here's the command I'm using to stream https://gist.github.com/Gary13579/5835860 [06:18] <vulture> warlock neerrrrrrrrrd :p [06:19] <Gary13579> haha [06:19] <Gary13579> it has been a while, but yes :P [08:05] <elkng> "quantum mechanics is just a theory not a fact", is that true ? [08:12] <sacarasc> elkng: ##physics [11:36] <MontyMoose> Hello, I have ffmpeg working in Ubuntu, but I need to add the libtheora codec. Can someone let me know how to recompile ffmpeg with this new codec without breaking it. I've read howtos and just confused myself& anyone? [12:25] <burek> <MontyMoose> Hello, I have ffmpeg working in Ubuntu [12:25] <burek> how exactly did you get it [12:25] <MontyMoose> I think I just did a simple apt-get thingy [12:25] <burek> then there is no easy way to do what you want [12:25] <MontyMoose> it was a while ago, it's been working fine, but I now need to add a new codec [12:25] <MontyMoose> I thought that might be the case :-( [12:26] <burek> that's a package, already bundled with some libraries [12:26] <burek> you can't just "add" libtheora to it [12:26] <durandal_1707> you need to get source and compile yourself [12:26] <burek> although this would be a nice feature of ffmpeg [12:26] <MontyMoose> Yup thought that might be the case too... [12:26] <MontyMoose> I'm hopeless at compiling things, every time something seems to go horribly wrong [12:27] <burek> we provided nice tutorials on ffmpeg compilation at our wiki [12:27] <MontyMoose> is there not a package out there somewhere with every goddam codec in the world all nicely packaged up and ready to go! [12:27] <burek> so you might check it out [12:27] <burek> or you can try static builds [12:27] <burek> it's like an exe file for windows [12:27] <MontyMoose> ooh now that sounds better [12:28] <burek> and no, you won't be able to get "a package with every goddam codec" due to copyright licenses and such legal s..tuff :) [12:28] <MontyMoose> that looks great, but how do I reliably remove all the rubbish already installed without? [12:28] <burek> no need to [12:28] <MontyMoose> so I can just leave ffmpeg installed? [12:28] <burek> download a static binary and run it with ./ffmpeg [12:28] <burek> yes [12:29] <burek> it contains all the libraries bundled in that static binary [12:29] <MontyMoose> ok - I'm on it& you may have to hold my hand though, cos despite being good and reading howtos - I'm still fairly thick with general linux tinkering... [12:31] <MontyMoose> ok so I've got that package dowacky& and I've stuck it on my server - what do I do next? unzip it I suppose? [12:32] <burek> yes [12:32] <burek> and after that just type ./ffmpeg [12:33] <MontyMoose> tar jxf ffmpeg-linux64-20130614.tar.bz2 yup [12:34] <MontyMoose> ok so ./ffmpeg just says - -bash: ./ffmpeg: No such file or directory [12:36] <MontyMoose> ok was in the wrong directory (oops) it's run now - and stuff has appeared saying things& any other steps? [12:42] <burek> well use your ffmpeg now [12:42] <burek> that's it [12:42] <burek> just refer to it either with ./ffmpeg (if you are in the same dir) [12:42] <burek> or with the full path to it [12:50] <MontyMoose> I see [12:50] <MontyMoose> thanks - I'll have a go [12:50] <MontyMoose> can I move the directory to a better place - or should I leave it where it is? [13:11] <xreal> librtmp really seems to be buggy :( [13:11] <xreal> curl and ffmpeg have problems, while rtmpdump works fine. [14:11] <seba-> how do you force drawtext to show date/time [14:12] <seba-> %F %T %Z doesn't work anymore [14:21] <durandal_1707> seba-: what version? [14:24] <seba-> durandal_1707 i get parsed_drawtext stray % near T [14:24] <seba-> :/ [14:24] <seba-> N-54141-g1a405c6 [14:28] <seba-> actually sorry if i use \%F \%T \%Z [14:28] <seba-> it says near F not near T :) [14:29] <durandal_1707> did you read documentation? [14:29] <durandal_1707> and what version that FTZ thing actually works? [14:30] <seba-> durandal_1707, i don't remember which, but it worked [14:30] <seba-> i think the previous GIT version in february or something [14:30] <seba-> late january [14:30] <seba-> oops [14:31] <saste> seba-, behavior changed since then [14:31] <seba-> saste, yes i can see that, so what should i do? [14:31] <saste> you need to specify the expansion mode [14:31] <seba-> yes i was looking in that [14:31] <seba-> but i didn't understand really wtf they mean by that [14:31] <saste> you can specify the old behavior, but is not default anymore [14:31] <seba-> yes [14:31] <seba-> i understood that [14:32] <seba-> but i didn't know how to do that [14:32] <seba-> how to specify this [14:33] <seba-> saste, do i do this at compile time? [14:33] <saste> seba-, expansion=strftime [14:34] <seba-> saste, oh just that? [14:34] <seba-> lol [14:34] <saste> or you read TFM about text expansion in drawtext [14:34] <seba-> lololol [14:34] <seba-> yes i did [14:34] <seba-> but you have like an example [14:34] <seba-> drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'" [14:34] <seba-> or drawtext='fontfile=FreeSans.ttf:text=%{localtime:%a %b %d %Y}' [14:35] <seba-> how would i know what they mean by that [14:37] <seba-> ok it works saste [14:37] <seba-> thank you [15:00] <xlinkz0> what was the thread option? [15:01] <seba-> saste, if you can change the docu, just add at the end expansion=strftime in the example [15:01] <seba-> it's not clear [15:02] <saste> seba-, strftime is deprecated in favor of the new expansion mechanism [15:02] <saste> that is you should use %{localtime=} or gmtime [15:02] <seba-> oh [15:02] <seba-> i see [15:03] <seba-> ah now i understand what they mean by that [15:46] <xlinkz0> the fps reported on the command line at the end is total average? [15:48] <durandal_1707> afaik yes [15:51] <xlinkz0> ty [16:49] <silv3r_m00n> i am trying to capture desktop video using avconv, like this, avconv -f x11grab -s xga -r 30 -b 2000k -g 300 -i :0 ~/session-recording.avi but the quality is very poor, how to improve quality of video ? [16:49] <JEEB> you didn't set any format so the default can be whatever :D [16:50] <JEEB> also you're setting bit rate before -i [16:50] <silv3r_m00n> JEEB so what shud be the correct command [16:51] <JEEB> try something like this: avconv -f x11grab -s xga -r 30 -i :0 -c:v ffvhuff ~/session-recording.avi [16:52] <silv3r_m00n> totem requires, video/x-avi-unknown decoder [16:52] <silv3r_m00n> how to get that now [16:52] <JEEB> lol totem [16:53] <JEEB> it sure fails [16:53] <JEEB> try mplayer or VLC [16:53] <JEEB> also do note that avconv is a libav binary, not ffmpeg binary (project-wise) [16:53] <silv3r_m00n> let me check [16:53] <JEEB> so the "correct" place to seek support is #libav [16:54] <silv3r_m00n> the quality is good now, but a 300mb file for just 5 seconds video ? [16:54] <JEEB> yes, lossless compression. You can try -c:v libx264 -preset ultrafast -crf 0 instead of -c:v ffvhuff [16:55] <JEEB> and then make the preset slower to make the compression better [16:55] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset [16:55] <JEEB> list of x264's presets [16:56] <silv3r_m00n> cant i do -c:v mpeg4 ? [16:58] <JEEB> that doesn't have a lossless mode [16:58] <JEEB> and if you want lossy you might as well use x264 [16:58] <JEEB> and raise the crf value [16:58] <JEEB> :P [16:58] <JEEB> but that really doesn't make sense [16:58] <JEEB> capture first in lossless [16:58] <JEEB> then re-encode with proper settings to lossy [16:58] <JEEB> for distribution [16:59] <JEEB> it only makes sense to use something fast and if possible lossless for the capture step [17:01] <silv3r_m00n> libx264 is more sensible, ffvhuff is strange, it will take up entire hard disk in 5 minutes [17:01] <JEEB> it's not strange, just a simple huffman-based format [17:02] <JEEB> libx264 with lossless mode is somewhat better, but of course preset ultrafast isn't that good [17:02] <silv3r_m00n> instead of -s xga can i specify the full screen resolution of 1380x768 ? [17:02] <dagerik> silv3r_m00n: use -s 1380x768 [17:03] <dagerik> or -vf scale=-1:768 [17:04] <silv3r_m00n> in -i :0 can i specify a specific window id to record only that window ? [17:09] <bencc> klaxa: how do you stream the mp3? [18:25] <bencc> how can I get stream of mp3 packets? [18:25] <bencc> I've tried: [18:25] <bencc> ffmpeg -i rtmp://127.0.0.1/audio/test -f mp3 - [18:25] <bencc> and: pipe:1 [18:25] <bencc> but I don't see any output [18:38] <macph> Hello ! I don't know whether I am here on the right place - but is someone here who could help with rtmpdump and rtlnow.de ? rtmpdump is build from source on Mac OS X and work fine on other sites. [19:50] <Angela1> hola [20:04] <burek> macph, you should contact rtmpdump supporters, not ffmpeg [20:05] <burek> bencc, does this work: ffplay -i rtmp://127.0.0.1/audio/test [20:07] <bencc> burek: what does it do? [20:08] <burek> it plays the given url? [20:08] <bencc> trying [20:10] <bencc> burek: I don't here it but I it's inside a VM so I'm not sure I'm supposed to here the audio [20:10] <burek> ok, type: ffmpeg -i rtmp://127.0.0.1/audio/test -f mp3 - [20:10] <bencc> ok [20:12] <bencc> burek: http://dpaste.com/1261994/ [20:12] <burek> *** THIS PROGRAM IS DEPRECATED *** [20:12] <burek> this is not ffmpeg [20:13] <bencc> burek: I 've installed ffmpeg so I thought I'm using ffmpeg :) [20:14] <burek> no, your distribution maintainer tricked you into thinking you were [20:14] <burek> you were actually using something named avconv [20:14] <burek> which is just a fork of genuine ffmpeg [20:15] <bencc> ok [20:15] <bencc> how can I use ffmpeg? [20:15] <burek> well, either use static binaries [20:15] <burek> or compile your ffmpeg [20:17] <bencc> thanks [20:18] <burek> :beer: :) [21:31] <bencc1> what is the framesize of speex in actionscript? [21:31] <bencc1> can ffmpeg tell me that? [21:37] <bencc1> ok, found it: "If you use the Speex codec, the sample rate is set to 16 kHz." [22:02] <LordDoskias> if i want to use avformat library to demux an input multicast stream can i do this directly by passing the stream's address to av_open_input_file or do i have to come up with a solution that receives packets from the networks and somehow passes them to avformat in order ? [22:33] <t4nk838> Hi, I'm trying to send per frame metadata with ffmpeg...Has anyone got a clue how this is feasible? [23:01] <amoxibos> Hello, I have a task of implementing a video transcoding (frame skipping, mostly in the frequency domain, so no decoding->encoding step). The videos are h264, I and P frames only. I have started using ffmpeg library, and I am a bit stuck at getting the macroblock DCT coefficients from the raw AVPacket. Can ffmpeg parse the AV packet in more detail, or do I have to interpret the raw packet data manually? Any other suggestions? [00:00] --- Sun Jun 23 2013

1 0

ffmpeg-devel.log.20130621
by burek 22 Jun '13

22 Jun '13

[00:07] <michaelni> durandal_1707, i gave multiple ffmpeg-devs coverity accounts last year [00:09] <michaelni> its possible that by default everyone who has an account gets mails now [01:07] <cone-963> ffmpeg.git 03Michael Niedermayer 07master:32fc8d6db65e: avcodec/jpeg2000dec: check that tp_end is after the start [01:07] <cone-963> ffmpeg.git 03Michael Niedermayer 07master:c17dd513e7f1: bytestream2_init: assert that buf_size is valid [01:25] <cone-963> ffmpeg.git 03Michael Niedermayer 07release/0.10:8ddc9790edc1: avformat/libmodplug: Reduce the probe score for small input [01:25] <cone-963> ffmpeg.git 03Michael Niedermayer 07release/0.11:a1e4d1f933c9: avformat/libmodplug: Reduce the probe score for small input [01:25] <cone-963> ffmpeg.git 03Michael Niedermayer 07release/0.9:340c1843c59c: avformat/libmodplug: Reduce the probe score for small input [01:25] <cone-963> ffmpeg.git 03Michael Niedermayer 07release/1.0:8d18dbda1e45: avformat/libmodplug: Reduce the probe score for small input [01:25] <cone-963> ffmpeg.git 03Michael Niedermayer 07release/1.1:2cfdf732efaf: avformat/libmodplug: Reduce the probe score for small input [01:25] <cone-963> ffmpeg.git 03Michael Niedermayer 07release/1.2:e4bb67bc5069: avformat/libmodplug: Reduce the probe score for small input [01:25] <cone-963> ffmpeg.git 03Michael Niedermayer 07release/1.2:073bde2b1fbb: avformat/iff: Byte seek is unsupported [02:08] <durandal_1707> the thing that shows configuration parameters is cut in middle, like fixed size array was used [03:18] <Daemon404> im retiring my linux fate instances i think [03:18] <Daemon404> since clang 3.3 is out with ioc built in [03:18] <Daemon404> adding icl instances to ffmpeg as we speak [03:39] <cone-963> ffmpeg.git 03Michael Niedermayer 07master:79cd5d39ba11: avcodec/utvideodec: Fix vlc len [03:40] <Daemon404> http://fate.ffmpeg.org/report.cgi?time=20130621013942&slot=i686-windows-icl [03:41] <Daemon404> sure is lavfi failures [03:50] <cone-963> ffmpeg.git 03Timothy Gu 07master:8cdea50f6eee: doc/decoders: Document libilbc decoder [04:04] <BBB> Daemon404: I think ubitux wanted to help as well [04:04] <BBB> maybe saste [04:04] <BBB> we'll see how many end up helping [04:04] <BBB> but yes aug should work, it'll be fun :) [07:53] <ubitux> BBB: yes i made some time space for exactly one month full time in august :) [08:04] <nevcairiel> interesting that ICL has the same fate rgb failure as MSVC. well, plus some more [08:04] <nevcairiel> I should really find some time to figure out wtf is up [08:06] <ubitux> [Parsed_amovie_0 @ 00A9AC20] Failed to avformat_open_input 'C' [lavfi @ 00AB9080] Error initializing filter 'amovie' with args 'C:/Msys/1.0/devel/fate/samles_ffmpeg/amrwb/seed-12k65.awb' [08:06] <ubitux> classic shit ;) [08:41] <ubitux> anyone tested -fsanitize=thread with clang? [08:44] <ubitux> same question with "undefined" and "integer" [09:15] <nevcairiel> ubitux: for these errors yes, its just the stupid parsing problem, but the rgb errors do not have that, so i'm going to assume its similar to the msvc problem [09:16] <nevcairiel> especially because they produce the same frame hashes [09:16] <ubitux> interesting :) [09:16] <nevcairiel> if i remember correctly, we saw some chroma shift in these, right? [09:16] <nevcairiel> maybe a rounding thing somewhere [09:16] <nevcairiel> well, i'll try to check, maybe on the weekend [09:17] <ubitux> yes it was something along those lines [09:17] <ubitux> i'm wondering if that's not related to the pad code [09:37] <ubitux> 09:33:31 <+av500> http://people.xiph.org/~xiphmont/demo/daala/demo1.shtml [10:50] <cone-11> ffmpeg.git 03Anton Khirnov 07master:eeeb5c291d3f: vsrc_movie: do not free avoption variables in uninit() [10:50] <cone-11> ffmpeg.git 03Michael Niedermayer 07master:5d509fbdcf62: Merge commit 'eeeb5c291d3f78eaade5b99c2614c7cab0e9be79' [11:18] <cone-11> ffmpeg.git 03Anton Khirnov 07master:720a1de52fb4: lavc: free the padded last frame during audio encoding properly [11:18] <cone-11> ffmpeg.git 03Michael Niedermayer 07master:98abe16522be: Merge remote-tracking branch 'qatar/master' [12:36] <cone-11> ffmpeg.git 03Michael Niedermayer 07master:1ee8fadb811f: avdevice/x11grab: allocate just one Cursor [12:36] <cone-11> ffmpeg.git 03Timothy Gu 07master:b43860ee0c27: doc/decoders: Document libopencore-amrnb decoder [12:56] <cone-11> ffmpeg.git 03Hendrik Leppkes 07master:659df32a9d89: mathops/x86: work around inline asm miscompilation with GCC 4.8.1 [15:24] <BBB> nevcairiel: I have some very old local todo that says that the c code rgb to yuv or yuv to rgb conversion has some byte inversions compared to the simd [15:25] <BBB> nevcairiel: since that (unscaled) code is inline mmx, it would mean that it is only triggered on systems missing any optimizations, or missing inline asm [15:25] <BBB> nevcairiel: I don't think we have a fate station that does --cpu-mask 0 [15:28] <JEEB> BBB, reminds me that it was a difference in the YCbCr->RGB conversion code that broke my Ut Video encoder tests last year :D Since mru told me to remove the 'bitexact' flag ("Don't put in flags that you don't know"), and I was basically matching to the output of the asm'ified code path. [15:31] <nevcairiel> BBB: well easy enough to test locally [15:32] <nevcairiel> i'll do that this weekend iguess [15:55] <nevcairiel> BBB: -cpuflags 0 still passes the test here [16:06] <nevcairiel> even funnier, if i run MSVC with cpuflags=0, it passes [16:06] <nevcairiel> o.O [16:13] <nevcairiel> so one of the very few yasm swscale functions isnt playing nice, i guess [16:18] <Chikuzen> old patch http://git.videolan.org/?p=ffmpeg.git;a=blobdiff;f=libavformat/avisynth.c;h… [16:18] <Chikuzen> new patch https://lists.ffmpeg.org/pipermail/ffmpeg-devel/2013-June/144927.html [16:18] <Chikuzen> wtf [16:19] <nevcairiel> wasnt the avisynth demuxer rewritten in between [16:28] <JEEB> yes, qyotwhatever dude added avxsynth support [16:28] <JEEB> and thus removed that thing [16:32] <BBB> nevcairiel: then why would it pass on linux etc.? [16:32] <BBB> nevcairiel: sounds more to me like an issue with stack alignment disabling certain optimizations or so [16:33] <BBB> nevcairiel: e.g. there being a mmx and sse2, sse2 requiring aligned stack, thus being disabled on msvc [16:33] <BBB> nevcairiel: can you try -cpuflags mmx etc. and up to see when it fails? [16:34] <nevcairiel> i did, mmx gives wrong result, sse2 crashes [16:34] <BBB> uh [16:34] <BBB> ok [16:35] <nevcairiel> not sure why "all" doesnt crash [16:35] <BBB> missing bitexact flag? [16:35] <nevcairiel> its set [16:35] <BBB> I believe ffmpeg and libav have different meanings for sse2 [16:35] <BBB> if you want libav's meaning of sse2, you need to use mmx+mmx2+sse+sse2 or so [16:36] <BBB> it was something horrific like that last time I looked [16:36] <BBB> (I had the same issue a while ago) [16:36] <nevcairiel> funny enough mmx+sse2 doesnt crash [16:36] <nevcairiel> i'm confused [16:36] <BBB> :) [16:37] <BBB> ok but so mmx fails [16:37] <BBB> I bet it's a missing stack alignment directive [16:37] <BBB> or so [16:38] <BBB> you can file a bug and assign it to me, maybe I'll have time to look [16:51] <shurick> is there any bad consequences from increasing MAX_SLICES in libavcodec/h264.h from 16 to 512? [17:02] <nevcairiel> memory requirements most likely [17:02] <nevcairiel> why change it? [17:04] <shurick> i have to work with heavily sliced h264 streams [17:05] <shurick> i'd like this change to be persistent, if it would not cause any harm to other users [17:05] <Skyler_> 16 is rather low IMO; it should be at least ~256 [17:05] <Skyler_> technically, most sliced streams work fine with it set to *1* [17:06] <Skyler_> but it'll spam warnings, which is actually the main problem [17:06] <Skyler_> the other problem is it's used in the calling code for one of the hardware decoders (dxva2?) so a many slice stream will actually crash there instead of just warn [17:25] <cone-11> ffmpeg.git 03Michael Niedermayer 07release/1.1:d8e76a531cd7: avdevice/x11grab: allocate just one Cursor [17:25] <cone-11> ffmpeg.git 03Michael Niedermayer 07release/1.2:fbb1af39e464: avdevice/x11grab: allocate just one Cursor [17:48] <cone-11> ffmpeg.git 03Andrey Utkin 07master:3857aea6d960: doc/protocols: document "srtp" protocol [17:48] <cone-11> ffmpeg.git 03Stefano Sabatini 07master:3f8750776f38: doc/protocols: apply very minor consistency fixes [18:05] <nevcairiel> ahah i fixed it [18:06] <nevcairiel> stupid swscale [18:11] <nevcairiel> whole swscale gets messed up if your cpu flags contain sse2 but not mmx, luckily this is not a problem in real world, but really can make you get odd errors when using -cpuflags <.< [18:28] <nevcairiel> \o/ msvc fate passes again with the patch [18:28] <cone-11> ffmpeg.git 03Timothy Gu 07master:83647ace735d: doc/decoders: Add libopencore-amrwb decoder doc [18:28] <cone-11> ffmpeg.git 03Stefano Sabatini 07master:7cc4115dd6d1: doc/decoders,decoders: add various missing final dots [20:49] <cone-11> ffmpeg.git 03Michael Niedermayer 07master:2005fddcbb4e: h264/ff_h264_check_intra_pred_mode: fix input value check [20:49] <cone-11> ffmpeg.git 03Michael Niedermayer 07master:d6a33f5d20b6: h264: fix size of arrays in ff_h264_check_intra_pred_mode() [21:17] <cone-11> ffmpeg.git 03Hendrik Leppkes 07master:7cdf574c22b8: swscale: fix filter alignment reduction without inline asm [21:54] <wingrime> michaelni: I hope you still remeber me, I need ask some help... [21:56] <ubitux> anyone interested in a tsan fate instance? [22:04] <nevcairiel> http://fate.ffmpeg.org/report.cgi?time=20130621200342&slot=x86_32-msvc10-dl… [22:05] <nevcairiel> yay, its green again [22:05] <nevcairiel> and it wasnt ubitux's fault afterall [22:05] <ubitux> :) [22:05] <ubitux> i guess moving to filter complex make a different filter insert [22:05] <nevcairiel> i still wonder why it started failing the day you changed the fate tests [22:05] <Daemon404> orite [22:05] <Daemon404> ill enable more icl stuff today [22:05] <Daemon404> i.e. 64 [22:06] <nevcairiel> you need to fix the pathes you use [22:06] <Daemon404> me? [22:06] Action: ubitux trying to add a tsan instance [22:06] <nevcairiel> well unless you want the tests to fail until someone is annoyed enough to write a new option parser [22:06] <Daemon404> eh [22:06] <Daemon404> what magic do i need [22:06] <Daemon404> this sounds annoying already [22:06] <nevcairiel> use relative path [22:07] <nevcairiel> the : in windows absolute confuses it [22:07] <Daemon404> for what [22:07] <Daemon404> there are like 6 paths to change [22:07] <nevcairiel> fate samples url [22:07] <Daemon404> o [22:07] <Daemon404> well that just feels dirty [22:07] <Daemon404> :( [22:07] <Daemon404> also relative to WHAT [22:07] <Daemon404> there are liek 4 possible places [22:07] <nevcairiel> to the target path [22:08] <nevcairiel> where the binary runs [22:08] <Daemon404> thats still too vague [22:08] <Daemon404> that coul be the traget path, .../install, .../install/bin [22:08] <Daemon404> etc [22:09] <Daemon404> whats teh CWD when fate runs [22:09] <nevcairiel> i have it relative to what is set in target_path [22:09] <Daemon404> ok [22:09] <Daemon404> what a crapshoot. [22:09] <nevcairiel> the problem is a combination of silly factors [22:10] <nevcairiel> mostly because a msvc exe is being run from a msys shell [22:10] <nevcairiel> it doesnt combine too well [22:10] <nevcairiel> but also because the damn option parser uses : as a separator [22:11] <Daemon404> wouldnt that fail being run from cmd.exe too [22:11] <nevcairiel> eh, i forget, its been a while since i looked into it properly [22:12] <Daemon404> if so [22:12] <Daemon404> im pretty sure windows has a function that checks if some arbitrar string is a valid path [22:12] <Daemon404> which could be useful [22:13] <nevcairiel> i think it would probably fail with cmd.exe too [22:13] <nevcairiel> curious that noone complained on trac yet (or i didnt see it), but guess most people still use mingw builds [22:14] <Daemon404> because nobody uses lavfi [22:14] Action: Daemon404 runs [22:14] <nevcairiel> yeah its really only a problem if you use this lavfi input device instead of using a "normal" input and then piping it through a filter graph [22:15] <nevcairiel> i can see how many people wouldnt use this special magic [22:15] <Daemon404> http://msdn.microsoft.com/en-us/library/windows/desktop/aa364963(v=vs.85).a… [22:15] <Daemon404> call this when in doubt [22:15] <Daemon404> to see if it fails [22:15] <Daemon404> ? [22:16] <nevcairiel> i'm sure the lavfi maintainers will enjoy MS specific code in their neat little files =) [22:16] <Daemon404> because purity is more important than working? ;) [22:17] <nevcairiel> that depends who you ask, i guess [22:17] Action: Daemon404 points nevcairiel at haskell [22:17] <nevcairiel> i dont know enough about the lavfi input parsing stuff to even figure out where to put that [22:17] <ubitux> lavfi "maintainers" mostly just can't test with windows related stuff [22:17] <nevcairiel> and frankly, i only use the libraries =) [22:17] <Daemon404> ubitux, well thats why fate exists! [22:18] <ubitux> that doesn't help to write windows code [22:18] <ubitux> when you're running *nix systems only [22:18] <Daemon404> the option parsing is reproducible in wine [22:18] <Daemon404> or should be. [22:18] <Compn> ubitux : help me in #mplayerdev :P [22:18] <nevcairiel> i can setup a win7 VM with visual studio for anyone that wants to work on it :D [22:18] <ubitux> Daemon404: ENOCARE then [22:18] <Daemon404> ? [22:18] <Daemon404> i dont follow the logic [22:18] <nevcairiel> even run it on my server for you to RDP into! [22:19] <Daemon404> i was thinking [22:19] <Daemon404> after that libav email about msvc [22:19] <Daemon404> i see enough of those emails [22:19] <ubitux> Compn :/ [22:19] <Daemon404> i might do a small sreencast [22:19] <Daemon404> "this is how you build it" [22:19] <nevcairiel> its really not that complicated [22:19] <Daemon404> nevcairiel, ... youd be surprised [22:19] <Daemon404> people work in MSVS 24/7 for years [22:19] <Daemon404> msys is scary and confusing [22:19] <Daemon404> etc [22:20] <nevcairiel> well i suppose you need a small scavenger hunt to gather the tools required [22:20] <Daemon404> i thought i documented it pretty well in platforms.texi [22:20] <Daemon404> apparently not well enough. [22:20] <nevcairiel> while i know where to find a finished msys package without much fuzz, people may not [22:20] <Daemon404> eh? [22:20] <Compn> isnt it better to crosscompile mingw using cygwin ? [22:20] <Daemon404> mingw-get installer [22:20] <Daemon404> is one click [22:20] <Compn> because then you get cygwin bash and mingw exe files... [22:20] <nevcairiel> thats "fuzz" for me, i rather get a zip and unpack, and there we go [22:21] <nevcairiel> Compn: cygwin is horrible [22:21] <Daemon404> cygwin runtime is not horrible [22:21] <Daemon404> its standard env is though [22:22] <nevcairiel> everytime i use cygwin i afterwards have to fix my filesystem permissions [22:22] <Daemon404> but yeah i get at least one email a month [22:22] <Daemon404> for help with msvc [22:22] <Daemon404> i dont know how much of that is stupidity or not reading the docs or what [22:23] <nevcairiel> i suppose if you also want zlib, and then zlib not dynamically linked (because carrying around zlib-1.dll or what its called is silly), it might end up scaring some people off [22:24] <Compn> most builds are static [22:24] <nevcairiel> the pre-built zlib for windows comes with the silly dll iirc [22:24] <Daemon404> Compn, static is not feasible for many windows things [22:24] <Daemon404> i.e.e not FOSS [22:24] <Daemon404> because LGPL. [22:25] <nevcairiel> can still static-link zlib into av* and then dynamic link those to your app [22:25] <nevcairiel> no violations there [22:25] <Daemon404> nevcairiel, and it builds with -MD b default [22:25] <Daemon404> zlib, i mean. [22:25] <nevcairiel> not really relevant if you static link it [22:26] <Daemon404> well yeah it is [22:26] <Daemon404> libav/ffmpeg are MT [22:26] <nevcairiel> i havent had to change anything in zlib config at least when i built it a few weeks ago [22:26] <Daemon404> mixing crt = n fun [22:26] <Daemon404> no* [22:26] <Plorkyeran> building ffmpeg/libav with MD is no longer difficult [22:26] <nevcairiel> but static libraries dont carry the crt anyway [22:27] <Daemon404> Plorkyeran, no but MT i still default [22:27] <nevcairiel> they rely on the stuff you link to to provide it [22:27] <michaelni> wingrime, what kind of help ? [22:27] <michaelni> ubitux, more fate instances are always wanted [22:27] <Daemon404> nevcairiel, that hasnt stopped msvc from crapping out botloads of problems with MT and MD generated code [22:27] <Plorkyeran> that doesn't help with MD/MT stuff [22:27] <Plorkyeran> the static library will still be looking for the wrong symbols [22:27] <ubitux> michaelni: a first report should arrive "soon" [22:27] <Daemon404> yea [22:27] <ubitux> michaelni: hopefully in less time than helgrind or drd [22:28] <ubitux> (and maybe less than a 800M report..) [22:28] <wingrime> michaelni: with understandment witch hw decoder choose for implemnt [22:28] <Plorkyeran> which runtime library to use isn't determined entirely by how you link the final executable/dll [22:28] <nevcairiel> hm, maybe i just forgot that i changed it [22:29] <nevcairiel> who knows [22:29] <nevcairiel> links fine anyway [22:29] <wingrime> michaelni: I mean hw decoder API [22:29] <Compn> michaelni : what hw decoder api is the current one? there is vda and ... something ? [22:30] <michaelni> wingrime, you know your hw, so you should be able to choose how to implement it better than i (who doesnt know at all) [22:31] <nevcairiel> doesnt ffmpeg already have all relevant hw APIs? [22:31] <nevcairiel> i suppose some stuff for ARM devices isnt present [22:31] Action: nevcairiel shrugs [22:32] <wingrime> blob use self-made API [22:33] <wingrime> michaelni: question was, witch API implement , vdpau, or .... [22:34] <wingrime> I simply don't know witch will work with ARM and hw overlays [23:00] <burek> this looks like an mpegts issue or so http://ffmpeg.gusari.org/viewtopic.php?f=16&t=957 [23:37] <michaelni> wingrime, thats really your dicission and i know even less which will work with "ARM and hw overlays", if someone else knows something iam sure he will reply though [23:39] <cone-11> ffmpeg.git 03Hendrik Leppkes 07master:1a405c683e03: smvjpeg: remove redundant frame init code [00:00] --- Sat Jun 22 2013

1 0

ffmpeg.log.20130621
by burek 22 Jun '13

22 Jun '13

[00:04] <dagerik> sacarasc: 15min [00:10] <dagerik> -preset ultrafast increased the encoding speed [06:09] <axorb> hey guys, I'm trying to generate HLS segments on the fly but I'm having trouble getting them to play back nicely with each other [06:11] <axorb> the segmenter is supposed to run though an entire file and generate all parts at once, but I'm trying to add seek support so the user can generate a part that hasn't been visited yet [06:12] <axorb> basically, ffmpeg -i input.mov -ss 30 -t 10 output-3.ts [06:13] <axorb> I want to know if there is something fundamentally flawed with this concept, or if I have to keep trying different settings until VLC/JWPlayer can play my stream properly [06:13] <axorb> At part boundries JWPlayer stops video (but audio continues), while VLC skips a few seconds of content [06:17] <axorb> it may seem like a weird approach but I cannot access the file until the user visits the page (so can't pre-encode), and I'd like to add seek support [06:24] <axorb> NEVERMIND, FINALLY FIGURED IT OUT [06:24] <axorb> #EXT-X-DISCONTINUITY [06:25] <elkng> fix your shift key [06:26] <axorb> sorry, I spent two days on this [07:04] <t4nk315> hey [07:06] <t4nk315> avformat_alloc_output_context2(&oc, NULL, "hls", filename); now how to change the ffmpeg parameters like time wrap list_size [07:15] <vulture> maybe look for functions that can take an options dictionary [07:34] <t4nk315> avformat_alloc_output_context2(&oc, NULL, "hls", filename); now how to change the ffmpeg parameters like time wrap list_size ? [07:34] <t4nk315> using it to mux audio and video streams . [08:04] <mootsadog> Hello all [08:05] <mootsadog> I hope I'm not wasting anyone's time with a dumb question, but it's 1am and I have a throbbing headache, so I'm kinda crossing my eyes at the documentation [08:06] <mootsadog> I use OS X and need to do a yadif deinterlace on a 50i MPEG2 stream file to make a 50p result. [08:08] <mootsadog> I'm having trouble figuring out how to tell ffmpeg to filter to an output file. Absolutely nothing else about the video needs to change, just deinterlace 50i -> 50p. [08:35] <mootsadog> oh yeah, i can tell this is going to do an awesome job. i'm missing something, though, because the output appears to be heavily compressed [08:36] <mootsadog> Unfortunately the aforementioned headache has overcome me and i have to call it a night. I'll be back around in the morning [09:11] <vulture> mootsadog: ffmpeg's default is to use very poor quality... you gotta up the bitrate generally to a sane amount [09:12] <vulture> dunno why the default is set to "bad" [09:14] <bencc1> I have live RTMP stream which I want to convert to mp3 and broadcast to several clients [09:15] <bencc1> do I need to call ffmpeg for each connected client or can I use the same mp3 output stream for several clients? [09:16] <vulture> once [09:18] <bencc1> vulture: ffmpeg is not a TCP server, right? [09:18] <bencc1> so I need to pass the mp3 stream to a media server? [09:28] <vulture> sure [09:28] <vulture> idk [09:31] <bencc1> idk? [09:32] <bencc1> what will be the output? constant stream of packets of mp3 to stdout? [09:36] <vulture> it has a large variety, whatever you tell it [09:37] <vulture> I dont know your options..... [09:37] <t4nk315> avformat_alloc_output_context2(&oc, NULL, "hls", filename); now how to change the ffmpeg parameters like time wrap list_size [09:37] <vulture> t4nk315: [00.15.17] <vulture> maybe look for functions that can take an options dictionary [09:38] <vulture> if you cant find it through one of the context structures [09:39] <t4nk315> cant find it [09:40] <vulture> how do you normally specify those parameters [09:42] <t4nk315> i know how to specify those in ffmpeg command in terminal [09:43] <t4nk315> need to know where to specify it in the Context [09:44] <vulture> how do you specify them in ffmpeg [09:45] <vulture> afaik you pass complex options like that through a dictionary [09:45] <vulture> just search through the context structures for "dict" and surely you'll find structures or functions that take them [09:45] <t4nk315> k [09:47] <t4nk315> i coud only find a pointer to option [09:47] <t4nk315> inside the avclass [09:53] <xlinkz0> does anyone have experience with receiving a stream from rtmpd ( crtmpserver ) ? [10:16] <praveenmarkandu> is it better to launch multiple ffmpeg processes or launch just one with multiple file outputs? [10:24] <t4nk734> hi [10:30] <Mavrik> praveenmarkandu, one with multiple outputs will make sure you only decode the file once :) [11:21] <t4nk315> hey [11:21] <Mavrik> g'day. [11:22] <t4nk315> I changed the dbl value in the hls_time option in the hls class but no effect [13:06] <StaRetji1> howdy folks, a quick question if someone can help [13:07] <StaRetji1> I installed a lot of ffmpeg x264 on ubuntu servers for encoding [13:07] <StaRetji1> but yesterday I tried again and after installation, I am getting "Unknown encoder 'libx264' even though x264 is compiled, ffmpeg compile with x264 [13:08] <StaRetji1> is there a known bug in git and what could I be doing wrong if I follow guide to the letter, as I always did [13:09] <JEEB> you are most probably just running the wrong binary [13:09] <JEEB> `which ffmpeg` [13:11] <StaRetji1> /usr/local/bin/ffmpeg [13:12] <JEEB> that looks like a location for a custom build, now it starts sounding like you didn't get ffmpeg compiled with libx264 after all :D [13:13] <JEEB> `ffmpeg -codecs:v | grep "x264"` [13:14] <JEEB> ..or you didn't install the ffmpeg binary with libx264 linked in :) [13:17] <StaRetji1> thank you JEEB http://pastebin.com/gVjWXhvZ [13:17] <StaRetji1> I was following https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [13:17] <StaRetji1> to the work [13:17] <JEEB> so yeah, that ffmpeg has no x264 linked in [13:18] <JEEB> also I think that guide now compiles the binary to your home [13:18] <JEEB> check if you have an ffmpeg binary in ~/bin [13:23] <StaRetji1> I checked, I have ffplay only [13:23] <StaRetji1> on ~/bin [13:24] <StaRetji1> so, I will try to remove all and try again, even though I tried 2 times already :/ [13:26] <JEEB> when you get to configuring ffmpeg itself, just make sure you check the configure script's output :P [13:26] <JEEB> it should tell you whether or not you have libx264 found and enabled [14:01] <determinant> hello. [14:03] <determinant> i want to decode a flac file to wav format and output to stdout from which neroAacEnc can read directly. [14:03] <determinant> but i don't know how. can anybody help me? [14:10] <JEEB> ffmpeg -i welp.flac -f wav - [14:10] <JEEB> something like this [14:10] <JEEB> I would guess [14:11] <durandal_1707> use pipe protocol [14:19] <Mavrik> o yuch [14:19] <Mavrik> ffmpeg stagefright support is totally broken [14:20] <StaRetji1> Thx JEEB, I am doing it all over from the begining [14:20] <StaRetji1> I hope will be okay this time, will watch every bit :) [14:20] <StaRetji1> btw, should I compile x264 with make or make fprofiled [14:21] <StaRetji1> will fprofiled give better performance, as I run several ffmpeg threads at the same time [14:22] <JEEB> you need raw video for fprofiled, and the performance boost you possibly get from fprofiling is very small [14:22] <JEEB> so just compile and install [14:22] <StaRetji1> roger that [14:23] <StaRetji1> thanks once again Jeeb, I am very grateful [14:25] <LordDoskias> hello, i'm oppening an mpeg2 stream and avformat lib prints this : mpeg_decode_postinit() [14:25] <LordDoskias> several times, but in the end i do receive the expected output [14:25] <LordDoskias> is this something i should be worried about? [14:27] <durandal_1707> Mavrik: what is broken? [14:28] <Mavrik> it doesn't even compile for starters and afterwards it expects to be linked to 2.x stagefright lib and after that it's buggy :) [14:45] <esperegu> anyone has a working example ffserver config for me to create a rtsp stream from a ffmpeg audio stream? [15:01] <StaRetji1> omg JEEB, clean install, no error, ffmpeg was showing libx264 in enabled encoders, but still the same Unknown encoder 'libx264' [15:01] <StaRetji1> it was step by step followed in the guide [15:01] <JEEB> calm down [15:01] <JEEB> pastebin config.log [15:01] <StaRetji1> lol [15:01] <JEEB> and make sure you have tried to run the compiled binary [15:02] <StaRetji1> config.log, let me look for it [15:02] <JEEB> as in, you're in the directory where you ran configure and make, and then run dot-slash-ffmpeg [15:02] <JEEB> ./ffmpeg [15:02] <JEEB> and try looking at its -codecs:v list for x264 [15:02] <StaRetji1> ffmpeg 2>&1 | head -n1 [15:02] <StaRetji1> ok [15:03] <JEEB> just make sure you ran the binary you just configured and built [15:09] <StaRetji1> yes, you are absolutely right [15:09] <StaRetji1> on new machine, clean, there is ffmpeg in /root/bin/ [15:10] <StaRetji1> and it works :/ [15:12] <JEEB> that isn't what I asked [15:12] <JEEB> also why the hell are people using ffmpeg as root :s [15:12] <JEEB> christ [15:14] <StaRetji1> hate typing sudo lol [15:16] <JEEB> you should only need root when installing the binary, if and only if you want to install it under a spot that is unavailable to normal users [15:16] <JEEB> you should not need it when compiling ffmpeg (or its dependencies), nor when using [15:17] <JEEB> and yes, I usually have a screen window open that has root rights, but I most definitely don't run random crap as root [15:17] <JEEB> but enough of that, have fun and I don't even know what you've done or doing [15:18] <StaRetji1> thanks man, whole time I as running wrong ffmpeg (which I was not aware it is installed with apt-get) [15:19] <StaRetji1> that's why I got unknown libx264 [15:34] <determinant> JEEB: thanks. [15:57] <xreal> Is this okay to get the flv and transport the stream to mp4? http://fixee.org/paste/o5vjcyt [16:40] Last message repeated 1 time(s). [17:24] <norbert_> hi, question: I'm trying to transcode/re-encode a 25 fps .mod to a 60 fps .mp4; this works fine if I don't use -deinterlace but I need to get rid of the interlaced scan-lines; when I use the following (that is, add -deinterlace), the video speeds up (I don't understand why): ./ffmpeg -r 25 -i in.mod -b 4000k -ab 160k -r 60 -acodec libvo_aacenc -deinterlace out.mp4 [17:24] <norbert_> any suggestions? [17:24] <norbert_> ffmpeg version N-54023-g6fe419b [17:25] <norbert_> crazy version number :) [17:41] <norbert_> wait, I probably need to use it on the input stream [17:43] <norbert_> will try that; if unsuccessful I'll ask again to see if someone's around; bye [20:02] <tatshwork> i'm wondering if my encode messed up because i'm getting a strange error with MP4Box [20:02] <vulture> blame sm5! [20:03] <tatshwork> what are you doing here :P [20:03] <tatshwork> when i use MP4Box, i get this error [20:03] <tatshwork> [avc-h264] invalid nal_size (106456)? Skipping 106452 bytes to reach next start code\n[avc-h264] error: no start code found (2147650100 bytes read out of 7264192457) - leaving [20:03] <tatshwork> and the mp4 comes out to be 2.6 GiB instead of 6.8 GiB + audio [20:05] <llogan> why are you using MP4Box? [20:06] <vulture> tatshwork: I wanted to try to resolve a ffmpeg api problem but apparently it's a design flaw of the library =/ [20:06] <tatshwork> llogan, what do you suggest? [20:07] <tatshwork> the format of the x264 file is rawvideo (-f rawvideo) [20:50] <bencc> I'm trying to connect to a local rtmp stream with [20:50] <bencc> ffmpeg -loglevel debug -i rtmp://127.0.0.1/audio/test -f test.mp3 [20:50] <bencc> the server sends the first handshake response but ffmpeg gives me [20:50] <bencc> Cannot read RTMP handshake response [21:20] <knIOO> Quick question, I'm converting uncompressed AVIs down to mkv using x264, currently the command I'm using is basically file.avi -acodec copy -vcodec libx264 -crf 23 out.mkv -- is there any way you guys would recommend improving this for rendering performance? [21:20] <knIOO> there are so many settings to play with [21:22] <sacarasc> For encoding speed or playing speed? [21:23] <knIOO> encoding speed - the video is getting uploaded to a video sharing website and being retranscoded again :( [21:24] <llogan> knIOO: note that, depending on your input, your output may not be yuv420 and therefore retarded players won't like it. [21:24] <knIOO> llogan: it's just going to Youtube basically [21:24] <llogan> i'd change -crf 23 to -crf 18 [21:25] <llogan> and if you want to encode faster then use a faster preset [21:25] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide [21:25] <knIOO> so really crf and preset are the only two settings I need to play with? [21:25] <knIOO> as in none of the other stuff like trellis or anything like that will significantly impact encoding performance/file size [21:26] <llogan> the presets deal with that [21:26] <knIOO> Makes sense, thanks! [21:26] <llogan> YouTube counts as a retarded player. [21:27] <llogan> so show your console output to verify that your output is yuv420p, or simply add "-pix_fmt yuv420p" or "-vf format=yuv420p" to be safe [21:27] <knIOO> I don't think it is funny you should mention that, I seem to remember FRAPS being dumb about that [21:28] <knIOO> I'll add that to the script though just to be sure, thanks for that! [21:28] <llogan> i haven't checked YouTube in a while, so maybe it is better lately [21:30] <xreal> Is this okay to get the flv and transport the stream to mp4? http://fixee.org/paste/o5vjcyt [21:55] Action: moot_ returms, wondering if anybody's feeling froggy to help figure out the double-deinterlacing issue he mentioned last night [21:56] <vulture> was the issue just the speed being better? or what [22:04] <mootsadog> vulture: i want to go from 50i -> 50p [22:04] <mootsadog> using yadif with defaults, i get 50i -> 25p [22:05] <mootsadog> i tried setting r: 50 but it appears to either double frames or just remain at 25p [22:05] <vulture> is there a deinterlacing algorithm you can set to double lines ? [22:06] <mootsadog> that's what i'm wondering [22:09] <llogan> you're parobably using yadif=1 which creates one frame for each field [22:09] <mootsadog> in the end, what i'm trying to do is go from 50i source material to 24p results [22:10] <mootsadog> hm, i suppose maybe there's a yadif=2 then? [22:10] <mootsadog> i'm totally new to ffmpeg [22:11] <llogan> wait, i've confused myself. using yadif=1 should double the frame rate... i missed your 50i -> 25p [22:11] <mootsadog> using yadif defaults gives me a 25p output [22:11] <llogan> see the docs: http://ffmpeg.org/ffmpeg-filters.html#yadif-1 [22:12] <mootsadog> thanks, i hadn't been able to find that previously [22:12] <mootsadog> sounds like yadif=1 is my winner [22:14] <xreal> Is this okay to get the flv and transport the stream to mp4? http://fixee.org/paste/o5vjcyt [22:31] <mootsadog> llogan: unfortunately the yadif=2 appears to be producing frame-doubled output [22:41] <mootsadog> can i use yadif 2x with ffmpeg? [22:46] <xreal> Is this okay to get the flv and transport the stream to mp4? http://fixee.org/paste/o5vjcyt [22:49] <burek> xreal, when remuxing, you usually want to keep all your streams intact, so I would advise you to use: ffmpeg -i input.flv -map 0 -c copy output.mp4 [22:49] <burek> btw, you might check http://www.ffmpeg.org/ffmpeg-all.html#rtmp [22:49] <xreal> burek: but my commandline does the same, normally? [22:49] <xreal> burek: I think of a generic use :) [22:50] <burek> and probably use "rtmp_live" too [22:50] <burek> xreal, you never know until you try and see :) [22:51] <xreal> burek: rtmp_live means: 1h download time? :) [22:54] <burek> i dunno, never used rtmp :) [22:57] <xreal> burek: I'm pretty sad, ffmpeg loses the metadata from rtmp [22:59] <burek> what do you mean it loses? did you tell ffmpeg to keep it? :) [23:00] <xreal> burek: how can I ? [23:02] <burek> try using "-c copy -map 0" instead of "-c:a copy -c:v copy" [23:06] <xreal> ok [23:06] <xreal> burek: But I think, librtmp is wrong somehow. rtmpdump works perfectly, ffmpeg only with -re [23:07] <burek> try using google to see some of ffmpeg rtmp examples, to get a clue how to properly use it [23:14] <xreal> burek: I did. It download fine, but not faster than realtime. It breaks after 10 seconds. [23:20] <burek> xreal, if you are viewing a live stream [23:20] <burek> it's already real-time [23:20] <burek> you can't read it faster than it arrives [23:20] <burek> so using -re is pointless [23:20] <burek> it only makes sense to use with file inputs [23:20] <xreal> it's no livestream. [23:22] <burek> http://ffmpeg.org/ffmpeg.html [23:22] <burek> ctrl+f '-re ' [23:30] <dichotoPangea> Hey people, I've been having some trouble recording my screen with ffmpeg, anyone tell me what all this noise means? http://pastebin.com/7s7kMADA [23:30] <vulture> maybe you need to run as root? idk [23:31] <dichotoPangea> I've run ffmpeg for recording without root before, but I'll try. O.o [23:32] <dichotoPangea> Nope, same output. [23:35] <dichotoPangea> Does anyone know an actual command that'll just record my desktop? I don't even need audio. [23:35] <vulture> I dont use linux sorry :D [23:36] <sacarasc> dichotoPangea: It means that you're not using ffmpeg, but avconv which is part of Libav. [23:37] <sacarasc> ^ Read that. [23:37] <dichotoPangea> flv format would be nice also. [23:39] <dichotoPangea> Or somewhere I could learn such a command? [23:41] <llogan> dichotoPangea: we only support ffmpeg here. [23:41] <dichotoPangea> I'm /trying/ to use ffmpeg. [23:42] <llogan> by ffmpeg, i don't mean the fake version from libav [23:42] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [23:43] <llogan> or ask in #libav [00:00] --- Sat Jun 22 2013

1 0

ffmpeg-devel.log.20130620
by burek 21 Jun '13

21 Jun '13

[01:08] <Daemon404> thats a lot of attachments. [03:08] <cone-902> ffmpeg.git 03Timothy Gu 07master:c16496c3770f: doc/decoders: Document libgsm decoder [03:08] <cone-902> ffmpeg.git 03Andrey Utkin 07master:f9739a34ad53: Document "crypto" protocol [09:52] <saste> uhm so, no more socis mentors? [11:24] <cone-963> ffmpeg.git 03Alex Smith 07master:e0d6e2fc67b8: configure: Remove unneeded icl inline compatibility definition [11:24] <cone-963> ffmpeg.git 03Michael Niedermayer 07master:99dc90ea59f6: Merge remote-tracking branch 'qatar/master' [15:24] <cone-963> ffmpeg.git 03Andrey Utkin 07master:2d70f11ef5d4: Document "sctp" protocol [15:24] <cone-963> ffmpeg.git 03Michael Niedermayer 07master:c605adbf562f: avformat/libmodplug: Reduce the probe score for small input [15:26] <durandal_1707> ok, i think encoder is now ready for testing [15:38] <BBB> Daemon404: spec is in the works [16:47] <durandal_1707> michaelni: fine to rename files? to make merges "easier"? [17:04] <michaelni> durandal_1707, sure [17:05] <durandal_1707> michaelni: even if that makes merges harder? [17:06] <michaelni> if someone volunteers to do the extra work [17:07] <durandal_1707> i can't volunter 24/7/12/356 [17:07] <wm4> durandal_1707: lazy! [17:10] <durandal_1707> wm4: no, i have real life [19:47] <durandal_1707> michaelni: so what's best scaler quality wise? [19:52] <michaelni> i dont really know, bicubic & lanczos are popular. [20:13] <cbsrobot> durandal_1707: I read once that sinc is best for scaling down, mitchell for scaling up [20:13] <ubitux> durandal_1707: that's a large encoder you ahve there :) [20:21] <durandal_1707> ubitux: hmm. perhaps i should add more macros [20:21] <ubitux> :) [21:32] <Daemon404> " Thank you for your participation in the Coverity Scan program. We are pleased to announce that we have enhanced our service to address the feedback of our members. These enhancements enable you to: " [21:32] <Daemon404> wtf? [21:32] <Daemon404> i dont recall havign a coverity acct [21:58] <cone-963> ffmpeg.git 03Michael Niedermayer 07master:03cd3bec6e66: avformat/iff: Byte seek is unsupported [22:00] <Compn> Daemon404 : you never looked at ffmpeg in coverity ? [22:00] <Daemon404> nope. [22:00] <Compn> its possible that you requested it, and michael sent in your name, but they never got around to making you an account ? [22:02] <Daemon404> Compn, i explicitly said i didnt [22:02] <Daemon404> iirc. [22:03] <Compn> ok then [22:06] <wm4> Daemon404: so what exactly is that "removed" copyright in paul's patch about? [22:08] <Daemon404> what do you mean [22:09] <Daemon404> he moved code from wavpack.c, copyright by kostya, to a new file, with no copyright [22:09] <wm4> ah [22:12] <Compn> is no good to do that [22:12] <Compn> gotta watch it [22:13] <Compn> seen a lot of people make that mistake lately [22:13] <wm4> might get sued by Libav otherwise [22:15] <Compn> wm4 : its in our rules that every file has to have a copyright , diego did a lot of work to get that [22:15] <Compn> standardized [22:17] <Compn> BBB : videolan wants h265 and vp9 decoders in ffmpeg :) [22:22] <JEEB> smarter's HEVC decoder is pretty ready, just needs some finishing up and the OpenHEVC dudes to merge in their useful changes [22:22] Action: michaelni also wants H265 & vp9 in ffmpeg [22:22] <michaelni> decoders and encoder [22:22] <michaelni> S [22:22] <JEEB> the OpenHEVC folk are the worst part I guess [22:22] <Daemon404> supposedly in aug BBB, myself, + others are workign on vp9 [22:22] <JEEB> since they mostly just hack stuff up on their C++ stuff [22:23] <JEEB> and don't even care if their hacked up libavcodec works with upstream [22:23] <JEEB> elenril ended up merging their stuff the last time [22:23] <JEEB> and now they have written even more [22:26] <smarter> I know I keep saying that, but openhevc stuff should get cleaned up/merged Soon//" :) [22:26] <smarter> they just got tiles working which is nice [22:27] <JEEB> it's not your problem, really :s [22:27] <JEEB> it's theirs [22:27] <JEEB> they have no respect for the upstream [22:27] <JEEB> if they had, the commits could be merge as-is [22:27] <smarter> they do, but it's not their priority [22:27] <JEEB> yes, getting money from Ateme is :-P [22:28] <smarter> mraulet keeps me up to date won what they're doing [22:28] <smarter> ateme is sponsoring them? I didn't know that [22:28] <JEEB> well all that PR they're getting by letting Ateme use them as a platform, I would guess they'd be getting money :P [22:28] <Daemon404> 'as a platform' ? [22:30] <JEEB> ugh, I'm kind of tired after a bicycle ride, but what I meant is that all the public PR messages about their project pretty much has Ateme's name on them, so I guess Ateme said that they'd get some money to get something working for various dates [22:30] <JEEB> not that I really know, it just feels just like x264's opencl [22:30] <JEEB> and "x265" [22:30] <JEEB> esp. the first [22:31] <JEEB> which was a case of "everyone works until the OpenCL is 'working' under AMD's money and then it's done. Except it isn't." [22:31] <JEEB> and yes, I have a rather negative opinion on those guys, just by looking at how they lack any kind of courtesy [22:31] <JEEB> (regarding upstream) [22:32] <smarter> well mraulet bought me beer once, that's courteous :p [22:32] <JEEB> well, sure -- your decoder is now being marketed as the first open source solution, with Ateme's name in the press releases :P [22:33] <JEEB> I'm sorry but this just smells like "let's just grab that code and hack it up for our use" :D [22:33] <JEEB> maybe I'm wrong [22:33] <JEEB> I hope I am, actually [22:34] <smarter> the PR stuff comes from the GPAC folks I think, I don't know them/their motivations [22:34] <Daemon404> ugh gpac [22:34] <JEEB> GPAC and OpenHEVC seem to be rather entangled :P [22:35] <Daemon404> i have nothing good to say about gpac [22:37] <JEEB> I think most people who have actually touched the code have that opinion [22:37] <Skyler_> JEEB: I'm willing to cut them a little bit of slack on the opencl mainly because they just got saddled with a mostly-impossible task :/ and it ran months and months beyond the intended date and stuff [22:38] <Skyler_> at least now we will maybe have fewer people asking "where is my gpu x264 support" [22:38] <Skyler_> "it's here, but it's not very good" [22:38] <JEEB> yeah... at least muggs used his food time for it and kind of finished it [22:39] <JEEB> I just used it as an example of a project where "<smarter> they do, but it's not their priority" kind of things lead to "budget ended? well, that's it" [22:39] <JEEB> kind of stuff [22:39] <Skyler_> Yeah [22:39] <Skyler_> I hope that doesn't happenthis time [22:39] <Skyler_> I'm mainly willing to go with it because, gosh, at least they're offering to do something and nobody else is, but... [22:39] Action: smarter is optimistic/naïve :) [22:40] <JEEB> also this was mostly regarding the hacking OpenHEVC/GPAC is doing with smarter's decoder [22:40] <JEEB> not regarding "x265" [22:40] <JEEB> I'd just hate it if elenril or someone had to start merging stuff again from OpenHEVC [22:41] <kierank> 21:27:57 <JEEB> yes, getting money from Ateme is :-P --> that's not true [22:41] <kierank> they get money from the french government [22:41] <smarter> they really should call it "x265" with the quotes :p [22:41] <Daemon404> x264++ [22:41] <JEEB> yes, they're a university-related project so naturally they get money from the French government [22:42] <kwizart> Hello, Can I default to --enable-thumb in arm arches such as armv7-a and armv6 hard float ? [22:42] <JEEB> anyways, I'm just being cynical as always [22:42] <smarter> they're also working with the industry on http://www.4ever-project.com/ [22:42] <kierank> ateme don't give them money [22:50] <JEEB> well, it was just a crazy idea in my head, so you shouldn't really put too much weight in that :) In general it just felt like they were dancing in order to get X ready, and then get whatever was promised from a third party Y. And screw the upstream :D But as I said, I am just being a cynical, negative person in this case. [23:51] <durandal_1707> michaelni: did you subscribed me to coverity spam? [23:57] <durandal_1707> is someone going to port lossywav as filter? [00:00] --- Fri Jun 21 2013

1 0

ffmpeg.log.20130620
by burek 21 Jun '13

21 Jun '13

[00:00] <vhann_> llogan: http://pastebin.com/ES4yvuiH [00:01] <xeberdee> I tried compiling ffmpeg on ubuntu by the guide but it failed on libopus - I'm not sure what to do with the static build? [00:01] <xeberdee> I just get install libavtools [00:02] <llogan> which ubuntu version? works for me on 13.04 [00:03] <vhann_> llogan: Ah shit, let me try again with the video stream this time [00:04] <vhann_> llogan: Same error: http://pastebin.com/CbRAEV72 [00:04] <xeberdee> llogan: Ubuntu 12.04.2 LTS [00:04] <llogan> xeberdee: you're not the first person to experience the issue, but i couldn't duplicate it (yet). i'll try on 12.04 [00:06] <llogan> xeberdee: until then you can see static build instrucitons: http://askubuntu.com/a/270107/59378 [00:08] <llogan> xeberdee: can anything play "THE_DARK_NIGHT1-1.vob"? [00:08] <llogan> oops... vhann_ [00:09] <vhann_> llogan: VLC plays it fine. Let me check mplayer and xine [00:10] <llogan> the audio specifically [00:11] <vhann_> llogan: xine and VLC play the file fine. I can switch audio tracks flawlessly in both [00:11] <vulture> look at the stream details for it [00:11] <vulture> and see what it says for the audio stream [00:14] <vhann_> vulture: Me? [00:15] <vulture> yes [00:15] <vulture> see if vlc's stream info matches ffmpeg's ? [00:16] <llogan> vhann_: can you create a sample and upload it somewhere? "dd if=THE_DARK_NIGHT1-1.vob of=output.vob bs=1024 count=10000" [00:19] <vhann_> vulture: http://postimg.org/image/escpp4ogd/ [00:22] <vulture> so stream1 is the audio? [00:23] <xeberdee> llogan: thanx for the help I got the static working. [00:24] <llogan> xeberdee: i updated the guide so compiling should now also work on 12.04. [00:24] <vulture> would be nice to see a more detailed stream list than w/e that app is spitting out [00:25] <vulture> especially since the streams dont match up with what ffmpeg says [00:36] <vhann_> vulture: llogan: I msg'ed you the output.vob file's URL [00:37] <llogan> thanks [00:40] <llogan> but i should have asked for a section with sound (using skip option in dd). [00:41] <vhann_> llogan: Gimme a min, I'll send you the first 100Mb [00:42] <elkng> what ? pirating ? [00:45] <llogan> vhann_: 10 mb should be fine if you skip the warnings and assorted other crap. [00:47] <vulture> Format : AC-3 [00:47] <vulture> Channel(s) : 6 channels [00:47] <vulture> ffmpeg says: Stream #0:1[0x80]: Audio: ac3, 0 channels [00:47] <vulture> so yeah ffmpeg fails [00:49] <vulture> but anyway, maybe it doesnt matter, what's your goal with it anyway? to re-encode it? or just stream copy? [00:50] <vhann_> vulture: I wanted to make a short derivative work to poke fun at a friend [00:51] <vhann_> It's allowed in Canada as the clip is short enough [00:51] <vhann_> s/is/would be/ [00:56] <vulture> different commandline shows 5.1, and channels=6, though I still get error [00:56] <vulture> maybe you can specify a manual option to merge to 2 channels [00:56] <vulture> idk [01:02] <xeberdee> llogan: ffmpeg compiled fine on ubuntu 12.0.4 LTS now [01:03] <xeberdee> thanx [01:03] <llogan> no problem [01:03] <vulture> vhann_: virtualdub has no problem opening/parsing/converting this vob [01:03] <vulture> even with audio [01:03] <vulture> so probably an ffmpeg issue idk [01:03] <llogan> ffmpeg will recognize the correct number of channels (after the 0 channel junk) with -analyzeduration 10000000 -probesize 10000000 [01:03] <llogan> as input options of course [01:04] <llogan> ...but it still ignores the other streams. [01:14] <vulture> just use virtualdub instead? :P [01:14] <vhann_> vulture: ... you would make a terrible marketer :p [01:15] <vulture> shrug, if it doesnt work out of box, try something else! [01:16] <vulture> I wish I could get ffmpeg to work in my code but it breaks down occasionally =/ [01:17] <durandal_1707> normal way to report issue is to fill bug report [01:24] <vulture> from the sheer number of open bug reports I'm guessing the normal way isnt very useful :D [01:24] <durandal_1707> whatever, looks like you are not interested in helping project [01:25] <vulture> I'm just being realistic [01:25] <vulture> I've submitted dozens of bug reports for other projects as well, and some even from like 2001 are still unanswered :P [01:27] <durandal_1707> vulture: you are not realistic, just extremly ignorant and lazy, you did not looked at list of closed bug reports [01:32] <vulture> well I can put in a ticket after work and we'll see [02:17] <Jan-> are you guys seriously trying to tell me that http://ffmpeg.org/libavcodec.html is the ENTIRE DOCUMENTATION to libavcodec? [02:17] <Jan-> It's two sentences! [02:17] <durandal_1707> no its not [02:19] <vulture> Jan-: http://ffmpeg.org/doxygen/trunk/group__libavc.html [02:20] <vulture> though google may be easier to navigate to where you want there [02:20] <Jan-> Okay, that's reference, but it isn't so much documentation. [02:20] <vulture> or actually it also has a search in the upper right [02:20] <Jan-> is there a broad architectural overview anywhere as to how the thing is supposed to work, at a high level? [02:20] <vulture> gl with that :D [02:20] <vulture> at least they include sample programs now [02:21] <vulture> in the -dev download, there's a doc/examples [02:21] <Jan-> What I want to do is get from a file on disk to (presumably) a pointer to some audio samples. [02:21] <Jan-> Presumably that's possible somehow. [02:21] <Jan-> But nowhere is it actually written down what the basic steps are to do that. [02:22] <vulture> demuxing.c in the doc/examples directory is what I used [02:22] <vulture> it's decent [02:22] <vulture> it's missing a couple key points though [02:23] <Jan-> I've found lots of examples but none of them ever work as they're more than 7 seconds old, which in ffmpeg is.... ages... [02:23] <durandal_1707> nonsense [02:23] <vulture> the examples included work [02:24] <vulture> demuxing.c [02:24] <vulture> try that one [02:24] <Jan-> In any case, my interest is in writing a C# interop layer for it, so I'd rather not have to dive into C if I can possibly avoid it. [02:25] <vulture> well, I dont think that's entirely unavoidable if you want a complete wrapper [02:25] <vulture> *avoidable [02:26] <Jan-> well it *ought* to be, if there were any docs. [02:26] <vulture> "open source" :P [02:26] <Jan-> Yeah. I know. [02:26] <durandal_1707> Jan-: there is doxygen [02:26] <Jan-> Compiling a useful avcodec and avformat under windows is not easy. [02:27] <Jan-> Wait, what am I saying, it's not easy under linux, it's downright impossible under windows [02:27] <vulture> just use the prebuilt [02:27] <Jan-> Sure, love to [02:27] <durandal_1707> Jan-: its possible to build on windows just fine [02:27] <durandal_1707> stop spreading lies [02:28] <vulture> building on windows isnt as impossible as it was 5 years ago [02:28] <vulture> where mingw was a big pile [02:28] <Jan-> AVFormatContext seems to have... quite a lot of members. [02:28] <vulture> anyway, there are legit prebuilts now :P [02:28] Action: Jan- looks a bit alarmed [02:28] <durandal_1707> you can compile with msvc now [02:28] <vulture> can you really [02:28] <vulture> how is that accomplished :P [02:28] <Jan-> Quite a lot of those members are, er, a bit complex in themselves. [02:28] <durandal_1707> there is fucking documentation [02:29] <vulture> msvc doesnt support c99 [02:29] <vulture> and it never will afaik [02:29] <durandal_1707> irrelevant [02:29] <vulture> seems pretty relevant [02:30] <Jan-> Eesh, you can have structs with structs in 'em? [02:30] Action: Jan- looks a bit alarmed [02:30] <vulture> well that's just basic programming :P [02:31] Action: Jan- isn't really a C coder [02:31] Action: Jan- knows enough to be dangerous [02:31] <vulture> "FFmpeg can be built with MSVC using a C99-to-C89 conversion utility and wrapper." [02:31] <vulture> slick [02:32] <vulture> although, it still requires msys, so that almost defeats the purpose.. :P [02:33] <vulture> Jan-: well in c# it has the same thing [02:33] <vulture> class/struct within a class/struct is quite common [02:33] <Jan-> I think if I wanted to directly invoke avformat, at least, I'd have to implement about 8 different structs with several hundred members overall. [02:34] <Jan-> Which is likely to be a suckfest. [02:35] <vulture> oh yes [02:35] <vulture> well certain other mpeg apis have several thousand members [02:38] <Jan-> I guess I'm just used to object oriented languages where "context" is implied and having to pass around this enormous AVFormatContext seems a bit, er, basic? [02:40] <vulture> right it's basically just implied/hidden for you in a class [02:40] <vulture> it's not really much different [02:41] <Jan-> No, it's exactly the same! Except for this enormous, extremely complicated struct. [02:41] Action: Jan- skritches her head [02:42] <vulture> you arent rewriting the whole struct every time, it's just 1 extra parmaeter [02:42] <vulture> *parameter [02:43] <vulture> it's being passed by reference :P [02:43] <Jan-> Sure, but if I'm gonna platform-invoke one of these APIs in C#, I need to create this struct. By hand. Manually. In my own code. [02:43] Action: Jan- looks a bit uncomfortable [02:43] <vulture> yeah have fun with that [02:44] <vulture> maybe you can copypaste the c headers or write a generator to parse them [02:44] <Jan-> It'd be nice to FIND the C headers. [02:44] <vulture> ffmpeg-20130613-git-443b29e-win64-dev\include [02:44] <Jan-> Oh. [02:44] <Jan-> Looks like I can't even do that. [02:44] <vulture> I'm just using this from the precompiled win64 package :P [02:45] <Jan-> Oh. [02:45] <Jan-> Seems there's a method (er, function) called avformat_open_input() which handles from-file input [02:46] <Jan-> which internally creates said struct. [02:46] <vulture> yeah [02:46] <Jan-> That said I might still need the layout of the damn thing. [02:47] <Jan-> I'm not sure if it's open_input or open_input_file [02:48] <Jan-> hm no you still need to pass one in [02:48] <vulture> I use avformat_open_input [02:49] <vulture> with a custom AVIOContext to read from memory [02:49] <Jan-> I have "stream" or "file" in this particular set of docs [02:49] <Jan-> but I have no idea when it's from [02:49] <Jan-> also it seems to need you to tell it what format the file is in [02:49] <Jan-> or is that an "out" parameter [02:50] <Jan-> is ANY of this documented? ANYWHERE? [02:50] <vulture> no idea [02:50] <vulture> just step through the demuxing.c example I mentioned [02:50] <vulture> it gives the basic process [02:50] <vulture> it will figure out the format for you generally [02:51] <Jan-> humph [02:52] <Jan-> this doxygen stuff is great for producing a lot of verbosity [02:52] <Jan-> but not actually very explanatory [03:03] <Jan-> do I need to start looking somewhere else for information on this thing [03:04] <Jan-> I mean, I'm looking at ffmpeg.org/libavutil.html [03:04] <Jan-> the entire documentation is three lines. [03:04] <vulture> yes I told you where [03:04] <vulture> several times [03:05] <Jan-> I'm talking about docs, not code. An actual overview of how it's supposed to operate. [03:05] <Jan-> Not just a list of function names and their parameters. That doesn't tell me much. [03:06] <vulture> yeah already answered that too, but I'm no expert [03:06] <Jan-> Looking around I found http://ffmpeg.org/trac/ffmpeg/wiki/Using%20libav* [03:07] <Jan-> but really it just goes on about how there isn't much of any documentation and what there is tends to be outdated, which I kinda figured out for myself. [03:12] <Jan-> I'm lookint at demuxing.c and there's almost no comments in it [03:13] <Jan-> it's not actually explained anywhere what the code is *intended* to do, let alone how it does whatever it is [03:14] <vulture> it wasnt that b ad [03:14] <vulture> wasnt great [03:14] <Jan-> sure vulture [03:14] <vulture> but was easy to follow imo [03:14] <Jan-> but... I mean... "demuxing.c" [03:14] <Jan-> what is it MEANT to do? [03:14] <Jan-> It doesn't even explain the purpose of the code, how it fits into what you might want to do in a practical application. [03:15] <Jan-> I don't think I'm particularly stupid but... I mean... how does this help? [03:15] <vulture> takes a file, demuxes it, and then decodes the audio/video streams into raw data [03:15] <vulture> * libavformat demuxing API use example. [03:15] <vulture> * [03:15] <vulture> * Show how to use the libavformat and libavcodec API to demux and [03:15] <vulture> * decode audio and video data. [03:15] <vulture> :P [03:15] <Jan-> it doesn't define what "demux" actually means in that context [03:16] <Jan-> demux as in unpack the compressed video frames? [03:16] <Jan-> demux as in separate out the audio and video? [03:16] <vulture> if you dont know that then you need to tke a basic multimedia course [03:16] <vulture> I can give you a nutshell [03:16] <Jan-> I've been working in television production for ten years. [03:16] <Jan-> "demux" is a very, very generic term. [03:16] <Jan-> If there's a specific meaning of it that's useful with regard to ffmpeg, fine, but someone needs to WRITE THAT DOWN SOMEWHERE> [03:17] <vulture> it's generic, and it also has a specific meaning for multimedia files [03:17] <vulture> you have codecs, which are encoder/decoders for specific types of data, e.g. audio, or video [03:17] <Jan-> one of the functions is called "decode_packet" [03:17] <vulture> libavcodec handles this [03:17] <Jan-> is this something to do with networking? ethernet packets? what? [03:17] <vulture> then you combine or mux those encoded streams into a container format, which is what libavformat does [03:18] <Jan-> I get the impression that avformat emits "packets" of some sort but I only got that by randomly inspecting some other file. [03:18] <vulture> when you demux a video file, you're pulling back out those compressed streams [03:19] <Jan-> Okay. Fine. [03:19] <Jan-> Now if you'd liketo imagine me screaming this into a megaphone at the top of my voice [03:19] <Jan-> WHY DOESN'T IT SAY THAT IN THE GODDAMN DOCUMENTATION. [03:20] <vulture> I think they expect you to know that going in [03:20] <Jan-> How?!?! [03:20] <Jan-> That's their own private meaning of the word "demux"!?!? [03:20] <vulture> no thats the entire world's definition :P [03:20] <Jan-> No, no, it really isn't. Go talk to a satellite engineer. [03:20] <vulture> it's actually a very very common word [03:21] <Jan-> Now on to this "packet" thing [03:21] <Jan-> is this network code or what [03:21] <vulture> dont think so [03:21] <Jan-> so... what then... [03:22] <vulture> just generic packets of data? idk :P [03:22] <vulture> I wrote a wrapper around ffmpeg so that I dont have to touch any internals [03:22] <vulture> and also so that I dont have to think about ffmpeg's weird api anymore [03:22] <Jan-> I don't blame you :/ [03:22] <vulture> several years ago we hired a guy to do it for us [03:23] <vulture> and it was highly broken [03:23] <vulture> it's changed a lot since then though... [03:23] <vulture> they include example source code now! [03:23] <Jan-> *sigh* I think a lot of it is about that to be honest [03:23] <vulture> and it even compiles out of box, and actually runs, without (too many) errors [03:23] <Jan-> I know a few people make a LOT of money out of consulting on ffmpeg [03:23] <Jan-> I have a feeling there's a bit of job protectionism going on [03:26] <vulture> I've used several commercial video libraries [03:26] <vulture> they're nice [03:26] <vulture> open() read() close() :P [03:27] <vulture> quite literally that easy [03:27] <Jan-> in all honesty 95 plus per cent of cases would be pcm audio in either avi, or mp4/quicktime/whatever [03:27] <Jan-> I'm just aware of how tricksy some of these file formats can be and how they can bite you in the ass [03:27] <Jan-> so I'm not that anxious to start writing RIFF file parsers. :/ [03:27] <vulture> windows provides a lot of that for you [03:28] <vulture> if you're relying on windows, maybe there's a directshow wrapper for c# ? [03:28] <vulture> you can use ffmpeg through directshow [03:28] <Jan-> There is. [03:28] <vulture> (but directshow sucks) [03:28] <Jan-> It does. [03:28] <Jan-> It really does. [03:28] <vulture> I did the wrapper thing for directshow too. lol :P [03:28] <Jan-> To be fair it sucks because it's extremely lightweight and therefore pretty quick. [03:28] <vulture> riiiiiiiiiiiiiiiiiight [03:28] <Jan-> I think people expect too much of it [03:29] <Jan-> Well, it depends what you try to do with it. [03:29] <vulture> lightweight :P except really slow and lacks all precision [03:29] <Jan-> Start running filters that do a lot of stuff and obviously it's gonna suck ass.\ [03:29] <vulture> it's based on COM so it's automatically heavy bloat [03:29] <Jan-> Precision I'll give you but in our case it was a driver fault with third party hardware. [03:30] <Jan-> totally bent 10 bit RGB video. [03:30] <Jan-> unusable. [03:30] <Jan-> But really I don't want to have to go into a whole bunch of DS code just to get the first second of audio out of some files. [03:31] <Jan-> *especially* as we then end up with complex end user requirements to install codecs for all their target files, which is a sucky thing to ask. [03:32] <vulture> if it's wav or mp3 then it's trivial [03:32] <vulture> (with windows) [03:32] <Jan-> it's video files from cameras [03:32] <vulture> if it's something else [03:32] <vulture> ah [03:33] <Jan-> now I don't actually need the video, I need the audio tracks [03:33] <Jan-> so if it's prores or something it doesn't really matter [03:33] <vulture> if you dont care about performance or latency, you could just shell to ffmpeg.exe [03:33] <Jan-> I could. But I do care about those things a bit. [03:33] <Jan-> And man, what a sucky solution :/ [03:34] <Jan-> It just seems crazy that they'd write all that code and then... you know... just sort of not bother writing it up in any useful way. [03:34] <Jan-> it's absurd [03:34] <Jan-> it's just completely stupid, why would you do that [03:36] <vulture> busy, lazy, or the "open source" excuse [03:37] <Jan-> Gah :( [03:39] <vulture> alternatively, make your own wrapper with simple structures (in c), and then you can call just the wrapper from c# without having to copy all the complex crap [03:39] <Jan-> That was how I first thought about doing it [03:39] <vulture> you have to do some c work but imo it needs a wrapper anyway [03:39] <Jan-> then I realised how often they change the API [03:40] <vulture> heh [03:40] <Jan-> I mean it changes several times a DAY Often [03:40] <vulture> it's more stable now than it was [03:40] <Jan-> it seemed like that layer would be a huge maintenance issue [03:40] <vulture> and you'd have to change it MORE if you were to use it natively in c# [03:40] <vulture> because your entire interop would become invalid [03:40] <Jan-> every time we updated the libav stuff it would be basically certain we'd need to hack the C [03:40] <vulture> whereas with a c wrapper it's mostly just the wrapper [03:40] <Jan-> sure but the main issue is that the only IDE I have for C is, you know, notepad.exe and gcc.exe [03:41] <vulture> I use notepad too :) [03:41] <Jan-> I'd really rather be able to just take avcodec.dll and avformat.dll and go from there [03:41] <Jan-> yes it will be tricky, but at least you're starting from a known place [03:41] <Jan-> or at least you would be IF THERE WERE ANY DOCS. [03:45] <durandal_1707> DOCS of what? [03:45] <Jan-> from what I read most windows compiles of ffmpeg or the libraries are cross compiled from linux anyway [03:45] <Jan-> so if we were to do a C wrapper we'd probably have to write that on linux, for windows, and cross compile it\ [03:45] <Jan-> and I am NOT really into that thanks [03:46] <vulture> you dont need linux at all [03:46] <vulture> my wrapper is a single .c compiled via mingw(gcc) to a .dll/.lib [03:47] <vulture> all windows [03:47] <Jan-> well, we can't even get ffmpeg to build with none of the libraries that actually make it useful [03:47] <Jan-> so that's a nonstarter [03:48] <vulture> do you have a reason to build it yourself? (e.g. lgpl or something) [03:48] <vulture> (or you want a stripped down version?) [03:48] <Jan-> couldn't care less [03:48] <vulture> if not save yourself the hassle and use the precompiled [03:48] <Jan-> the licencing is probably irrelevant, the people who will be using the final product won't even know what "source code" is. [03:49] <vulture> http://ffmpeg.zeranoe.com/builds/ [03:49] <Jan-> yeah, I know [03:49] <Jan-> that's what we'd do [03:50] <Jan-> in all honesty we probably would never need to update anyway [03:50] <Jan-> 99 per cent of all commits on ffmpeg seem to be minor performance tweaks and network security, neither of which would bother us much [03:51] <vulture> if only they could make avformat_open_input actually work :D [03:51] <vulture> seems KIND OF IMPORTANT [03:51] <Jan-> I have no clear idea what it's supposed to do. [03:52] <Jan-> I have no idea what ANY of it is supposed to do, how it works, what you pass from where to where, what half the terms mean. [03:52] <vulture> try to find some tutorial then [03:52] <Jan-> I mean jesus, I know once you've been writing code for a project for years it all starts to become obvious TO YOU, but they need to get a goddamn clue. [03:52] <Jan-> good docs are coding 101, ffs [03:52] <vulture> great coders dont have time for documentation ;) [03:52] <Jan-> I'm sure they think so. [03:53] <durandal_1707> what docs is missing? [03:54] <vulture> he wants a primer/overview doc [03:54] <Jan-> *ahem* it's short for Janine :) [03:54] <Jan-> all there is, is API reference. Which is fine. But there's no actual description of how it's supposed to work. [03:54] <Jan-> There's no intro, no overview. [03:55] <Jan-> The examples are practically uncommented and rely on a lot of knowledge complete newbies to ffmpeg won't have. [03:55] <Jan-> There's no way to *start*. [04:03] <vulture> sorry I know a lot of guys named Jan :P [04:04] <Jan-> everyone assumes I'm a Swedish guy. [04:04] Action: Jan- scowls [04:04] <vulture> or any central euro country too :P [04:04] <Jan-> mutter [04:04] <Jan-> grumble [04:09] <vulture> anyway, if there was real docs I might not be here either, since I cant seem to find how custom io contexts are supposed to handle pending data reads [04:15] <Jan-> I need to sleep [04:15] <Jan-> it's like 3am here [04:15] <Jan-> thanks vulture [04:15] <Jan-> ...no thanks ffmpeg people :/ [06:57] <praveenmarkandu> can someone explain when exporting a hls formatted m3u8 using the -hls commands in FFMPEG, why the quality is lower than using the segment format [08:56] <ilove11ven> command line you use? [09:57] <khali> ubitux: I have investigated my "Non-monotonous DTS in output stream" warning flood issue which I see with recent version of ffmpeg and not with 1.0.6 [09:58] <khali> ubitux: it only happens when input file has a damaged audio stream, and only when using -af aresample=async=24000:first_pts=0 [09:58] <khali> it takes the combination of async > 1 and first_pts=0 to trigger the warning flood [09:59] <khali> ubitux: I hit this because my encoding script uses -async, which translates to these aresample options now [09:59] <khali> in earlier versions it translated to different options, async and first_pts did not even exist [10:07] <ubitux> well, maybe you should open a ticket then ;) [10:08] <khali> ubitux: yes, I think I have enough information (and a sample) to open a decent ticket [10:09] <khali> ubitux: I found two tickets related to -async (#2421 and #2309) but the warnings are different [10:09] <khali> so I suppose these are different issues [10:09] <khali> I'll open a separate ticket [10:11] <khali> ubitux: one thing I don't get is that -async is deprecated and the man page says to use the asyncts audio filter instead [10:11] <khali> still when using -async ffmpeg translates it to -af aresample, not -af asyncts [10:12] <ubitux> the manpage should not be saying that since a while now [10:12] <ubitux> make sure that's not the fork manpage [10:13] <khali> ubitux: fork manpage? [10:15] <ubitux> my manpage from 1.2 doesn't say so [10:15] <khali> I was reading the 1.0.6 man page and you're right, the current version says -af aresample [10:15] <ubitux> so you're using an old ffmpeg, or the fork [10:15] <ubitux> ok, first choice then [10:15] <khali> I did not know there was a fork of ffmpeg, who would be silly enough to do that? [10:16] <ubitux> random drama [10:29] <saste> khali, never underrate people silliness [10:31] <khali> saste: true enough :( [10:36] <praveenmarkandu> @ubitix is it possible to point me to the person who committed the hls code from libav? [10:40] <ubitux> praveenmarkandu: assuming you're talking to me, i think that's lu_zero, but as you can guess he's not contributing directly to this project [10:45] <praveenmarkandu> @ubitux yeah im talking to him now on #libav [10:54] <khali> what's the attachment size limit in trac? I have a bug sample but it's a bit large - 3.4 MB [10:59] <khali> hmm, 2.5 MB that is [11:20] <khali> ubitux: ticket created, #2693 [11:21] <khali> I don't know if I should start bisecting it right now or if developers will be smart enough to figure it out just by looking at the code [11:23] <edoardo> How do I record audio with ffmpeg from microphone? [11:24] <edoardo> I tried ffmpeg -f dshow -i video="VIDEO_DEVICE":audio="MICROPHONE" "out.flv" [11:48] <keyzs> sirs for converting a ffmpeg -i input.flv output.mp4 with multiple files what would be the sintaxe command? [11:51] <khali> keyzs: define "multiple files" and what output(s) you expect from them [11:53] <keyzs> khali now that iŽm thinking its more multiple inputs and multiple outputs, so, ffmpeg -i input1.flv input2.flv.. output1.mp4 output2.mp4 [11:54] <keyzs> i have 156 files on this [11:55] <Mavrik> there is no command for that [11:55] <Mavrik> use a bash script [11:55] <Mavrik> or anything else like that [11:56] <khali> keyzs: for i in `seq 1 156` ; do ffmpeg -i input$i.flv output$i.mp4 ; done [11:56] <khali> or something similar [11:57] <khali> (assuming a bash-like shell...) [11:57] <keyzs> yes [11:59] <orioni> hi to all . i`m trying to use ffprobe to get info about a multicast channel but the script doest timeout in case there is no data on the multicast channel [11:59] <orioni> any help [11:59] <orioni> ? [11:59] <keyzs> khali how long would ffmpeg take to convert 156 files from flv to mp4 assuming each file has like 3mb [12:00] <khali> keyzs: it depends on too many factors... CPU speed, video and audio codecs, codec options... [12:00] <keyzs> in average [12:00] <khali> keyzs: just measure how much it takes for one, and multiply by 156 [12:00] <keyzs> have a quad core 2.66ghz with 8mb ram [12:00] <orioni> so if there is no stream on the udp://239.123.10.40:58040 for whatever reason ( broken pipe , no link ) i want the ffprobe to timeout after X amount of time [12:01] <Mavrik> orioni, you can pass timeout for udp as part of url [12:01] <khali> (or convert two and multiply by 78, for better accuracy) [12:01] <Mavrik> orioni, check input protocol documentation. [12:01] <keyzs> khali ok [12:01] <keyzs> thanks for the info [12:01] <orioni> Mavrik: can you send me url .... i checked but nothing found [12:02] <khali> keyzs: you can use -threads 2 to parallelize if encoding with libx264 (and probably other codecs too) [12:02] <orioni> tried with probesize , timeout , max_delay , analyzeduration but it doesnt timeout [12:02] <Mavrik> orioni, go to ffmpeg site, click "documentation", click "ffmpeg-protocols", click "udp" [12:03] <orioni> ok , gime a sec [12:03] <keyzs> khali by my calculations an average of 30m [12:04] <orioni> i see , so it will timeout after 3 secs with this ffprobe udp://239.123.10.40:58040?timeout=3000 ? [12:05] <keyzs> khali, the command will work nice if the files have different names? [12:05] <Mavrik> orioni, yep :) [12:05] <orioni> thanks , lets try [12:05] <Mavrik> orioni, actually, it'll timeout after 0.3 secs [12:05] <Mavrik> since unit is microseconds not milliseconds [12:07] <orioni> not good .... i did used the comand with 3000 timeout but it takes like 12-13 secs to "terminated" [12:10] <orioni> http://pastebin.com/ibk5ytiV [12:12] <Mavrik> orioni, avprobe isn't ffmpeg. [12:12] <orioni> but on debian it says that ffprobe is replaced by avprobe [12:13] <orioni> lrwxrwxrwx 1 root root 7 Mar 24 08:26 /usr/bin/ffprobe -> avprobe [12:14] <Mavrik> *shrug( [12:15] <Mavrik> orioni, sorry, but I have no idea how libav people messed that up and we really can't support everything distros mess up in their package managers :) [12:15] <Mavrik> I think #libav is the channel for libav project support here on freenode [12:15] <orioni> *** THIS PROGRAM IS DEPRECATED *** ................ This program is only provided for compatibility and will be removed in a future release. Please use avconv instead. [12:16] <orioni> ok man , thanks [12:24] <praveenmarkandu> @ubitux, would you know the difference between using the -f segment command and just using the newer hls commands [12:25] <ubitux> hls muxer is simpler iirc [12:25] <ubitux> it was a NIH from libav when they saw we updated their segment muxer to add HLS support [12:27] <praveenmarkandu> yeah the command looks simpler [12:34] <praveenmarkandu> thanks [12:34] <praveenmarkandu> when running 3 ffmpeg transcodes at a time, i get around 15 fps per transcode [12:35] <praveenmarkandu> when one process finishes, the fps for the remaining two do no increase [12:38] <khali> praveenmarkandu: I suppose you have more than 3 logical CPUs (cores or HT sibling) and you did not enable threaded encoding in ffmpeg [12:40] <praveenmarkandu> khali, if not specified, isnt it automatic threading? [12:41] <khali> praveenmarkandu: to my knowledge, no, by default no threading takes place [12:41] <praveenmarkandu> threads is set to 0 [12:41] <praveenmarkandu> -threads 0 [12:46] <durandal_1707> for what? [12:49] <khali> keyzs: no, my example assumed uniform naming of input files [12:50] <keyzs> khali could you give me a hint on non uniform? [12:50] <khali> keyzs: if you need something more generic, try the following: [12:51] <khali> for file in *.flv ; do ffmpeg -i "$file" $(echo "$file" | sed -e 's/\.flv$/.mp4/') ; done [12:51] <khali> (might need extra quotes if file names contain spaces or special characters) [12:53] <khali> praveenmarkandu: the man page says auto... could be that auto works OK for internal codecs but explicit -threads is needed for external ones [12:56] <orioni> hi again , how can i get the stream name / provider name by using the Json output format with ffprobe [12:57] <orioni> i mean , the "metadata" [13:02] <praveenmarkandu> ok [13:04] <praveenmarkandu> khali: i currently launch 3 seprarate ffmpeg processes [13:04] <praveenmarkandu> shoudl i launch 1 ffmpeg process but make multiple outputs? [13:26] <MArcin2> hi everyone [13:28] <MArcin2> may I ask a question about cutting h264 clip? [13:33] <MArcin2> I have two files recorded by the hardware card (Sensoray 2253), when I try to cut it one cuts almost imedietly and on the second ffmpeg counts zeros from the begining of clip to the point I want. I use that command line ffmpeg -ss 00:19:00 -t 00:00:45 -i "source.mpg" -qscale 0 -vcodec h264 -f mp4 -y dest.mpg [13:34] <MArcin2> why for some h264 it counts those frames at the begining on some not [13:34] <MArcin2> why for some h264 it counts those frames at the begining on some not? [13:41] <MArcin2> anyone knows? [14:35] <Mavrik> MArcin2, because your encoder probably puts I-frames too far between each other [14:35] <Mavrik> and ffmpeg can't find one at that position [14:35] <Mavrik> that is, if you ALWAYS use that command pasted [14:38] <MArcin2> but when for example I frame is each 10 secs [14:39] <MArcin2> will i find the on on 0:10:00 and then goes to 0:19:00 [14:41] <MArcin2> the files differ with tbn param [14:42] <Mavrik> I don't understand what are you trying to say. [14:42] <Mavrik> what does "I find onon 0:10:00& " mean? [14:42] <Mavrik> and why do you have I frames only on every 10 seconds? [14:42] <t4nk530> Hey guys anyone know why my interrupt_callback code isn't being fired with the latest version of ffmpeg? [14:57] <MArcin2> 10 sec is default value of -g param , and i think the hardware procduces the clip also with I frame each 10 secs [15:13] <t4nk275> my interrupt code has stopped working with the latest version of ffmpeg - when i call avformat_open_input with no network connection the application hangs - the interrupt is never called - anything changed here? [16:35] <khali> is there anywhere I can upload large sample files which do not fit in trac? [16:36] <khali> I seem to recall reading something about that a few days ago, but I can't find it again [16:36] <JEEB> the mplayerhq ftp [21:16] <jimjones> Does anyone know why this doesn't work? cat animated.gif | ffmpeg -f gif -i - example.avi? It gives a "pipe:: Input/output error" and "Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)". if I just do: ffmpeg -f -i animated.gif example.avi, it works fine. [21:17] <durandal_1707> because there is no gif parser [21:18] <durandal_1707> and gif demuxer needs seekable input [21:20] <durandal_1707> i did not explored if this can be fixed, aka writing gif parser [21:20] <durandal_1707> eg. if its possible at all [21:22] <jimjones> thanks [21:42] <tomlol> I have a bunch of videos which I want to stream randomly, is there a way to keep a stream always up and just push new content to it? Or a way that I could setup a callback to grab the next file whenever its ready for it? [21:43] <tomlol> I tried a named pipe, but twitch thinks it disconnects in between every file. [22:04] <kwizart> http://paste.fedoraproject.org/20007/71758436/ [22:04] <kwizart> does this build error talk to someone ? (from ffmpeg 1.2.1) [22:05] <klaxa> >-marm [22:05] <klaxa> building for arm architecture? [22:06] <klaxa> afaik x86 asm doesn't work on arm :V [22:06] <klaxa> did you run ./configure correctly? [22:07] Action: kwizart checks the configure line [22:09] <kwizart> + ../configure --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib --mandir=/usr/share/man --arch=armv7hl '--optflags=-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -grecord-gcc-switches -march=armv7-a -mfpu=vfpv3-d16 -mfloat-abi=hard' --enable-bzlib --disable-crystalhd --enable-frei0r --enable-gnutls --enable-libass --enable-libcelt --enable-lib [22:09] <kwizart> dc1394 --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopencv --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug - [22:09] <kwizart> -disable-stripping --shlibdir=/usr/lib --disable-runtime-cpudetect --arch=arm --disable-neon [22:09] <kwizart> oups, I could have pasted it [22:10] <xreal> Can I set the episode of mp4 using ffmpeg ? [22:16] <klaxa> kwizart: what compiler are you using? [22:17] <klaxa> or rather, are you cross-compiling? [22:24] <kwizart> klaxa, gcc 4.8.1 native compilation [22:24] <klaxa> ah hmm... i have no arm computer at hand, sorry :( [22:25] <kwizart> I'm about to switch to --enable-thumb for armv7-a, this file only support arm/asm for armv6 without thumb and neon as I understan [22:25] <kwizart> d [22:26] <kwizart> well, actually I wonder if this is thumb or thumbee [22:31] <dagerik> trying to downscale a video here [22:31] <dagerik> The encoder 'aac' is experimental but experimental codecs are not enabled, add '-strict -2' if you want to use it. [22:32] <dagerik> adding that option yielded the same error msg [22:32] <xreal> Can I set the episode of mp4 using ffmpeg ? [22:32] <klaxa> try libfdk-aac instead https://ffmpeg.org/trac/ffmpeg/wiki/AACEncodingGuide [22:32] <klaxa> dagerik: ^ [22:33] <klaxa> xreal: what does "episode" mean in this context? [22:39] <xreal> klaxa: TV series in iTunes. episode, season etc. [22:39] <klaxa> sounds like metadata then [22:40] <klaxa> http://wiki.multimedia.cx/index.php?title=FFmpeg_Metadata#QuickTime.2FMOV.2…. [22:40] <klaxa> that maybe? [22:42] <xreal> klaxa: I can try [22:45] <beginer_user> hey everybody, I have a "little" question, im begginer though. Im on Win 7, and trying to capture video and audio both. The command I use: "ffmpeg -f dshow -i audio="virtual-audio-capturer":video="screen-capture-recorder" -vcodec libx264 -r 25 -crf 1 -ac 1 -acodec aac -strict -2 -ar 44100 -pix_fmt yuv420p -q 1 -y -f flv output.flv" the video is perfect but the sound is lagging. If I record or [22:45] <beginer_user> the video, or the audio standallone, it works very good. They seems to conflict together. I have a "good" computer, and recording to disk, so cant make it faster... "real-time buffer 259% full! frame dropped!" I got though [22:45] <dagerik> klaxa: okay. but why am i not able to use the builtin free aac encoder? [22:45] <klaxa> you should be able to, it is disencouraged though, since it's not very good [22:46] <dagerik> help me use it [22:46] <dagerik> i added -strict -2 [22:46] <dagerik> didnt work [22:46] <klaxa> pastebin your output of ffmpeg [22:48] <dagerik> klaxa: http://bpaste.net/show/108666/ [22:48] <beginer_user> klaxa, if its for me: http://pastebin.com/MCZPtKmG [22:49] <klaxa> include your command and encoding output, right now it's just input analyzation [22:52] <klaxa> beginer_user so if you don't record audio it works well? sorry, i don't run windows, i don't know a thing about dshow [22:53] <dagerik> klaxa: turns out i had to place -strict experimental after -i option but before -acodec [22:56] <beginer_user> klaxa yes. And if I record audio only, it also works good. They fail together, but thx anyway [22:57] <klaxa> maybe you could try a different container... remove the "-f flv" and name the output file output.mkv or something ending in .mkv [22:57] <klaxa> if you specify an output file ffmpeg will guess the container from the extension [22:59] <beginer_user> klaxa, its still the same :) [23:09] <beginer_user> WOW I cant belive it: I separated the video and audio parameters, and it does work! [23:27] <dagerik> i got a lowend laptop using cpu for gfx. im unable to play this 1080p video(mpeg-4, aac). what is a smart way to make it playable? convert process must take less than 1 min. [23:28] <klaxa> what cpu? what mpeg-4 version? [23:29] <klaxa> you are also probably asking for the impossible [23:29] <dagerik> klaxa: Intel(R) Atom(TM) CPU Z520 @ 1.33GHz, Video: mpeg4 (Simple Profile) (mp4v / 0x7634706D), yuv420p, 1920x1080 [23:29] <sacarasc> dagerik: If your CPU can't decode it quick enough, you won't be able to reencode it in under 1 minute. [23:29] <klaxa> does it have an integrated gpu chip? [23:29] <sacarasc> Encoding generally takes longer than decoding. [23:30] <dagerik> it has gma500. it is proprietary. no driver for accelerated gfx [23:34] <klaxa> well it has hardware decoding for h264 :V [23:34] <klaxa> at least according to wikipedia, no idea how to use vaapi though [23:36] <dagerik> mpeg4 -> libx264 [23:37] <dagerik> is there an alternative to h264 here? [23:37] <dagerik> for faster encoding [23:37] <klaxa> no [23:37] <dagerik> im able to encode 1s video in 5s real time [23:37] <klaxa> like i said, you are asking for the impossible [23:37] <dagerik> with 480p [23:37] <vulture-> it's an Atom what do you expect :D [23:38] <vulture-> it's built for low power not performance [23:38] <klaxa> get your videos in h264 and buy a raspberry pi [23:38] <klaxa> or something... [23:40] <vulture-> http://www.techarp.com/article/x264_HD_Results/4_results_01.png you'll have to scroll to the bottom to see the Atoms [23:50] <sacarasc> dagerik: how long is the video? [00:00] --- Fri Jun 21 2013

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