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February 2013

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ffmpeg-devel.log.20130202
by burek 03 Feb '13

03 Feb '13

[00:03] <cone-774> ffmpeg.git 03Carl Eugen Hoyos 07master:985e93a86518: Do not fail for mixed interlaced / non-interlaced YUV4MPEG streams. [00:35] <teratorn> if im generating synthetic audio/video frames for testing, what timestamp-related invariants/constraints should I adhere to in order to keep the muxer happy interleaving the packets? [00:36] <teratorn> oh, err, I meant to ask on #ffmpeg :/ [00:58] <tg2> anybody know where to start looking to override the -RE [01:00] <tg2> to set it more than 1:1? [01:09] <tg2> rate_emu ffmpeg.c line 1642 ? [01:11] <saste> tg2: what's your final goal? [01:12] <tg2> the end goal is to render an mp4 being written to disk to flv so it can be streamed while it is encoding [01:12] <saste> tg2: what's wrong with -re? [01:12] <tg2> can it be made variable? [01:12] <saste> no [01:13] <tg2> if (rate_emu) { [01:13] <tg2> 01643 int64_t pts = av_rescale(ist->pts, 1000000, AV_TIME_BASE); [01:13] <tg2> 01644 int64_t now = av_gettime() - ist->start; [01:13] <tg2> 01645 if (pts > now) [01:13] <tg2> 01646 usleep(pts - now); [01:13] <tg2> 01647 } [01:13] <tg2> seems quite easy to override [01:13] <saste> sure it could be extended, but we need at least a meaningful use case [01:13] <tg2> say you wanted to control the speed of encoding [01:13] <tg2> for resource reasons [01:13] <saste> usually you can control that at the OS level [01:14] <tg2> hm [01:14] <saste> e.g. setting the niceness level [01:14] <tg2> yeah you can do that [01:14] <tg2> the goal actually isn't to save rousrces [01:14] <tg2> since the main encoding process is going as fast as it can [01:14] <saste> which is a more general approach (can be applied to all processes) [01:14] <tg2> this is only doing a streamcopy of the mp4 [01:14] <tg2> into flv [01:14] <tg2> basically rewrapping it [01:14] <saste> -re is usually used for streaming [01:14] <tg2> yea [01:14] <tg2> to simulate a live stream [01:14] <tg2> but if it was variable [01:15] <tg2> you could make -re 1.0 by default [01:15] <tg2> and assigning any other value [01:15] <tg2> would give you realtime * that value [01:15] <tg2> in input speed [01:15] <tg2> this works very well for throttling [01:15] <tg2> for example, the main encoding goes about 10x realtime [01:15] <tg2> so 230~fps [01:15] <saste> tg2: patches are welcome, or open a ticket and discuss [01:16] <tg2> ok great [01:16] <tg2> i'll write it up and test it first [01:16] <tg2> it should be nondestructive in the sense that anybody using -re now will see no difference [01:16] <saste> nice [01:16] <tg2> the logic is quite simple [01:16] <saste> tg2: yes [01:16] <tg2> change the argiment parse [01:16] <saste> if you have a meaningful use case it should be fine [01:16] <tg2> and change the rate_emu to be non-boolean [01:16] <tg2> as far as I can see [01:16] <tg2> this is the only place it is used [01:17] <saste> tg2: but you need a separate option [01:17] <tg2> -respd ? [01:17] <tg2> or -refactor ? [01:17] <tg2> -re -refactor 1.2 ? [01:17] <saste> otherwise you break syntax and get many angry users [01:17] <tg2> shouldn't you can just set it to 1.0 if omitted [01:17] <tg2> that way it acts identical [01:17] <saste> yes [01:17] <tg2> ok so you mean make a second case for arg + argv [01:18] <tg2> with the same switch [01:18] <tg2> hmm [01:18] <saste> uhm no i don't think that would be possible [01:18] <saste> you need a separate option [01:18] <saste> then -re code will wrap around the new option (will set a specific value) [01:18] <tg2> ok [01:19] <saste> we will guide you on ML once you have a working patch [01:19] <tg2> great [01:22] <tg2> another quick question, if you will [01:22] <tg2> actually nvm I'll do some more research first ;) [01:22] <tg2> thanks [02:21] <ulatekh> Hello! I submitted a bug-fixing patch to the mailing-list 6 days ago, but no one commented on it. Is there anything else you need me to do? See http://article.gmane.org/gmane.comp.video.ffmpeg.devel/158184 [02:22] <Compn> ulatekh : dunno who checks makefile stuff [02:22] <Compn> cehoyos does sometimes [02:23] <ulatekh> I just didn't want my bug fix to get lost. And cehoyos seems to have his hands full :-) [02:24] <Compn> hes a busy guy [02:25] <ulatekh> Should I just leave my fix to its fate, then? [02:26] <Compn> 6 days old? [02:26] <Compn> ffmpeg docs say wait 2 weeks for reply, then ping [02:26] <Compn> :) [02:26] <ulatekh> OK...there was a lot to read, I must have missed that...I'll ping later...thanks [02:26] <Compn> we are all on volunteer time here :) [02:26] <Compn> no problem [02:26] <Compn> sorry i cant review, i have no clue ... [02:27] <Compn> barely have my head screwed on [02:27] <ulatekh> Still, it's a shame...I'm like a rash on the open-source world. I track down bugs in myriad projects when I find them, and try to submit fixes...and the vast majority of the time, all I get is apathy. [02:28] <Compn> well [02:28] <Compn> you can ask one guy who works on it [02:28] <Compn> ah hes not online atm [02:29] <Compn> ulatekh : try mailing diego biurrun [02:29] <ulatekh> I'll mail him in 8 days, how's that. [02:29] <Compn> or michaelni is the project admin around here too. he can review patches when hes active [02:29] <Compn> lol ok [02:34] <iive> actually I like the idea of creating libav<module>.<major_version> [02:34] <iive> it would allow different api-breaking versions to be installed for the programs that need them. this includes forks too. [02:34] <Compn> ulatekh : see, iive cares :) [02:35] <iive> i'm cheap at talk :P [02:35] <ulatekh> Right now, the --build-suffix option to configure gives you libav<module><buildsuffix>, and --progs-suffix does similarly for the executables. They work fine, but the pkgconfig files needed a fix. [02:35] <ulatekh> Nice to see someone cares...the apathy I get when trying to fix bugs in open-source code really does get me down sometimes. [02:36] <ulatekh> iive: I've got git-master ffmpeg installed on my machine along with the ffmpeg from my dist's yum repos...they coexist just fine. [02:37] <iive> does buildsuffix separate the include locations too? [02:37] <ulatekh> Yes. [02:37] <ulatekh> The only issue left unfixed (if my patch is considered) is the name of man pages. [02:38] <iive> aha, so your patch is just fixing the pkgconfig files [02:39] <ulatekh> Yeah, I couldn't figure out how to fix the man pages...the bash scripting was way too tense for me ;-) [02:40] <ulatekh> I'm fixing the man pages in my rpm .spec file, post buildroot-install. [02:56] <michaelni> ulatekh, your patch is missing a commit message [02:57] <ulatekh> "Fix pkgconfig files when ffmpeg built with --build-suffix"? [02:57] <ulatekh> I dunno, the commit message kinda writes itself... :-) [02:57] <michaelni> it should start with something like build-sys: or configure: [02:58] <michaelni> also it should list [02:58] <michaelni> what you tested this against [02:58] <michaelni> it seems working fine on ubuntu [02:59] <ulatekh> I tested it on Fedora Core 14, 16, and 17. [03:00] <ulatekh> Sorry, I didn't know all the bureaucratic steps it takes to submit a bug fix. It seems like most open-source projects treat "outsiders" like this...then the insiders check in something that breaks the code 8 ways from Sunday but that's OK. (I finally gave up on mono-devel after a few such outrages.) [03:01] <burek> ulatekh reminds me of me :) [03:01] <burek> but it all boils down to the thing that people dont have enough of free time [03:02] <ulatekh> Still, the fish jumped right in the boat, and filleted itself...all you have to do is eat it :-) [03:02] <michaelni> burek, yep, time is much much more an issue than any bueacrat shit [03:02] <burek> well, i now understand both sides so i dont get upset that much as before ^^ [03:02] <burek> still, some kind of improvement in that way would be welcomed :) [03:03] <ulatekh> Mostly, it bugs me when I get the runaround & "insiders' can break code horribly and then give me the runaround on my attempts to fix the bug. (That hasn't happened to me here, so don't take offense. :-) [03:04] <michaelni> ulatekh, do you want to join become an insider and take over configure maintaince ? [03:04] <burek> ulatekh, dont worry that much, if your fix helps anything, it will be applied sooner or later [03:05] <ulatekh> I'd have a HUGE learning curve to take over something like that. [03:06] <ulatekh> My frustration with ffmpeg is that the code is so persistently undocumented. I've tried to trace bugs before, and have encountered nothing but pages and pages of undocumented god-knows-what. [03:06] <burek> we lack the time to document it while coding stuff [03:06] <burek> and people to document it after the thing has been coded [03:06] <burek> consider constant radical changes here and there :) and you get the idea why is it so :) [03:07] <michaelni> if everyone who didnt understand something would document it once he does figure it out then docs should improve alot [03:07] <ulatekh> I must say, I completely reject the idea that "there's no time to document". Documentation SAVES time by reminding you what the heck you were thinking when you wrote it...and telling others what you were thinking when they try to join in. [03:08] <burek> I agree with you, but how to make the developers agree with you too? :) [03:08] <burek> their todo lists are scary [03:09] <ulatekh> I don't know. My 20-year career has been a nightmare, and lack of programmers commenting their code is the biggest reason. [03:10] <Compn> ulatekh : so document one file [03:11] <Compn> :P [03:11] <Compn> no ones stopping you from adding comments to code [03:11] <Compn> what i'm tired of is people who stop by our community for one day (or a week tops) and complain about the code, then never return [03:11] <Compn> its like, put up or shut up [03:11] <burek> Compn, i believe the idea he is standing behind is for developers to document their own code :) [03:11] <Compn> and the developers say no. [03:11] <ulatekh> Well, knowing my patches might actually get applied could convince me it's worth my time...and yes, burek, I'm talking about developers documenting their own code. [03:12] <ulatekh> @compn: and thus my career is a nightmare. [03:12] <ulatekh> I'll never understand the "no to documentation" mentality if I live to be 1000. [03:13] <ulatekh> And as far as me adding comments to ffmpeg's code...I'd have to understand it first. And the lack of any documentation is an insurmountable barrier. Plus, I have to wonder if anyone on the other side will even care if I do something like that. Like I said earlier, my usual experience with submitting fixes in the open-source world, is apathy. [03:14] <Compn> uh oh [03:14] <Compn> no offense [03:14] <Compn> but i am detecting open source troll [03:14] <Compn> my warning flags have gone up [03:15] <ulatekh> "open source troll"? [03:15] <Compn> yes [03:15] <ulatekh> Define? [03:15] <Compn> comparing open source world to commercial software world comes next [03:15] <ulatekh> I hate commercial software. I run it as little as possible. [03:16] <Compn> oh good, then i was wrong :) [03:16] <burek> wait Compn, the guy has got a point [03:16] <burek> no need to flame him [03:16] <Skyler_> the documentation could be improved in many places, patches are probably welcome [03:16] <Compn> patches are always welcome :) [03:16] <Compn> i got a better one [03:16] <ulatekh> Are you suggesting I'm somehow anti-open-source? Dude, Google my username...I'm like a rash all over the open-source world. Mostly I get ignored, but I'm there in the devel mailing lists. [03:16] <Compn> ulatekh : what code confuses you ? [03:16] <michaelni> ulatekh, I will apply the pkgconfig suffix patch, i just want to test it a bit more [03:17] <Skyler_> however video/audio coding has an extra catch that generally people are expected to understand the format before reading the code (unless it's a format where there's no documentation at all, like some proprietary thing), so often it won't document things that are 'obvious' to someone who knows how the format works [03:17] <Compn> ulatekh : i can almost guarentee someone will comment code that you say is hard to understand :) [03:17] <ulatekh> @michaelni: Awesome. All I really wanted was feedback. [03:17] <Skyler_> e.g. an h264 decoder won't explain the h264 spec [03:17] <Compn> h264 spec wont explain h264 :P [03:17] <Compn> ehe [03:17] <Skyler_> Pfffff [03:18] <ulatekh> Well, it'd help if e.g. the h264 decoder would explain how parts of the code relate to the spec...when I'm trying to track down a specific bug, I'd like some idea of what the code is doing, so that I don't necessarily have to be an expert in the codec to track the issue I'm dealing with. [03:18] <Skyler_> yeah, that would help [03:18] <Compn> Skyler_: i dont see many copyrights in h264* [03:18] <Compn> hmm i thought someone just added specs to one of the decoders [03:18] <Compn> but i dont know which one [03:19] <Compn> well , not 'just' but maybe 2 yrs ago [03:19] <burek> :) [03:19] <burek> see what i mean about free time :D [03:19] <burek> people loose the sense of time coding ffmpeg :))) [03:20] <ulatekh> Recently I submitted ticket #2190, while I was trying to get raw video packaged so that kdenlive would be able to import it. Bless cehoyos for knowing what to look for...I would have never been able to find what he did. [03:20] <Compn> i barely know what date is today :\ [03:20] <Compn> ulatekh : its a lot of code to memorize :) [03:21] <ulatekh> @compn: for an example of code that confuses me...MPEG 1/2 encode. One thing I've wanted in ffmpeg for like forever is support for soft telecine, i.e. giving it a 24000/1001 fps video and getting it soft-telecined to 30000/1001. And being able to transcode soft-telecined video w/o it getting mucked up. But digging through the source code, I have no idea what is doing what or why, so I get nowhere. [03:23] <Compn> ulatekh : ehe, ffmpeg -vf mp=telecine or =softskip [03:23] <ulatekh> mpeg2enc from the mjpegtools project does a GREAT job of converting analog video to mpeg1/2, but it doesn't do 2-pass encoding, so I'd like ffmpeg to transcode my soft-telecined video so that I can achieve my desired bitrate (i.e. first encode video with mpeg2enc at the maximum bitrate). [03:25] <ulatekh> If it finally got implemented, that's great...the last time I scoured the Net for how to do that with ffmpeg, what little info I could find seemed to suggest that most ffmpeg developers were non-U.S. and thus didn't care to spend time on soft-telecine. And last week or so, I was on the #ffmpeg IRC channel, discussing soft-telecine with michaelni, and he didn't seem to know it had been implemented... [03:26] <ulatekh> In any case, soft-telecine was an example of something I would have been willing to try to implement, if I could understand the source code at all. [03:27] <ulatekh> And hey, maybe this is a case of "coming home from a bad day at work and kicking the dog"...the programmers ruining my life right now are the ones I have to deal with through my job ;-) [03:28] <cone-874> ffmpeg.git 03Steven Boswell II 07master:6289a8296a89: build-sys: Fix pkgconfig files when ffmpeg is built with --build-suffix [03:28] <ulatekh> Hey, my faith is restored partially...nice :-) [03:29] <ulatekh> Tell you what...let me go get a beer & let's see if I stop whining so much LOL [03:29] <Compn> michaelni was the one who added mplayer filter support [03:29] <Compn> but maybe soft-telecine and telecine are different things that i'm confusing ... [03:30] <ulatekh> Based on what I'm reading in https://ffmpeg.org/trac/ffmpeg/ticket/1782 it seems ffmpeg _does_ support it now. So...hell yeah! [03:32] <Compn> yeah, michaelni works hard for ffmpeg :) [03:33] <ulatekh> My big forays into open-source programming tend to happen when I have time and energy, i.e. when I'm unemployed. While employed, all I have time/energy for is supplying bug fixes for the stuff I run into while trying to do what I want to do. [03:36] <ulatekh> y4mdenoise (i.e. http://mjpeg.cvs.sourceforge.net/viewvc/mjpeg/mjpeg_play/y4mdenoise/ ) happened during a nasty period of unemployment. [06:08] <burek> oh im in gmt+1 so it counts more [06:08] <burek> :) [06:16] <burek> ok it works now :) [06:16] <burek> michaelni, please let me know if something doesn't work as expected, so i can fix it [09:37] <creep> hi [09:37] <creep> i have an mkv file taht ffplay hangs in the middle [09:38] <creep> if a specific part is skipped over, then it will play ok again [10:00] <wm4> did ffmpeg stop extracting cover art images from .aac? [10:01] <wm4> (using ID3v2) [10:01] <wm4> it works with an older version, but not recent git [10:36] <wm4> cehoyos: what the fuck? [10:38] <wm4> I guess I'll post a useless paste of ffplay output on all tickets if it makes you happy [10:39] <wm4> even though it's completely useless, and any developer attempting to reproduce this problem wouldn't get anything out of it (other than wasting his time reading useless crap) [10:39] <cehoyos> wm4: Only paste ffplay output if the problems are not reproducible with ffmpeg, see http://ffmpeg.org/bugreports.html [10:39] <cehoyos> And please do not forget your newer tickets [10:39] <wm4> what's the point? [10:40] <cehoyos> I don't understand. [10:40] <wm4> I've provided all information needed to reproduce the problem [10:42] <cehoyos> I disagree: Either provide the failing ffmpeg command line together with complete, uncut console output (ffplay command line if the problem only affects or is only reproducible with ffplay) or provide minimal source code that allows to reproduce the problem if it is neither reproducible with ffmpeg nor ffplay [10:44] <wm4> it's an illusion to think that simple ffmpeg or ffplay usage with no interaction could show some of these problems [10:44] <wm4> or that people would actually bother writing complicated test programs [10:45] <cehoyos> If this is the case (this is very unusual over the total number of tickets - and issues - ever opened) then please take the time to explain why the problem is not rerproducible with ffmpeg (the application): Very often, this explanation was already sufficient to solve the problem for the reporter [10:52] <wm4> ffmpeg fails to inject an attached picture in the packet stream when seeking before the start of the file - in libav it works [10:53] <wm4> cehoyos: how would you report this issue? [10:54] <cehoyos> Apart from the fact that I don't understand it (and I suspect you don't mean ffmpeg but FFmpeg): What about source code that allows to reproduce this? (I don't think this was ever reported.) [10:54] <cehoyos> Does this correspond to one of the tickets you opened? Which one? [10:54] <wm4> the only source to reproduce this is my mplayer fork [10:55] <wm4> not yet [11:02] <nevcairiel> wouldnt it be better to announce the machines that have newly joined the group of failed/idle machines? [11:10] <divVerent> wm4: one idea to make a test case: [11:11] <divVerent> would it be viable to edit ffmpeg.c to make the issue visible? e.g. a debug print when getting the picture? [11:11] <divVerent> ffplay.c, actually [11:36] <cehoyos> bcoudurier, TimNich: I sent a patch that writes the fiel atom independently of the codec_id. Is there anyhting I miss? Ie, should the atom not be written for some codecs? [11:38] <wm4> cehoyos: is #2010 ok now [11:38] <cehoyos> still downloading... [12:04] <michaelni> burek, i think the duron box is too slow for a dumb 24h threshold [12:29] <ubitux> hey [12:30] <ubitux> please tell Paul i don't mind any change to showspectrum; i'll review history when i come back, if something is wrong i'll do what's necessary [12:32] <cone-454> ffmpeg.git 03Carl Eugen Hoyos 07master:b45a3e167f49: Map the interlaced flag of yuv4mpeg streams to AVCodecContext->field_order. [12:35] <wm4> so how can you set arbitrary avopts with the ffmpeg frontend? [12:35] <cehoyos> ubitux: If you have time, there are still two patches in ticket #2144 [12:36] <wm4> or rather, how do you know where they end up [12:45] <ubitux> cehoyos: i thought i "reviewed" them [12:47] <ubitux> http://ffmpeg.org/pipermail/ffmpeg-devel/2013-January/137682.html http://ffmpeg.org/pipermail/ffmpeg-devel/2013-January/137684.html http://ffmpeg.org/pipermail/ffmpeg-devel/2013-January/137686.html [12:47] <cehoyos> ubitux: Did you review this one? http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/157781/focus=157860 [12:48] <ubitux> oh last version, will look [12:49] <cehoyos> I did not apply 3/3 because I thought it might be better to apply 2/3 first [12:50] <ubitux> yep sure [13:19] <wm4> cehoyos: do you still have any tickets reported by me on your close-soon list? [13:20] <ubitux> http://superuser.com/questions/545329/automate-running-a-program-when-input… would be nice to script this with ebur128 filter [13:22] <wm4> oh right, ticket 2002 [13:22] <wm4> but how the hell do you demonstrate a bandwidth issue [13:22] <wm4> maybe there's a program that can measure network bandwidth usage of a single program by playing ld_preload tricks? [13:23] <cone-454> ffmpeg.git 03Diego Biurrun 07master:76d90125cdc0: vf_hqdn3d: x86: Add proper arch optimization initialization [13:23] <cone-454> ffmpeg.git 03Michael Niedermayer 07master:0d13a7b786b9: Merge remote-tracking branch 'qatar/master' [13:23] <cone-454> ffmpeg.git 03Michael Niedermayer 07master:d593f2b2416f: avfilter/x86/vf_hqdn3d_init: fix author attribution & project name [13:23] <cone-454> ffmpeg.git 03Michael Niedermayer 07master:6726fb653320: avfilter/vf_hqdn3d.h: restore author attribution, history and project name [14:04] <DEATH> Hi, I'm having some issues building FFmpeg 1.1.1 on MinGW [14:05] <DEATH> libavcodec/x86/dsputil_mmx.c: In function 'dsputil_init_mmx': [14:05] <DEATH> libavcodec/x86/dsputil_mmx.c:2309:14: error: 'gmc_mmx' undeclared (first use inthis function) [14:05] <DEATH> libavcodec/x86/dsputil_mmx.c:2309:14: note: each undeclared identifier is reported only once for each function it appears in [14:06] <DEATH> Configure line: [14:06] <DEATH> ./configure --disable-everything --enable-decoder=aac --enable-shared --disable-swscale --disable-network --disable-avdevice --disable-postproc --disable-avfilter --disable-avformat --disable-debug --enable-runtime-cpudetect --disable-swresample --disable-ffprobe --enable-decoder=mp3float --enable-decoder=vorbis [14:06] <DEATH> Goes away with --disable-mmx but that appears to also kill all CPU specific optimizations, configure shows SSE and such being all disabled after appending --disable-mmx [14:11] <michaelni> DEATH, cant reproduce this (linux) [14:23] <cehoyos> wm4: Regarding 2002: You don't have to explain why it makes sense not to request all streams. Just add the console output and explain how ffmpeg should know which stream you want to receive. [14:57] <cone-454> ffmpeg.git 03Nicolas George 07master:55910e18948d: lavd/alsa: simplify reordering functions definition. [15:30] <saste> wm4? [15:30] <wm4> saste: ? [15:30] <saste> wm4, ping on my lavf/http cookie patch [15:31] <saste> that should address the first issue [15:31] <wm4> saste: oh ok... I don't follow the ML [15:33] <wm4> saste: I think that's not really ideal... IMO it should always return with an error if an error happened, instead of silently ignoring it [15:33] <wm4> at least on OOM [15:33] <saste> wm4, that's unrelated [15:33] <wm4> but then I don't know what ffmpeg's policy is on OOM issues [15:34] <saste> yes and i don't like how errors are handled there [15:34] <saste> but at least it avoids the (null) [15:34] <wm4> hm looks like I git reset --hard'ed the patch I wrote for this [15:35] <cone-454> ffmpeg.git 03Stefano Sabatini 07master:6032a1c977cc: ffplay: extend doxy for audio_decode_frame() [15:35] <cone-454> ffmpeg.git 03Stefano Sabatini 07master:fd6b6efbd212: lavfi/crop: fix help message for the keep_aspect option [15:35] <cone-454> ffmpeg.git 03Stefano Sabatini 07master:6d9c21dc0e82: doc/indevs: add missing final dot in v4l2 option value description [15:35] <wm4> saste: IMO get_cookies should return whether there is an error (and if there is, return from http_connect too), and the check whether cookies is NULL should be a separate if block entirely [16:18] <cone-454> ffmpeg.git 03Marton Balint 07master:571ef42dd4eb: ffplay: dynamically allocate audio buffer [16:18] <cone-454> ffmpeg.git 03Marton Balint 07master:4fd6e5af1e33: ffplay: fix order of setting show_mode [16:18] <cone-454> ffmpeg.git 03Marton Balint 07master:5de3f724f152: ffplay: remember last window dimensions [16:18] <cone-454> ffmpeg.git 03Marton Balint 07master:c5eab4bb70cc: ffplay: move up pause functions [16:18] <cone-454> ffmpeg.git 03Marton Balint 07master:4ea7fbb2ec1a: ffplay: step to next frame if paused when seeking [16:19] <cone-454> ffmpeg.git 03Michael Niedermayer 07master:4be0b9109421: Merge remote-tracking branch 'cus/stable' [16:27] <cone-454> ffmpeg.git 03Michael Niedermayer 07master:cdc48860a8cb: h264: silence warning about array index being out of bounds [16:38] <saste> wow we're fast ;-) [18:51] <wm4> with ffmpeg I can't play concatenated ogg files, while I can with libav [18:54] <Daemon404> i've been finding an increasing number of ffmpeg-only bugs recently... [18:55] <wm4> the end is near [18:57] <cehoyos> wm4: Please provide a sample [18:57] <wm4> cehoyos: ffmpeg -i something.mp3 out1.ogg && ffmpeg -i something.mp3 out2.ogg && cat out*ogg > evil.ogg [18:58] <wm4> cehoyos: I'll test with ffmpeg/avconv later and create a proper ticket just to make sure it's not my fault [19:04] <cehoyos> wm4: Why do you think concatenating ogg files produces valid ogg files? (I don't know if it is valid.) [19:04] <michaelni> wm4 works here [19:05] <michaelni> even with changinhg samplerate & channels between the files [19:06] <wm4> cehoyos: apparently it's similar to how ogg streaming can work [19:06] <cehoyos> Daemon404: I find that impressive: Originally (at the time of the 0.7 and 0.8 release) there were around 50 regressions in the fork, perhaps 70. Later, there were a few hundred regressions, now, I can typically not even test new tickets with avconv because it often completely fails to process the given sample [19:06] <cehoyos> And concerning ogg: The reason I asked for a sample is that FFmpeg supports a number of ogg samples (and features) that simply do not work with avconv [19:07] <wm4> cehoyos: I tried it because I saw this: https://bugzilla.mozilla.org/show_bug.cgi?id=455165#c29 [19:09] <cehoyos> Yes, you are right: Chained ogg support is missing in avconv [19:09] <Daemon404> impressive as it may be [19:09] <Daemon404> it's true [19:10] Action: Daemon404 goes to see if the known bugs have been fixed and queue some to report later [19:10] <wm4> cehoyos: who's right? I said it works with libav but not ffmpeg [19:10] <wm4> cehoyos: though I still haven't tested with ffmpeg/avconv directly [19:11] <Daemon404> please provide full uncut output... [19:11] <Daemon404> /troll [19:12] <cehoyos> Yes, that is exactly what I mean! [19:12] <wm4> anyway, I still don't know if libavformat handles ogg streams correctly now... does it still add new tracks on track changes with some ogg streams? [19:21] <michaelni> wm4 it should only create new streams if theres no existing stream with matching codec [19:21] <cone-454> ffmpeg.git 03Nicolas George 07master:23686c72e5ae: doc: update filter_design.txt to API changes. [19:21] <michaelni> i tried vorbis in ogg and flac in ogg both work with chained oggs [19:22] <wm4> michaelni: and how does the application know it has to switch to the new stream, and that the old stream won't have new packets? [19:22] <michaelni> you have chained oggs that mix different codecs ? [19:22] <wm4> probably not [19:23] <wm4> well, the question is whether ogg internet radio streams do this [19:23] <wm4> but they probably don't [19:23] <wm4> what about vorbis extradata? [19:23] <michaelni> will be sent in stream on the point where old swicthes to new [19:24] <wm4> as part of the packet? [19:24] <michaelni> yep [19:25] <wm4> nice [20:02] <wm4> hm I guess I accidentally tested seeking [20:02] <wm4> which appears to behave slightly differently between ffmpeg and libav [20:03] <wm4> ffmpeg still prints a weird message, though: "[NULL @ 0x8ade2c0]Invalid packet" [20:03] <wm4> and reports a bogus length [20:06] <wm4> oh wait, the bogus file length is only with libav [20:08] <cone-454> ffmpeg.git 03Michael Niedermayer 07master:0e9b9a6748aa: flacdec: skip in stream header packets [21:28] <cone-454> ffmpeg.git 03Michael Niedermayer 07master:695af8eed642: h264: skip error concealment when SPS and slices are mismatching [22:37] <cone-454> ffmpeg.git 03Carl Eugen Hoyos 07master:a60530e3ee1d: Require at least three frames to autodetect loas. [00:00] --- Sun Feb 3 2013

1 0

ffmpeg.log.20130202
by burek 03 Feb '13

03 Feb '13

[00:24] <llogan> nilsge: using jack1 or jack2? [00:25] <nilsge> llogan: jack1 [00:26] <nilsge> ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -f jack -i default -ac 2 -c:v libx264 -preset fast -pix_fmt yuv420p -s 1280x960 -c:a libmp3lame -ab 128k -ar 44100 -threads 0 -f flv "rtmp://live.twitch.tv/app/live_**********************" [00:26] <nilsge> it makes no difference what a/v values I'm using. I went down to 320x240 and -ab 48k -ar 22050 [00:36] <teratorn> if im generating synthetic audio/video frames for testing, what timestamp-related invariants/constraints should I adhere to in order to keep the muxer happy interleaving the packets? [00:48] <nilsge> llogan: I tested it with jack2, it is the same. [00:56] <llogan> you can cut out the repeated messages [00:58] <llogan> are you using realtime scheduling? [00:58] <nilsge> yes [00:59] <nilsge> I am running a pro-audio system with recording chain and super soundcard etc. it is not my system itself [00:59] <nilsge> http://pastebin.com/xuZp8HMZ [01:00] <nilsge> also those xruns do not show up in jack itself (e.g. qjackctl display) [01:00] <nilsge> in this example there was no sound at all. just silence [01:00] <nilsge> the jack ports weren't even connected [01:00] <nilsge> system was idle, except ffmpeg [01:01] <nilsge> The web is full of cry for helps with xruns but no solution. [01:01] <nilsge> xruns in ffmpeg I mean [01:06] <llogan> is the output still ok? [01:07] <nilsge> llogan: absolutely not. You can't listen to it. [01:07] <nilsge> llogan: with the audio hiccpus there also comes a video speed issue. [01:07] <nilsge> I have read in the web that using audio alone with jack seems to work, and video alone also. It is the combination. [01:07] <nilsge> so it might be a sync issue, but I don't know much about these things [01:12] <llogan> i haven't used jack in years, but maybe i'll try to duplicate the issue since i'm going to be using hydrogen for a drumtrack over the weekend [01:14] <llogan> nilsge: maybe you can get an answer at ffmpeg-user mailing list [01:14] <llogan> or linux-audio-user [01:15] <nilsge> lau does know nothing. I am well connected there. It is a known issue with the answer "avoid doing it, don't do live screencasts and use recordmydesktop for the rest" [01:16] <nilsge> (as a sidenote: libav behaves exactly the same) [01:16] <llogan> i guess that's their way of saying "i don't know" [01:16] <llogan> if you think it is a bug you can always search/submit a bug report. [01:17] <nilsge> I don't know if it is a bug. It sounds like not knowing JACK. [01:44] <burek> nilsge [01:44] <nilsge> burek: yes [01:45] <burek> try this: ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -f jack -i default -c copy -f null - [01:45] <burek> does it still throw xruns? [01:45] <burek> octe, did you solve your issue [01:46] <nilsge> burek: no xruns [01:46] <nilsge> and my system is under heavy load right now, I am compiling [01:46] <burek> that's good [01:46] <nilsge> plus I am actually sending audio to the default input [01:46] <burek> that means your card can capture at that rate without problems [01:46] <nilsge> so, what does it mean? [01:47] <nilsge> yes, I thought so. [01:47] <nilsge> RME Hammerfall [01:47] <burek> now, we need to see if your bus/cpu is also good enough to sustain the load when both capturing and encoding [01:47] <burek> ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -f jack -i default output.mpg [01:47] <burek> or [01:48] <burek> ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -f jack -i default -c:v libx264 -crf 0 -c:a pcm_s16le output.flv [01:48] <nilsge> that generates xruns on mass. I'll abort the compilation first. [01:48] <burek> ok [01:48] <nilsge> still xruns [01:49] <nilsge> too many. [01:49] <burek> yes i see [01:49] <burek> the problem is (i guess) [01:49] <burek> that your audio card is sending the data at such rate that your machine can't quite respect [01:49] <nilsge> the second command does not work since it does not set a sample rate. I am on 48k which is not in flv [01:49] <burek> since it wastes time on encoding, it can't capture audio properly [01:50] <burek> what kind of cpu do you have [01:50] <burek> (try mp4 or mkv) [01:50] <JEEB> <nilsge> the second command does not work since it does not set a sample rate. I am on 48k which is not in flv <- actually it is [01:50] <JEEB> at least for AAC [01:50] <JEEB> page 71 of the current version of the FLV specification [01:50] <nilsge> yes, I have not read this [01:50] <burek> i wanted to avoid any kind of compression [01:50] <nilsge> [flv @ 0x27bd9c0] FLV does not support sample rate 48000, choose from (44100, 22050, 11025) [01:51] <nilsge> anyway [01:51] <JEEB> nilsge, you should file that as a bug [01:51] <JEEB> as it is one [01:51] <JEEB> if you used AAC [01:51] <burek> nilsge, which version of ffmpeg are you using [01:51] <JEEB> anyways, yeah -- no compression for testing at first is a good idea [01:51] <nilsge> burek: git from 15 minutes ago [01:51] <JEEB> possibly even using a separate wav file as output, otherwise mkv should be fine [01:52] <nilsge> burek: the latest stable in the archlinux repositories was build without jack [01:52] <nilsge> burek: Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz [01:52] <burek> oh ok [01:52] <burek> sorry [01:52] <nilsge> (00:59) ( nilsge) http://pastebin.com/xuZp8HMZ [01:52] <JEEB> I think libav's flvenc.c didn't have that limitation as I poked them about it lately, so I wonder if and why ffmpeg would still have the 44kHz limitation with flv + aac (if it still has such) [01:52] <burek> ok :) [01:53] <JEEB> since ffmpeg merges libav's stuff [01:53] <JEEB> (At least most of it) [01:53] Action: JEEB goes check ffmpeg's flvenc.c [01:53] <burek> oh, you probably have hit an alsa issue [01:54] <burek> try: ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -f alsa -i hw:0 output.mpg [01:54] <burek> it should still show those xruns [01:54] <JEEB> hmm... [01:54] <JEEB> ffmpeg's flvenc.c seems similar [01:54] <nilsge> that does not work since jack is active. well.. I could deactivate it for now.. one moment [01:54] <JEEB> (unsurprisingly) [01:54] <JEEB> so I guess he wasn't using aac [01:55] <JEEB> because it shouldn't go into that one switch statement in that case [01:56] <nilsge> burek: I can't start this. my hw:0 returns Input/output error. cannot set access type (Invalid argument) [01:56] <nilsge> hw:0,0 the same [01:57] <nilsge> which is indeed my main device [01:58] <burek> did you check with arecord -l [01:58] <nilsge> yes [01:58] <nilsge> and cat /proc/asound/cards [01:59] <nilsge> but I have a loopback device, that works [01:59] <tlp> now that ffmpeg supports encoding 24-bit flac, is there any way to tell it to go back to producing 16-bit? [01:59] <nilsge> produces xruns. not so many [02:00] <nilsge> burek: that looks stable enough so far [02:01] <JEEB> tlp, if this was raw audio I'd say "look at -formats | grep "pcm" and pick a similar 16 one instead of the 24 one, but since your output format will be flac (probably in some container, too) you'd have to call libavresample out in some other way :) [02:01] <JEEB> s/libavresmaple/libswresample/ [02:01] <nilsge> burek: after ctrl+c there were a lot buffer underflow i=0 bufi=235019 size=265650 packet too large, ignoring buffer limits to mux it [02:02] <tlp> hmm [02:02] Action: JEEB wonders if it's -precision [02:02] <JEEB> oh, for soxr only [02:03] <JEEB> oh found [02:03] <JEEB> http://ffmpeg.org/ffmpeg-filters.html#aconvert [02:03] <JEEB> this I'd guess? [02:03] <burek> hmh [02:04] <nilsge> burek: as additional information: all the web search showed me that people who used pulseaudio instead, or sometimes alsa directly, got away with it. [02:04] <nilsge> but pulse is not suitable for a realtime audio system so I would rather not use that. [02:04] <burek> nilsge i wonder if something like this would work: ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -f jack -i default -c copy -f SOME_FORMAT_I_DONT_QUITE_KNOW - | ffmpeg -f THAT_SAME_FORMAT -i - -c:v libx264 ... output.flv [02:05] <burek> now, that format could be something like mpegts or nut [02:05] <burek> or im not sure which one is the most suitable for that [02:05] <burek> but the idea is to separate capturing from encoding [02:06] <burek> to see if the encoder spends "too much" time in encoding process, making alsa have to wait with its data, which fails [02:06] <burek> also, as a quick hope, try putting audio input first [02:06] <burek> it worked for me sometimes [02:07] <burek> ffmpeg -f jack -i default -f x11grab -s 1920x1080 -r 60 -i :0.0 output.mpg [02:07] <nilsge> what format shall I use for the first command? [02:09] <nilsge> burek: the second one produces the usual xruns [02:13] <burek> nilsge, try ffmpeg -formats [02:13] <burek> try with -f nut [02:13] <nilsge> the 3 dots...? [02:14] <burek> -c:v libx264 -crf 0 -c:a pcm_s16le [02:15] <nilsge> I have to add -ar 44100 [02:15] <nilsge> xruns [02:15] <nilsge> usual amount [02:17] <nilsge> burek: what kind of computer is needed to encode and record at the same time? [02:18] <nilsge> now that it is mentioned, I remember with recordmydesktop I can't use on the fly encoding. I get sync problems, the video runs too fast, audio hiccups etc. [02:19] <burek> well it could be that alsa issue then [02:19] <burek> the point is (as i understood it that way) that alsa doesnt buffer enough data [02:20] <burek> and when encoder doesn't pick it up quickly enough, the issue happens [02:20] <burek> usually the data is being dropped, encoder then asks for the dropped data and alsa says please wait while i acquire more data for you [02:20] <burek> something like that [02:21] <burek> anyway, you could see if there is any chance to increase that buffer size [02:21] <burek> in alsa directly (or it was in ffmpeg's source code, not sure now) [02:21] <burek> or you might use your own kind of buffering [02:21] <nilsge> some posts said to increase it in ffmpegs source code [02:21] <nilsge> but with no difference, apparently [02:21] <burek> well, one thing that comes to my mind [02:22] <burek> since our very first test didn't produce any errors [02:22] <burek> we could send that output to a localhost/udp port [02:22] <burek> and with another ffmpeg pick it up [02:22] <nilsge> sounds interesting [02:23] <burek> that way udp proto handler in ffmpeg would buffer the data [02:23] <burek> and you might solve your issue [02:23] <nilsge> ok, I want to try that [02:23] <burek> of course, its a dummy workaround, but worth trying if you are in a rush to finish something [02:24] <nilsge> it has become more of a technical quest by now [02:24] <burek> ok, then try for the 1st ffmpeg: ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -f jack -i default -c copy -f SOME_FORMAT_I_DONT_QUITE_KNOW udp://localhost:1234 [02:24] <burek> and for the 2nd: ffmpeg -f THAT_SAME_FORMAT -i udp://@:1234 -c:v libx264 ... output.flv [02:25] <burek> usually mpegts format should work, but i cant guarantee [02:25] <nilsge> nut again? [02:25] <nilsge> ah [02:25] <nilsge> mpegts [02:26] <burek> try both [02:26] <burek> mpegts had some issues too recently (at least in my test cases) :) [02:26] <nilsge> the first runs. produces xruns already without the second even started [02:26] <burek> so if nut works better, then great [02:26] <burek> oh sweet [02:26] <nilsge> they stopped...? [02:26] <burek> wait [02:26] <burek> try this [02:27] <burek> ffmpeg -f x11grab -s 1920x1080 -r 60 -i :0.0 -f jack -i default -c copy output.mkv [02:27] <nilsge> looks stable after 3 initial xruns. The command before that went stable after a few seconds as well, it seems [02:27] <nilsge> but this is better [02:28] <nilsge> now the second... [02:28] <burek> hmh [02:28] <burek> so it appears to not be even fast enough for just capturing [02:28] <burek> how much ram memory do you have [02:28] <nilsge> (02:17) ( nilsge) burek: what kind of computer is needed to encode and record at the same time? [02:28] <nilsge> 4 GB [02:28] <nilsge> 64bit system [02:28] <burek> that obviously shouldn't be an issue [02:29] <nilsge> nothing of the ram is in use. minimal linux with i3 window manager, no bling [02:29] <burek> it depends on your encoding/capturing parameters [02:29] <burek> try for a test, to lower the frame size to 640x360 [02:29] <burek> (in the last command) [02:30] <nilsge> one moment, now that the xruns have stopped I want to try the current one with the 2nd command first. I had not started it yet [02:31] <burek> well, there's no point in it [02:31] <burek> since i've changed udp output to file output [02:31] <nilsge> that is strange. If I do nothing at all after starting the first (with nut format) it runs ok. Only when I start typing on my keyboard the xruns start as well [02:31] <nilsge> no I meant the udp version [02:31] <burek> yes, that's the same issue i had [02:32] <burek> somehow ffmpeg process doesn't get enough of either cpu time, bus access time or ram memory [02:32] <nilsge> I don't understand it. I started the first command alone again, just ctrl+c, up-arrow, enter [02:32] <nilsge> now zero xruns [02:32] <burek> and the worst thing is, the issue could never be successfuly reproduced at developers' machines.. :( [02:33] <burek> :) [02:33] <burek> you have to admit that it's not boring at least :D [02:33] <nilsge> Such things should not be possible with computers [02:33] <burek> it always throws something new at you :) [02:33] <nilsge> I have done it again to be sure, still everything ok [02:34] <nilsge> now, finally the 2nd command [02:34] <burek> what linux/kernel are you using? [02:34] <nilsge> archlinux 64bit, 3.7.5 [02:35] <burek> you just dont give me any reason to hold on to :) [02:35] <nilsge> there was a series of runs right now again but they stopped. so strange [02:35] <burek> probably those xruns happen when you move the mouse over your desktop or something cpu "intensive" :D [02:35] <nilsge> burek: I am no newbie with all that stuff. I know most details of my system, background processes etc. [02:36] <nilsge> burek: I tried that before I reported "zero xruns". Tried to generate some cpu noise [02:36] <nilsge> I also have an audio player running in the background, the whole time. seems to do nothing [02:37] <nilsge> they start/stop not at random. something I do with the programs/UI triggers it, but I don't know what yet [02:37] <burek> hmh.. did you try asking around in alsa community for some clues maybe? [02:37] <burek> just to make sure it's not (so far) a known issue? [02:37] <nilsge> I am afraid there is no alsa community [02:38] <burek> #alsa ? [02:38] <nilsge> from time to time some user comes to "us" in the #lad (linux audio developer) channel and asks for alsa help. We tell him #alsa and they usually report they send them to #lad in the first place [02:38] <burek> :D [02:38] <burek> that sounds great :) [02:39] <nilsge> Circular buffer overrun. To avoid, increase fifo_size URL option. To survive in such case, use overrun_nonfatal option [02:39] <nilsge> that is the second command [02:39] <burek> yes, that's the same issue i had [02:39] <burek> but that's for udp [02:39] <burek> it's not quite related to alsa issue [02:39] <burek> or it might be [02:39] <nilsge> oh wait, I made a typo [02:40] <nilsge> no, that was nothing. same result [02:41] <burek> :/ [02:42] <nilsge> it is strange [02:42] <burek> take a look at this issue: https://ffmpeg.org/trac/ffmpeg/ticket/615 [02:42] <burek> it's using usb webcam input instead of x11 [02:42] <burek> and alsa [02:43] <burek> and try to increment that buffer size [02:43] <burek> i hope it will work for you [02:43] <nilsge> ah yes, I have read that report earlier today [02:44] <nilsge> ok, lets see... [02:47] <nilsge> why again is this alsa related when using jack? [02:49] <nilsge> If I were using ffado (firewire soundcards)... [02:49] <burek> well, iiuc, jack is the wrapper around alsa lib, right? [02:50] <nilsge> not exactly. You don't need alsa. It runs with ffado, a null driver, over network without any soundcard information at all [02:50] <nilsge> (windows, bsd, osx) [02:51] <nilsge> but I'll try it with alsa buffer size first. If all goes wrong I can attempt to use alsa directly [02:56] <nilsge> burek: ok, compiled and installed [02:57] <nilsge> burek: 1st command is running, one xrun at the start, but that is quite normal for all programs here. [02:58] <nilsge> second does not run. I'm trying my original one. [02:59] <burek> if you compiled that increased buffer size thing, then try your own commands [02:59] <nilsge> doing it right now. I get xruns. too many I fear [02:59] <burek> :( [02:59] <nilsge> but that was with jack, not alsa directly [02:59] <burek> and with alsa? [03:00] <burek> btw, maybe there is the same thing for jack in ffmpeg [03:00] <burek> so if it works for alsa [03:00] <burek> you could try to fix the buffer size for jack too, i guess [03:01] <nilsge> burek: I had a quick look in the jack audio source file but I think it is not there. [03:01] <nilsge> which is quite correct. Jack buffer size is regulated by the daemon, and I have set it quite high to rule that out [03:02] <nilsge> well, maybe I could set it to a ridicoulous value... [03:03] <burek> :D [03:04] <nilsge> set it to 4096 which generates 186ms latency on 44100 Hz ( I set this to 44.1 on my hardware site as well so I can drop the resampling in ffmpeg) [03:05] <nilsge> nope [03:05] <nilsge> still bad [03:09] <burek> :S [03:09] <nilsge> burek: with alsa only... better still a few, but so far apart that you can listen to it [03:10] <nilsge> now I have to find out how to stream my sound to that alsa device :| [03:10] <burek> well, you just have to make sure if that really helped or the errors just turned out to become less frequent out of the blue [03:10] <burek> it's really difficult to debug such a thing [03:10] <burek> when you don't have a constant (expected) errors each time you run your tests [03:13] <nilsge> burek: so far they are less frequent. much less [03:14] <burek> sounds good :) [03:14] <burek> try double-increasing it again :) [03:14] <nilsge> I still have not found out how to use that device [03:24] <nilsge> burek: the video is working fine and no xruns [03:24] <nilsge> but no sound also :) but that should be only a settings problem [03:28] <burek> really? [03:28] <burek> what did you do to make it work? [03:28] <nilsge> the buffer size I think [03:29] <nilsge> the only problem is that I don't know how and which alsa device I have to specify to actually hear my sound [03:30] <nilsge> there is not much choice, though. I have 4 devices, but 3 of them can be ruled out (Webcam, Hdmi, Onboard Intel chip) [03:30] <nilsge> hw0 is my actual card, which is blocked by jack, and hw1 is my loopback device that I use to forward alsa-only sound to jack. [03:31] <nilsge> AND I thought the loopback works backwards as well so I can send jack back to an virtual alsa soundcard, e.g. to use with ffmpeg. But there is nothing coming out when I do that [03:31] <burek> oh i see [03:31] <burek> arecord -l [03:31] <burek> should help [03:31] <burek> also [03:31] <burek> arecord -L [03:32] <burek> probably ffmpeg -f alsa -i hw:X would be enough [03:32] <nilsge> -L is new to me [03:32] <nilsge> that is what I'm trying. hw:1 or hw:1,0 or hw:1,1 [03:32] <burek> but, sometimes you need to use -i hw:X,Y to specify a device Y on the card X [03:32] <nilsge> nothing [03:32] <burek> also, alsamixer might help [03:33] <nilsge> like I said, this sound stuff with alsa and jack is not new too me [03:33] <burek> just run it, press F6 and choose the card [03:33] <burek> then press <tab> to see only recording devices [03:33] <burek> ok [03:33] <burek> :) [03:33] <burek> im just saying it might be something trivial [03:33] <nilsge> I still hope that as well :) [03:34] <nilsge> alsamixer is of no use here. hw0 is RME, it has its own mixer. and hw1 is a virtual device, has no faders at all [03:36] <nilsge> maybe my loop_in device is wrong [03:37] <burek> well, im not sure [03:37] <burek> usually alsa is just enough to me [03:37] <burek> so i didnt experiment too much with other tools [03:38] <nilsge> the situation is a bit more complicated because JACK is running. But that should only be a slight disturbance [03:38] <nilsge> I think the main issue is solved [03:38] <nilsge> so thank you very much for that, burek! [03:38] <burek> well, im glad if we did solve something :) [03:38] <burek> :beer: :) [03:39] <nilsge> I have to check with a few basic tools if I'm getting any signal I think [03:49] <pietro10> Is it possible to have ffmpeg produce the same data on two output sources? I want to stream something over the internet at the same time as I save the stream to a file on my computer. Thanks. [04:31] <nilsge> burek: no luck. I think I give up for now [04:33] <burek> pietro10 check our wiki [04:34] <burek> nilsge :( [04:34] <nilsge> burek: wait! just now I tried arecord --v -fdat foo.wav for the first time and it shows me level [04:34] <burek> i told you, fun never ends with alsa :))) [04:35] <pietro10> ok thanks [04:36] <nilsge> aaargh, I don't understand it. The alsa device clearly receives data. why don't I hear it in ffmpeg [04:36] <pietro10> so just to check [04:36] <pietro10> ffmpeg <output1flags> <output1name> <output2flags> <output2name> ? [04:37] <nilsge> burek: http://de.twitch.tv/nilsge do you hear something? [04:37] <nilsge> or anybody [04:37] <pietro10> ok yes that is it thanks [04:38] <burek> im on a headless server atm without audio card :) [04:39] <burek> try using vlc [04:39] <burek> see if there are any spectrum changes [04:39] <burek> in visualizer [04:45] <nilsge> this is the final mystery: if I just do arecord -vv -fdat foo.wav I can see the VU Meter is going up, I connected my webcam to it, works. Total control. THere is sound coming to my alsa device [04:46] <nilsge> but if I use -D to specify the device is stays 0%. I don't know which device it captures if I don't give one. [04:55] <nilsge> according to this (arecord -vv -fdat foo.wav) http://pastebin.com/qW9NF1f1 it should be hw:Loopback,1 (or hw:1,1) but it is not. The device is silent when used with -D [04:55] <nilsge> burek: I joined #alsa to try it out myself. Maybe I never even get an answer :) [04:59] <burek> :) [04:59] <nilsge> can't I do the same in ffmpeg? Just omitting the device? [05:00] <burek> wait [05:00] <nilsge> somewhere it seems to exist, but not in the hw:x,y terminology [05:00] <burek> oh i see [05:00] <burek> try -f alsa -i dsnoop [05:01] <nilsge> that reads promising [05:01] <nilsge> cannot open audio device dnsoop (No such file or directory) [05:02] <burek> can you pastebin your arecord -L [05:03] <nilsge> http://pastebin.com/pLhc4i11 [05:03] <nilsge> U0x46d0x81d is my webcam [05:06] <burek> and you need to get the input from which device [05:06] <nilsge> http://pastebin.com/Ncbpt9ta [05:06] <nilsge> this is aplay -l and cat /proc/asound/cards [05:06] <nilsge> I am not sure, that is the point [05:06] <nilsge> I would guess it is 1,1 [05:06] <burek> your mic? [05:06] <burek> your cam? [05:06] <burek> what [05:07] <nilsge> it should be Loopback,1 [05:07] <burek> i mean logically from which device you would want to get your input :) [05:07] <nilsge> yes, Loopback. I am sending audio over jack to this [05:07] <burek> wait, what do you use to send your audio to jack [05:08] <nilsge> jack apps. music players etc. [05:08] <burek> what is the source of your audio :) [05:08] <nilsge> the point is, with arecord --vv -fdat foo.wav I actually see the signal when I connect anything to the jack end of the loopback device [05:08] <burek> your voice? your line in? your camera mic? pre-recorded audio file? what? [05:08] <nilsge> moc, the music player [05:08] <burek> oh [05:09] <burek> so "what I hear" [05:09] <burek> right? [05:09] <nilsge> kind of, yes. [05:09] <burek> well yes, you need dsnoop [05:09] <burek> so, try this: ffmpeg -f alsa -i dsnoop output.wav [05:09] <nilsge> cannot open audio device dsnoop (Device or resource busy) [05:09] <nilsge> dsnoop: Input/output error [05:09] <burek> wait let me check [05:11] <burek> i see.. can you try just for test to shut down jack [05:11] <burek> and see if the device gets ready [05:11] <nilsge> which device? [05:11] <burek> dsnoop [05:12] <nilsge> I can shut down jack, but I have no way to test it/send audio in [05:14] <burek> no need to [05:14] <burek> i just want to know if jack is blocking that device [05:15] <nilsge> no, it is not [05:15] <nilsge> arecord says when a device is blocked [05:15] <nilsge> hw:0 is completely blocked. that is my main card and used by Jack [05:15] <burek> well, the point is [05:15] <nilsge> hw:1 is the free loopback. The rest should be uninteresting [05:15] <burek> if the jack has blocked dsnoop for its own purpose [05:15] <burek> you then must use -f jack -i .. [05:16] <burek> cant workaround it [05:16] <nilsge> maybe, but I don't think that is the case. How does arecord receive the audio? It can't use blocked devices [05:16] <burek> good point [05:17] <nilsge> the key is that arecord without -D receives the audio, exactly as I want including the manual jack connection only to that loopback device, and when I use -D with anything it does not work anymore [05:17] <nilsge> and I need -D(evice) to find out what to give to ffmpeg [05:17] <nilsge> so most likely I missed something or there is more syntax than "hw:x(,y)" [05:20] <burek> maybe [05:20] <burek> -i hw:1,0,1 [05:21] <nilsge> I tried that before with an error message in arecord [05:21] <nilsge> ah, that was only a format error, not a device. [05:21] <nilsge> lets see ffmpeg... [05:23] <nilsge> all right! I got it in arecord at least. ffmpeg does not like the format as well [05:24] <burek> what format [05:24] <nilsge> float 32 bit is supported [05:24] <burek> pastebin :) [05:24] <nilsge> FLOAT_LE [05:25] <nilsge> just one line: cannot set sample format 0x10000 2 (Invalid argument) [05:29] <nilsge> burek: PCM 32-bit floating-point little-endian is needed. f32le [05:29] <burek> cant say without pastebin [05:29] <nilsge> pastebin of ffmpeg? [05:30] <burek> yes, show me the log to see what exactly ffmpeg is complaining about [05:30] <nilsge> http://pastebin.com/tG1UT9A7 [05:33] <burek> try using only ffmpeg -f alsa -i ... -f null - [05:33] <burek> remove all the video stuff [05:33] <nilsge> k [05:33] <burek> because when you set -r [05:33] <burek> it may apply to audio too [05:33] <burek> if you don't set -r:v and -r:a [05:34] <burek> maybe -r is not mutual option for a/v but you get the point [05:34] <burek> also, -ac 2 is after -i [05:34] <burek> which means that you are not requesting a 2 channel audio from your card [05:34] <nilsge> http://pastebin.com/Bc0AWjnD [05:35] <burek> but you actually are resampling it after you got the audio (in who knows what format) from your audio card [05:35] <burek> ok, now try [05:37] <burek> ffmpeg -f alsa -ac 2 -i hw:1,0,1 -f null - [05:37] <nilsge> the same [05:39] <nilsge> got it [05:39] <nilsge> burek: ffmpeg -f alsa -acodec pcm_f32le -ac 2 -i hw:1,0,1 -f null - [05:39] <burek> yes [05:39] <burek> i was looking to see what is the proper string :) [05:39] <nilsge> http://ffmpeg.org/trac/ffmpeg/wiki/audio%20types here [05:40] <burek> yes, the same you can get with ffmpeg -sample_fmts i think [05:40] <burek> or -formats [05:40] <burek> anyway, try joining all that now [05:41] <burek> see if it works, and if there are xruns again any more [05:42] <nilsge> http://de.twitch.tv/nilsge [05:42] <nilsge> burek: ! [05:43] <burek> it works ? :) [05:43] <nilsge> ah yes, you have no sound [05:43] <nilsge> it works [05:44] <nilsge> it seems the sound is 10s or so behind [05:45] <burek> buffering i guess [05:45] <burek> but is there any xrun? [05:45] <nilsge> but at least something is coming through [05:45] <nilsge> no, not one [05:45] <burek> oh great [05:45] <burek> it would be lovely if you could just check with the reverted version [05:45] <burek> if that buffer size was really an issue [05:45] <burek> so we can close it once and for good [05:45] <nilsge> the alsa buffer size should be a runtime value [05:45] <burek> i agree [05:46] <nilsge> ok, I'm installing the release version of ffmpeg again [05:46] <burek> thank you [05:46] <burek> im asking that because it is not very easy to reproduce the issue... [05:46] <nilsge> I can do that now since I don't need jack for that anymore [05:46] <nilsge> I found it rather easy. In 999 of 1 cases it did not work :| [05:48] <burek> yes, but you know Murphy... [05:48] <burek> if there is a chance it wont be present at developers' machines, it won't be present :) [05:49] <nilsge> there were a few xruns and the video/audio was not continous anymore [05:49] <nilsge> but not as bad as with jack [05:50] <nilsge> but the audio was still 15 seconds behind my host (and the video 4-5 seconds) [05:50] <nilsge> so +10s for audio [05:51] <burek> so the buffer might not be the only cause for that issue.. :s [05:52] <nilsge> I have to play around with the values a bit to see if that has any influence [05:52] <nilsge> at least I know that my jack bridge only adds ~20ms to that latency [05:53] <nilsge> for example the audio stream was uncompressed right now [05:55] <nilsge> burek: ah! with mp3 and 128k it suddenly gets down to ~5 seconds as well [05:55] <burek> geez [05:55] <nilsge> I have to see how stable it is because now the xruns become audible again [05:55] <nilsge> maybe revert to the buffer version of ffmpeg again [05:55] <burek> :S [05:56] <nilsge> since the delay was the same anyway [05:56] <nilsge> maybe I get no xruns AND proper sync [05:56] <burek> well play around and see what you get.. i lost a lot of time trying to resolve that issue [05:56] <nilsge> yes, thank you very much [05:57] <nilsge> I didn't expect it to be this long [05:57] <burek> :beer: :) [05:57] <burek> you see, it's never boring here :) [05:59] <nilsge> burek: and at last I can confirm that the sync is still as good as before (I don't know how much that is if I stream games with real action sfx sync) and no xruns through the buffersize [05:59] <burek> that's why games should dup their display to some udp port or something [05:59] <burek> so that we don't have to grab the screen [06:00] <nilsge> you still need audiosync then [06:00] <nilsge> the graphiccard and soundcard should internaly be able to generate a video stream [06:00] <nilsge> well, actually that is what a hardware recorder does :) [06:00] <burek> :) [06:01] <nilsge> that must be the most reliable solution. Just dvi through and analog audio through [06:02] <nilsge> if you know the youtube channel Angry Video Nerd this is what he does, playing retro games. Never emulator and screen captures, he just records the output of the consoles directly to a DVD recorder. May sound a bit inconvenient and ancient but it is rock solid [06:03] <nilsge> burek: good night. I should have been in bed 7 hours ago :| [06:03] <nilsge> but a good solution is worth 7 hours of sleep [06:04] <burek> :) [06:04] <burek> good night and take care :) [09:15] <Crosma> Yo. I'm just doing -vcodec copy -acodec copy on an mov to mp4 and it says, "multiple edit list entries, a/v desync might occur, patch welcome" when I try to play the mp4. The mkv only has one audio and one video stream. [10:54] <kode54> http://pastie.org/6028534 [10:54] <kode54> that's the readme for the source package that is available for building ffmpeg 1.0.1 for foobar2000 [10:55] <kode54> is there something about that process which would make it inherently incompatible with any other build of same version numbered av*.dll [10:55] <kode54> avcodec-54.dll etc [10:55] <kode54> same numbers [10:55] <kode54> it's not so much a problem with users building ffmpeg to replace it, which nobody has reported attempting [10:56] <kode54> so much as some random piece of crap Windows software appearing which forcibly loads itself into every process in the system, and auto loads its own copies of the same named but ABI incompatible ffmpeg [10:56] <kode54> which I know is not standard ffmpeg practice [10:56] <kode54> but apparently it's not compatible with some other stock builds of av*.dll either [11:55] <DEATH> Hi [11:56] <DEATH> I'm trying to build FFmpeg DLLs with different file names (as specified in http://www.ffmpeg.org/legal.html - add own suffix to the file names to prevent a clash with certain other software using FFmpeg). [11:56] <DEATH> Is there a ./configure option to do that? I cannot just rename the DLLs because that will break references between avcodec/avutil and so on [11:57] <DEATH> Must have other DLLs reference each DLL by the new name [11:57] <DEATH> In fact avcodec and avutil are all I care about [12:17] <dentex> HI everyone! [12:18] <dentex> I have a question about crosscompiling FFmpeg for Android. [12:21] <dentex> it works with a standard set of flags. But when I add '--enable-libmp3lame' don't [12:23] <dentex> I have also compiled liblame for android, but I don't know where to put the libs for the ffmpeg compiler to find them; or what to change to point there. [12:24] <dentex> I have the error 'libmp3lame not found' [12:34] <klaxa> dentex: did you add -lmp3lame -L/path/to/libmp3lame to ./configure? [12:35] <dentex> no... let me try [12:35] <dentex> thanks [12:36] <dentex> just one thing: [12:36] <dentex> from another project to build lame for android, I have now the structure [12:36] <dentex> ./libs/armeabi/liblame.so [12:37] <dentex> do you know if this is the missing library? [12:45] <nilsge> hello. what does the parameter -bufsize do? [12:53] <dentex> -lmp3lame is unknown... [12:54] <Nano> hi, i get the following error trying to use --enable-libmp3lame [12:54] <Nano> ERROR: libmp3lame >= 3.98.3 not found [12:54] <Nano> this is my configure call: [12:54] <Nano> ./configure --arch=x86 --target-os=mingw32 --disable-asm --disable-doc --disable-ffplay --disable-ffserver --disable-ffprobe --disable-everything --enable-libmp3lame --extra-cflags=-I/usr/local/include --extra-ldflags=-L/usr/local/lib [12:55] <Nano> lame/lame.h exist in /usr/local/include. why does the configure script fail ? [12:56] <klaxa> dentex: ah probably is just -llame [12:57] <klaxa> Nano: are you sure the version is >= 3.98.3? [12:57] <dentex> klaxa: ok. [12:57] <dentex> trying [12:59] <Nano> klaxa: yes it's 3.99.5 [13:02] <klaxa> it also exists as liblame.so in /usr/local/lib then i assume... hmm that should actually work though i think... i'm not too sure though [13:02] <Nano> where is the configure script command check_lib defined ? i'd like to check where it looks for the sources... [13:03] <Nano> or is there a kind of verbose mode ? [13:03] <klaxa> you can check config.log i guess [13:03] <klaxa> not sure whether or not that will help a lot though :X [13:04] <dentex> klaxa: with "did you add -lmp3lame -L/path/to/libmp3lame to ./configure?" you mean an flag when calling ./configure or somewhere inside the file? [13:04] <dentex> *as [13:04] <klaxa> flag when calling ./configure [13:04] <dentex> ok both unknown [13:05] <klaxa> :S [13:05] <dentex> [OT]are those from Mplayer?[/OT] [13:05] <klaxa> i'm not too familiar with compiling ffmpeg although i compile it myself every now and then, but cross-compiling is something different again [13:05] <Nano> the config.log says that it found the lame.h file but there seems to be a reference to the original lame configure location, which doesn't exist any longer.... [13:07] <dentex> klaxa: I know, thanks. And I'm not an expert. [13:07] <Nano> dentex: what you're tryin todo ? [13:07] <dentex> HI: same as you. But crosscompiling for android [13:07] <Nano> :-) [13:08] <Nano> and what's youre problem ? [13:08] <dentex> I cannot make the build script find the lame libs [13:08] <dentex> 5sec [13:08] <dentex> ... [13:08] <Nano> did you configure/make/make install lame ? [13:08] <dentex> it works with a standard set of flags. But when I add '--enable-libmp3lame' don't [13:08] <dentex> <dentex> I have also compiled liblame for android, but I don't know where to put the libs for the ffmpeg compiler to find them; or what to change to point there. [13:09] <dentex> it installs for the host system, but useless for the android build. Anyway I tried [13:09] <dentex> libs not found [13:09] <dentex> same error as for you [13:10] <dentex> do you have any hint? [13:10] <Nano> the make install copiesthe libs to /usr/local/lib [13:10] <dentex> right [13:10] <Nano> see if you have libmp3lame.a in there [13:11] <Nano> for me copying the lame/lame.h failed [13:11] <dentex> 1) no [13:11] <dentex> 2) same [13:11] <Nano> so i created a folder lame in /usr/local/include and copied the lame.h in it. [13:13] <Nano> and then i added the --extra-cflags=-I/usr/local/include to the "./configure --enable-libmp3lame" call [13:13] <Nano> cause the configure script is looking for lame/lame.h file [13:13] <dentex> Nano: did you try adding '--extra-ldflags=-L/usr/local' ? [13:14] <Nano> i've added --extra-ldflags=-L/usr/local/lib [13:16] <dentex> yes, with /lib [13:18] <Nano> ok, the config.log seems to be helpful now. [13:18] <Nano> there i get: error: undefined reference to '_fpgetmask' [13:18] <Nano> /usr/local/lib/libmp3lame.a: error: undefined reference to '_fpsetmask' [13:18] <Nano> collect2: ld returned 1 exit status [13:18] <Nano> ERROR: libmp3lame >= 3.98.3 not found [13:19] <Nano> so both files (header and lib) has been found. but theres a build error in lame. [13:20] <Nano> dentex: check your config.log [13:21] <Nano> if you need help put the output in pastebin [13:21] <dentex> Nano: ok thanks [13:35] <blight> hi [13:36] <blight> can someone tell me what kind of subtitle support ffmpeg has and how to use the API? [13:37] <nilsge> yesterday I searched and tried for hours how to get proper sound for live streaming (together with burek). It is now stable and the quality is good, but I still have a 4 second delay. Video is first, audio is behind. I tried reducing the quality of the audio, but to no audible effect. I searched the web now for an hour again, without luck. So: [13:37] <nilsge> do you know any parameters or tricks to make sure a+v are in sync? [13:38] <nilsge> my current ffmpeg commandline is: ffmpeg -f x11grab -s 640x480 -r 30 -i :1.0 -f alsa -ac 2 -acodec pcm_f32le -i hw:Loopback,0,1 -c:v libx264 -preset fast -pix_fmt yuv420p -s 640x480 -c:a libfaac -ab 128k -threads 0 -f flv "rtmp://live.twitch.tv/app/*********************" & [13:40] <dentex> Nano: http://pastebin.com/K0hfymDF [13:47] <nilsge> additional info to the sync problem: if I record instead of streaming everything looks fine. So I guess it isn't really my fault? [13:48] <rascal999> i have nice --20 ffmpeg -y -f alsa -ac 2 -ar 48000 -i hw:0,0 -f x11grab -r 30 -s 960x540 -i :0.0 -acodec pcm_s16le output.wav -an -vcodec libx264 -preset ultrafast -crf 0 output.mp4 -- which generates and audio and video output, is it possible to create one file with audio and video in it? [13:53] <XATRIX> Hi guys, is it possible to say ffmpeg to encode my mp3 with fraunhofer instead of lame ? [13:57] <JEEB> XATRIX, I don't think fraunhofer's mp3 encoder is available as open source [13:57] <JEEB> unlike their AAC encoder [13:57] <XATRIX> Currently i'm using such a string to catenate multiple files into the one big mp3 file [13:58] <XATRIX> /usr/bin/ffmpeg -i "concat:temp790-1.mp3|temp790-2.mp3|...." -acodec copy output.mp3 [13:58] <XATRIX> But i have a strange issue [13:58] <Nano> dentex: this is the problem i've mentioned [13:58] <XATRIX> Sometimes the output mp3 is not readable by the JS players via web [13:59] <XATRIX> Or it doesn't play the whole mp3 ifles [13:59] <XATRIX> Or some mp3 files missing inside the big one [13:59] <Nano> dentex: call './configure --enable-static --disable-shared --disable-rpath --disable-frontend --disable-gtktest' in lame-3.xx.x folder [14:00] <Nano> dentex: then 'make', then 'make install' [14:00] <Nano> dentex: then copy the lame-3.xx.x/include/lame.h to /usr/local/include/lame [14:01] <JEEB> hmm, shouldn't it copy the header automagically/ [14:01] <JEEB> ? [14:01] <JEEB> during 'make install' [14:11] <XATRIX> how can i list available codecs that i can use to encode my audio ? [14:12] <dentex> Nano: Ok I'm trying [14:12] <dentex> thanks [14:13] <JEEB> XATRIX, I'm not sure if there's a way to only output the audio ones, but ffmpeg -codecs lists all codecs in that ffmpeg [14:13] <XATRIX> yea, thanks [14:13] <dentex> Nano:I'm just not sure that using a toolchain ./configure will look there [14:13] <JEEB> I guess you could use grep or something after -codecs [14:14] <JEEB> because third symbol would be 'A' [14:14] <JEEB> for audio [14:17] <dentex> NAno [14:18] <dentex> Nano: I can't 'make' and then 'make install' in the lame-3.xx.x folder because I already have the libs built for android [14:18] <dentex> Nano: I don't need the libs built for the host system in /usr... [14:18] <dentex> sorry [14:19] <dentex> I just don't know how to make the build script know where they are. [14:20] <makara> any idea why this fails: avconv -i Track\ 1.m4a -acodec copy -f aac track1.aac [14:21] <dentex> makara: did you tried 'avconv -i Track\ 1.m4a -vn -acodec copy track1.aac' ? [14:21] <JEEB> does it tell you that you need a bit stream filter? [14:21] <JEEB> to ADTS [14:21] <dentex> makara: should work [14:22] <makara> dentex, does, but I need to build a script to go through multiple directories, so I want to leave out the output filename and force format type [14:22] <makara> says: Requested output format 'aac' is not a suitable output format [14:23] <makara> but 'ffmpeg -formats' includes aac [14:25] <dentex> makara: is Track\ 1.m4a a video file? [14:26] <makara> no, it works fine with the line you gave me [14:26] <makara> but why it says aac not acceptable format? [14:26] <dentex> :S [14:27] <dentex> but you can leave out '-vn' [14:28] <dentex> makara so you always need .aac output format? from different input types? [14:28] <dentex> all audio input? [14:32] <makara> i'm trying to convert all my m4a's into straight aac's so that I can distinguish between the alac's I'm getting, which also - gag me with a spoon - come as m4a's. [14:34] <relaxed> why don't you ffprobe to see instead of demuxing everything? [14:34] <relaxed> You're probably losing the metadata with that method. [14:37] <dentex> then why don't you simply do it with '-i track.m4a track1.aac' [14:38] <dentex> it will force .aac from the extension [14:48] <makara> no re-encoding, and as relaxed pointed out I'd like to keep metadata [14:51] <makara> relaxed, metadata is preserved conversion [14:58] <XATRIX> guys, can you support me a bit ? I'm totally in a trouble. Can you tell me was the ecoding OK ? Or there was problem ecnountered ? http://fpaste.org/f1Bv/ [14:58] <makara> ok, I see now. can demux but not mux aac [15:01] <makara> raw aac [15:06] <XATRIX> ? [15:07] <Fjorgynn> ? [15:08] <XATRIX> http://fpaste.org/f1Bv - is it ok ? Oh some troubles detected ? [15:09] <XATRIX> s/Oh/Or [15:30] <XATRIX> Guys, any idea how to control "pitch" while encoding ? [15:30] <XATRIX> The trouble is i have a few files to concat [15:30] <XATRIX> It has mixed, mp3 en/ru voice [15:31] <XATRIX> And, after i finally get result mp3, it has en voice ok, but the ru voice is changed, it become like a robo-voice [15:31] <XATRIX> a lower pitch rate [15:33] <XATRIX> Just for example this is the initial ru voice, http://devel.rf.ua/1359732328_118_1.mp3 . This is the file catenated with en.mp3 + ru.mp3 + en.mp3 http://devel.rf.ua/output1.mp3 [15:33] <XATRIX> I do encoding with #ffmpeg -i "concat:ru1359813352_0.mp3|1359732328_118_1.mp3|1359732328_118_2.mp3|ru1359813352_0.mp3" -acodec libmp3lame /mnt/www/rf.ua/output1.mp3 [15:34] <XATRIX> I'm totally screwed up with it :( [15:34] <XATRIX> I have no idea what's the trouble it has while encoding [15:35] <klaxa> shouldn't you use -f for concat? [15:36] <Keshl> Different bitrates, oÉo? Dunno, not an expert here. [15:41] <relaxed> XATRIX: -acodec copy [15:47] <XATRIX> relaxed: the same crap it does [15:47] <XATRIX> Keshl: the bitrates are the same [18:35] <pietro10> I have ffmpeg from https://launchpad.net/~jon-severinsson/+archive/ffmpeg with all the -extra packages and it's complaining that there is no vcodec libx264 when I have libx264 installed on mys ystem. What am I doing wrong? Thanks. [18:38] <pietro10> er unknown decoder libx264 [18:39] <pietro10> http://pastie.org/6030257 here is what I am trying to do [18:39] <pietro10> following the -c copy instructions http://ffmpeg.org/trac/ffmpeg/wiki/Creating multiple outputs [18:46] <pietro10> if I take out the decoder from the second part of the pipeline (just have -f flv), ffmpeg complains with [18:46] <pietro10> pipe:: could not find codec parameters [18:55] <sgfgdf> hello, guys! i have a video with the following audio properties: "Audio: ac3, 48000 Hz, 5.1(side), fltp, 384 kb/s (default)". now i want to convert it to stereo AAC audio. i use this arguments -- "-c:a libfaac -ar 48000 -ab 384k -ac 2" and the output has the following audio properties: "Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 165 kb/s". so the bitrate is not the same. why is this happening? [18:56] <pietro10> am I going to hae to do a parallel encoding? [19:38] <pietro10> removing -f flv and just leaving -i - and -c copy gives me "pipe:: Invalid data found when processing input [19:38] <pietro10> " [19:45] <pietro10> am I going to have to repeat the options? [20:21] <Youka> What's the common way to get the frame number from decoded AVFrame? [20:28] <Mavrik> Youka: what do you mean by frame number? [20:29] <Youka> The frame index. [20:30] <Youka> I want it from AVFrame, not counting manually be decoding. [20:30] <Youka> *by [20:32] <Mavrik> Youka: you have frame_number field on your decoder context [20:32] <Mavrik> there are also picture_number fields on AVFrame itself, but I don't see them being set on decoding [20:37] <Youka> frame_number of codec is what's already decoded, so av_seek_frame isn't usable. [20:39] <Mavrik> av_seek_frame takes timestamp not frame number. [20:44] <Youka> thx, i will use coded_picture_number, this seems to solve my problem. frames are not always aligned, so i have to look forward sometimes, but when i have the number of frames i wanted, i can stop [20:45] <Mavrik> um [20:45] <Mavrik> you did hear me that av_seek_frame doesn't take the frame count, but timestamp right? [20:45] <Youka> right [20:58] <mantas322> Hi guys [20:59] <Mavrik> hey [20:59] <mantas322> I'm trying to convert from a .MOV video type to a avi or mpg [20:59] <mantas322> however when i do I lose colors [20:59] <mantas322> twindows says the "Type" is QuickTime Movie [21:00] <Mavrik> mantas322: can you please paste your command line and full output of then command to pastebin? [21:00] <Mavrik> alot easier to help you that way? :) [21:00] <Mavrik> also, what is your goal? where are you trying to play the output file? [21:00] <mantas322> i've tried all sorts of different combinations [21:00] <mantas322> my goal is I want tthe same quality [21:00] <mantas322> and dont want to lose any colors [21:01] <mantas322> ffmpeg -i C:\ffmpeg\bin\1.mov C:\ffmpeg\bin\OUT1.avi [21:01] <mantas322> -i C:\ffmpeg\bin\1.mov -f avi -vcodec mpeg4 -qscale 4 -g 300 -vtag DIVX -acodec libmp3lame -ab 192k -ar 44100 -ac 2 -y output.avi [21:02] <mantas322> I would assume that a simple "ffmpeg -i C:\ffmpeg\bin\1.mov -qscale 0 C:\ffmpeg\bin\OUT1.avi" should get me the same type of video [21:03] <mantas322> anyways I think im being too picky [21:03] <mantas322> its really not that big of a deal. [21:04] <LithosLaptop> mantas322: does this work? ffmpeg -i C:\ffmpeg\bin\1.mov -f avi -vcodec copy -acodec copy -y C:\ffmpeg\bin\OUT1.avi [21:04] <mantas322> lemme try [21:04] <Mavrik> mantas322: .avi can contain a lot of different video and audio formats [21:05] <Mavrik> if you don't pass them ffmpeg will choose a default [21:05] <Mavrik> if you do a copy you'll get another one [21:05] <mantas322> LithosLaptop, no I lose video [21:05] <mantas322> sound only [21:05] <Mavrik> that's why I asked you "What are you trying to achieve?" [21:05] <LithosLaptop> oh ok [21:05] <Mavrik> so we know which format you're trying to create [21:05] <Mavrik> also you still didn't paste full command output so we still don't know what your input file is. [21:05] <mantas322> it doesnt have to be avi [21:05] <mantas322> it could be mpg mpeg [21:05] <Mavrik> "avi" is a format. [21:06] <Mavrik> not video codec, not audio codec [21:06] <Mavrik> what codecs do you need on the output? [21:06] <mantas322> 1.MOV QuickTime Movie from a s100 cannon camera [21:06] <Mavrik> what device are you trying to play the output on? [21:06] <mantas322> whatever codecs achieve the original video quality [21:06] <mantas322> windows movie maker [21:07] <mantas322> I'm trying to make a cute kitten video for youtube [21:07] <Mavrik> ah I see [21:07] <LithosLaptop> oh do you want to be able to use your video in windows movie maker? [21:07] <mantas322> yes. [21:07] <mantas322> but it doesnt except .mov [21:08] <Mavrik> mhm [21:08] <Mavrik> mantas322: can you please do a "ffprobe" on your .mov file and paste output to a pastebin? [21:09] <mantas322> okay [21:09] <Mavrik> it's easier to tell you what to do when we know what resolutoin, quality and pixel format your video is in ;) [21:09] <mantas322> http://imgur.com/Q2qaTLq [21:10] <LithosLaptop> h.264 [21:10] <LithosLaptop> hmm [21:11] <Mavrik> yeah, not supported in movie maker [21:11] <mantas322> is there a free alternative to windows movie maker? [21:11] <Mavrik> mantas322: ffmpeg -i "1.MOV" -vcodec mpeg2video -vb 20000k -acodec mp2 -ab 128k output.mp2 [21:12] <Mavrik> should work [21:12] <Mavrik> without significant loss of quality [21:12] <Mavrik> if I see correctly movie maker will take mp2 files [21:13] <mantas322> 200mb to 700kb [21:13] <mantas322> sound only [21:13] <mantas322> didnt work, but thanks anyways. [21:13] <LithosLaptop> or maybe uncompressed [21:14] <LithosLaptop> what -vcodec is uncompressed? [21:14] <Mavrik> LithosLaptop: none recognised by movie maker :\ [21:14] <Mavrik> plain yuv output [21:14] <Mavrik> mantas322: what? [21:14] <mantas322> your command didnt work, it makes a 700kb mp2 file [21:15] <mantas322> form a 200mb mov file [21:15] <mantas322> "ffmpeg -i "1.MOV" -vcodec mpeg2video -vb 20000k -acodec mp2 -ab 128k output.mp2" [21:16] <LithosLaptop> somebody had the same problem: http://forum.doom9.org/archive/index.php/t-152345.html [21:16] <LithosLaptop> looks like windows movie maker is a bit picky [21:17] <mantas322> oh well [21:17] <Mavrik> mantas322: a 20Mbit/s video can't be 700kb -_- [21:18] <mantas322> no i was discribing what happened when i ran your command [21:18] <mantas322> input file was 200mb and it created an mp2 output file with sound only 700kb [21:57] <sgfgdf> guys, is it possible to use QuickTime AAC codec to encode audio with ffmpeg? [22:00] <pietro10> is it possible to have alsa record the speaker output or the sound from a specific program, not my mic? [22:00] <pietro10> right now I have -f alsa -ac 2 -i pulse \ [22:08] <LithosLaptop> sgfgdf: don't think so. the best AAC output you would be able to get from ffmpeg is with fdkaac [22:15] <sgfgdf> LithosLaptop, i don't know if my message got delivered? i got disconnected. [22:15] <LithosLaptop> didn't receive any messages [22:16] <JEEB> fdk-aac with -afterburner 1 is the best you get with ffmpeg [22:16] <JEEB> which should actually be pretty good [22:16] <JEEB> regarding AAC encoder [22:16] <JEEB> *encoding [22:16] <sgfgdf> LithosLaptop, is the fdkaac and libfaac the same thing? [22:17] <LithosLaptop> nope [22:17] <JEEB> faac is quite worse [22:17] <LithosLaptop> libfaac is old and inferior quality encoder [22:18] <JEEB> it's one of the reference code based encoders, not to mention it was improperly licensed (the reference code wasn't GPL-compatible although the creator tried to pass faac as GPL IIRC) [22:18] <JEEB> the other reference code encoder, vo-aacenc is even worse than the internal libavcodec aac encoder, which is quite hard to achieve to be honest :D [22:19] <sgfgdf> so if i install fdk-aac how should i tell ffmpeg to use it as audio ancoder? -c:a libfdk-aac? [22:19] <JEEB> libfdk_aacenc [22:19] <JEEB> you have to enable it naturally in ffmpeg as well [22:19] <JEEB> when compiling it [22:20] <JEEB> compile fdk-aac -> compile ffmpeg with --enable-libfdk_aacenc (I think it was like this?) -> see it with ffmpeg -codecs |grep "aac" [22:20] <sgfgdf> JEEB, where can i check the references like fdk-aac -> libfdk_aacenc and other libraries matching? [22:22] <JEEB> ./configure --help [22:22] <JEEB> see the External library support: aprt [22:22] <JEEB> *part [22:28] <sgfgdf> JEEB, does people encode their video using ffmpeg and then use external audio encoder such as QuickTime to encode the audio if they want the QuickTime AAC audio? is this popular practice? [22:29] <Mavrik> nah, too much work [22:29] <JEEB> with ffmpeg usage not really, but I've seen that kind of usage with people who use avisynth + x264 and so forth more manually for encoding. Might have been more popular before, but now with fdk-aac I think most people are OK with that quality. [22:29] <JEEB> To be honest, most people were OK with faac, which wasn't that great to begin with, so :D [22:29] <JEEB> fdk-aac is fraunhofer's [22:30] <JEEB> and with -afterburner 1 it should be pretty damn good :3 [22:30] <sgfgdf> JEEB, what does this option do? [22:30] <JEEB> enables fdk-aac's mode with the same name [22:31] <JEEB> seems to enable slower algorithms for better quality [22:34] <sgfgdf> JEEB, where can we put the Nero AAC and the default aac which comes with ffmpeg (which i don't know how to name it)? so far -- 1. QuickTime AAC; 2. fdk-aac; 3. faac; 4. vo-aacenc; if i understood you correctly. [22:35] <JEEB> I wouldn't necessarily put QT's AAC on top [22:35] <JEEB> I remember fraunhofer's other versions doing rather well on tests [22:35] <JEEB> @ hydrogenaudio [22:37] <LithosLaptop> Nero AAC is alot better than FAAC [22:37] <sgfgdf> JEEB, is Nero AAC somewhere near QT AAC and fdk-aac? [22:38] <LithosLaptop> Nero AAC/QT AAC/fdk-aac are very close to each other. hard to judge [22:38] <sgfgdf> JEEB, i mean the three above to be almost equal? [22:38] <sgfgdf> ah okay [22:38] <JEEB> QT | nero | fdk-aac > faac > ffaac (internal) > vo-aacenc [22:38] <sgfgdf> and about ffmpeg's aac (how should i call it?)? [22:40] <sgfgdf> are all of the codecs free for personal use? [22:41] <sgfgdf> and which of the codecs will charge you if you use them for commercial purposes? [22:41] <JEEB> I have no idea about the licensing of nero or QT's encoder for any commercial purpose [22:43] <JEEB> all the others have a license that you can read (Except for faac, that one tries to be GPL, but it isn't because it uses non-GPL reference encoder code) [22:43] <JEEB> and then there's the whole thing about taking an AAC license from VIA licensing [22:44] <JEEB> if you are going to do business it's better to ask a lawyer if you don't want to license AAC [22:45] <sgfgdf> JEEB, no, i was just curious. [22:45] <JEEB> I think I had someone try license nero for their encoding chain [22:45] <JEEB> they didn't get an answer [22:45] <JEEB> not sure if you can license QT either *shrug* [22:47] <JEEB> vo-aacenc is Apache2, ffaac is LGPL, faac is an invalid mish-mash of supposedly GPL code and one of the reference encoders, fdk-aac has a custom license from fraunhofer that is not (L)GPL compatible [22:47] <JEEB> most of these software licenses only apply towards distributing binaries [22:47] <JEEB> and the rules you have to take in case you distribute builds of them, or use the libraries somewhere [22:48] <JEEB> I think fdk-aac's license told you you had to license AAC (via VIA licensing methinks) to distribute it [22:48] <JEEB> (binaries) [22:48] <sgfgdf> btw i forgot to ask about aacplus, but we talked with a guy here about it. it appears to achieve a good quality in very low bitrate. [22:49] <JEEB> that's HE-AACv2 specific methinks, fdk-aac can do it too [22:49] <JEEB> HE-AAC(v2) is a version of AAC that cuts more information out [22:49] <JEEB> it's specifically meant for use cases of 64kbps or less [22:49] <JEEB> aka "it probably sounds close enough for the bit rate" [22:50] <sgfgdf> JEEB, is it normal to use it for encoding a movie audio? [22:50] <JEEB> if your primary need is to lower the bit rate [22:51] <JEEB> if you want keep quality, higher bit rate and LC-AAC cuts less information out [22:54] <sgfgdf> JEEB, if i lower the bit rate i will get smaller file, right? so i think i can't distinguish a difference while using 64kb/s aacplus encoded audio and 192kb/s AAC audio (which i don't know the encoder of). [22:54] <LithosLaptop> you won't expect to see HE-AAC on normal/high bitrate 720p or 1080p videos. LC-AAC more suitable [22:55] <JEEB> sgfgdf, yes -- so if you're encoding for very low bit rate scenarios, HE-AACv2 is very useful [22:55] <JEEB> as you can push the audio down to 48kbps or so while still kind of feeling similar [22:55] <sgfgdf> JEEB, what can be a such scenario? [22:56] <sgfgdf> i mean if i had to choose i will choose smaller filesize. [22:56] <sgfgdf> but you said that it is very unlikely to see it in HD videos. [22:57] <LithosLaptop> the reason is HE-AAC can't acheive transparency [22:57] <JEEB> yes, because in such scenarios generally the video bit rate is already so high that 48kbps becoming 192kbps for better audio quality doesn't really cost you much at all [22:57] <JEEB> and yes, HE-AAC(v2) in general cuts more information out [22:59] <sgfgdf> LithosLaptop, what should i understand under transparency? [22:59] <LithosLaptop> sound just like the original source [23:01] <sgfgdf> LithosLaptop, but isn't it just like the original if i can't distinguish a difference between a video and then the same video re-encoded with HE-AAC(v2)? [23:01] <LithosLaptop> 64Kbps HE-AAC sounds pleasant(not annoying like 64Kbps LC-AAC would), but not transparent. in other words it won't sound like the original [23:02] <sgfgdf> probably it's just me and other people will notice it. i don't know. [23:02] <LithosLaptop> If you can't distinguish then it is transparent to you, but I find it hard to beleive. Are you using headphones? [23:09] <sgfgdf> LithosLaptop, yes, but i'm pretty bad at such things, so probably it is just me. everytime i try to do such a comparison i find myself really struggling. [23:11] <sgfgdf> JEEB, LithosLaptop, thank you a lot for the help guys! [23:11] <LithosLaptop> no prob [00:00] --- Sun Feb 3 2013

1 0

ffmpeg-devel.log.20130201
by burek 02 Feb '13

02 Feb '13

[00:14] <Compn> pross has a bayer swscale eh [00:15] <Compn> now i forgot what codec it was that needed that [00:17] <pigoz> is swig supposed to run ok on libavformat headers? seems like it doesn't like attribute_deprecated at all. Maybe I'm doing something terribly wrong. swig -cffi path/to/avformat.h [00:18] <pigoz> oh well I'll figure out tomorrow :) [00:56] <cone-469> ffmpeg.git 03Michael Niedermayer 07master:131fac1c12ed: vc1dec: fix block_off [00:56] <cone-469> ffmpeg.git 03Sebastian Sandberg 07master:c6779c513117: vc1dec: fieldtx is only valid for interlaced frames [00:59] <michaelni> nevcairiel, do you want to review the remaining patches from "Sebastian Sandb ( *K) [FFmpeg-devel] [PATCH */3] vc1dec: *" ? [01:10] Action: Compn pings bayer rgb patch [01:11] <Compn> michaelni : should i ask kostya about those patches ? [01:14] <Compn> vc1 patches that is [01:15] <michaelni> Compn, nevcairiel worked on interlaced vc1 [01:16] <Compn> ah [01:16] <michaelni> i dunno if kostya did, but he is of course welcome to review/approve/appy the patches [01:16] <michaelni> apply [01:16] <Compn> kostya did not iirc [02:09] <cone-469> ffmpeg.git 03Allan Kristensen 07master:4789c25d83c4: matroska: fixed missing S_DVBSUB subtitles [03:20] <cone-469> ffmpeg.git 03Michael Bradshaw 07master:c58c67398d03: libopenjpegenc: add support for pix fmt gbrp (8-16 bit) [03:20] <cone-469> ffmpeg.git 03Michael Bradshaw 07master:28f36dce3e5c: libopenjpegenc: simplify switch statement [03:20] <cone-469> ffmpeg.git 03Michael Niedermayer 07master:ffd02d0c1974: Merge remote-tracking branch 'mjbshaw/master' [03:39] <highgod> Hi,I use git to get the latest version, but I can't comile correctly.There seems some errors in file libm.h,./libavutil/libm.h:52:32: error: static declaration of 'cbrt' follows non-static declaration [03:53] <michaelni> highgod, you did try make distclean ? [03:58] <Compn> highgod : did you rm the old checkout and pull a new one ? [04:02] <cone-469> ffmpeg.git 03Michael Niedermayer 07master:086566a557ee: dict: fix memleak [04:50] <highgod> Hi,I create a new folder and get the code [05:00] <Compn> does it work highgod ? [05:05] <highgod> I try to reget again,Thanks Compn. [05:05] <highgod> I want to add our openCL filter code in the latest code.we will add a OpenCL wrapper.Is it OK? [05:15] <Compn> highgod : you should ask michaelni [05:15] <Compn> someone on the list said that opencl was only good for newest intel cpu [05:16] <Compn> i dont know enough about opencl to make an opinion [05:30] <highgod> OK, I think add a option to make user can choose whether to use CPU or GPU [05:31] <highgod> I reget the code, but the same error [05:38] <Compn> hmmm [05:39] <Compn> highgod : what ./configure options are you using ? [05:42] <highgod> yes,just ./configure [05:42] <highgod> and my git set autocrlf = false [05:44] <Compn> are you using ccache ? [05:44] <Compn> what are you compiling on ? [05:44] <Compn> what arch [05:45] <Compn> highgod : we ask, because our automated ffmpeg system says everythings ok http://fate.ffmpeg.org/ [05:46] <highgod> no,I just compile on my computer using mingw [05:47] <highgod> I'm confused that why alldevice.o error. [05:48] <Compn> paste make output to pastebin or similar site [05:49] <wm4> Compn: tired of mplayer support? [05:50] <Compn> wm4 : why, people asking for me to return / [05:51] <highgod> sorry,Compn,what is pastebin? [06:00] <Compn> highgod : pastebin.com or pastebin.ca [06:01] <Compn> highgod : just a website you can paste text at for public share [06:01] <Compn> it may be banned in certain countries.... [06:01] <highgod> Thanks [06:01] <Compn> or just put the make output somewhere we can see it [06:02] <Compn> but not in the channel , too much text [06:03] <highgod> http://pastebin.ca/2309543 [06:05] <Compn> highgod : did you try rm libavutil/libm.h && git pull [06:06] <highgod> Compn : No,but I create a new folder and get the code.Should I need to do this operation? [06:07] <Compn> cant hurt to try :) [06:07] <Compn> but i dont know whats wrong [06:07] <highgod> OK,Thanks [06:07] <highgod> I will try [06:08] <Compn> your copy of libm.h looks broken , does it have the windows line endings again ? [06:08] <Compn> dos2unix app might fix it [06:15] <Daemon404> disable autocrlf in git. [06:20] <highgod> Daemon404 : I have set the autocrcf = false [06:20] <Daemon404> you need to re-clone after ofc [14:24] <cone-686> ffmpeg.git 03Diego Biurrun 07master:cb30bf7f03ac: doc/developer: Add web links for all suggested licenses [14:24] <cone-686> ffmpeg.git 03Diego Biurrun 07master:8b0fda180e99: doc/developer: Add ISC license to list of acceptable licenses [14:24] <cone-686> ffmpeg.git 03Diego Biurrun 07master:8787847dc6b3: doc/developer: Drop obsolete MPlayer reference [14:24] <cone-686> ffmpeg.git 03Anton Khirnov 07master:f2960097e42d: bink: fix a check for the first frame. [14:24] <cone-686> ffmpeg.git 03Michael Niedermayer 07master:69a6eddd607a: Merge commit 'f2960097e42ddf9a356bab6547f87906f6999e0a' [14:29] <cone-686> ffmpeg.git 03Anton Khirnov 07master:1730ca2eca42: bink demuxer: check malloc return value [14:29] <cone-686> ffmpeg.git 03Michael Niedermayer 07master:1bc51a7a74fc: Merge commit '1730ca2eca42974af0c29f38a0770997cba2f0da' [14:39] <cone-686> ffmpeg.git 03Anton Khirnov 07master:9ec8971060a0: bink demuxer: set framerate. [14:39] <cone-686> ffmpeg.git 03Diego Biurrun 07master:4eef2ed70744: ppc: fmtconvert: Drop two unused variables. [14:39] <cone-686> ffmpeg.git 03Diego Biurrun 07master:6c1a7d07eb41: Use proper "" quotes for local header #includes [14:39] <cone-686> ffmpeg.git 03Michael Niedermayer 07master:911e27068840: Merge remote-tracking branch 'qatar/master' [14:43] <divVerent> how are patches to be submitted BTW? have a question not answered on the ffmpeg.org/developer.html page [14:43] <divVerent> just git-send-email? but how would I then put in extra explanation (screenshot links, etc.) [14:44] <divVerent> or are attachments allowed on the list [14:46] <iive> attachments are ok, afaik [14:52] <Compn> divVerent : its best to put attachments and screenshots on the bug trac [14:52] <Compn> i think [14:53] <Compn> otherwise you'll be sending a screenshot to 1000+ subscribers (however big ffmpeg-devel is) [14:55] <Compn> oh [14:55] <Compn> i misunderstood [14:55] <Compn> :D [14:56] <Compn> divVerent : isnt there a intro message to get-send-email where you can add links and description and stuff ? [14:56] <Compn> i've seen people send them, a 0/* mail [14:56] <JEEB> yeah, cover thingy [14:56] <JEEB> cover page or whatever it was [14:56] <Compn> yeah that [15:00] <divVerent> Compn: I'd have sent the screenshot as link [15:00] <divVerent> 14:55:54 @Compn | divVerent : isnt there a intro message to get-send-email where you can add links and description and stuff ? [15:00] <divVerent> not that I know of [15:00] <divVerent> but well, can't use git-send-email directly anyway [15:01] <divVerent> or rather, not from the email address I would subscribe [17:48] <Marco-123> hi all. I have a board with an ARM processor (and a NEON hardware accelerator), which is running Ubuntu Linux v10.04. I am building ffmpeg and my goal is to make sure it is taking full advantage of the hardware on the board. It's a freescale IMX53 btw. After much googling, I set a few flags in the configure script, and the script tells me "NEON enabled: yes" but the encoding performance is quite slow (same as when it says " [17:48] <Marco-123> can someone provide advice on how to build this project properly? [17:48] <Marco-123> or if there is a conclusive way to tell if the hardware accelerators are being used? another point is that the configure also tells me "enabled hwaccels: " and nothing is listed there. [17:51] <durandal_1707> the options in table for lut(component options), afade(curve types) are displayed wrong, who is doxygen expert? [17:53] <divVerent> durandal_1707: speaking of the power-of-two division issue BTW which patcheck shows on my patch... just saying because I saw some other post about it [17:53] <divVerent> I intentionally ignored it, as I want to highlight that 32768 is the max sample value of S16 [17:53] <divVerent> so >> 15 is not really an option :P [17:54] <divVerent> oh, and I evaded the patcheck message this way :P [17:54] <divVerent> const double w = 1. / (sqrt(nb_freq) * 32768.); [17:54] <durandal_1707> what patcheck reported? [17:54] <divVerent> and it now says "sqrt(nb_freq) * 32768. is the max value of a fft bucket" [17:54] <divVerent> const double w = 1. / sqrt(nb_freq) / 32768.; [17:54] <divVerent> I had this in a previous version [17:54] <divVerent> then patcheck wanted me to replace the / 32768 by >> 15 [17:55] <divVerent> which doesn't even work for double :P [17:55] <durandal_1707> ahh, patcheck have own bugs... [17:55] <divVerent> just mentioning it, as some other mailing list post mentioned the same patcheck message and that it's useless now :P [17:55] <divVerent> especially given that we can expect compilers to handle this too [17:56] <divVerent> or rather, "mostly handle it" [17:56] <divVerent> i.e. for unsigned types [17:56] <durandal_1707> what other mailing list? [17:56] <divVerent> ffmpeg-devel [17:56] <cone-686> ffmpeg.git 03Carl Eugen Hoyos 07master:99eedfc40086: doc/muxers.texi: Fix mp3 picture attachment documentation. [17:56] <divVerent> there was some other unrelated post [17:56] <divVerent> right after I subscribed [17:56] <divVerent> http://ffmpeg.org/pipermail/ffmpeg-devel/2013-February/138323.html [17:56] <divVerent> this one, last two lines [17:57] <divVerent> oh, oops... haha [17:57] <divVerent> I thought this referred to patcheck, but it does not :P [17:57] <cone-686> ffmpeg.git 03Carl Eugen Hoyos 07release/1.0:5c1a1b47f378: doc/muxers.texi: Fix mp3 picture attachment documentation. (cherry picked from commit 99eedfc40086972987aa27df8b1259c8bf15b20c) [17:57] <cone-686> ffmpeg.git 03Carl Eugen Hoyos 07release/1.1:9bcb84810f53: doc/muxers.texi: Fix mp3 picture attachment documentation. (cherry picked from commit 99eedfc40086972987aa27df8b1259c8bf15b20c) [18:00] <divVerent> durandal_1707: but anyway, I did notice more false positives with patcheck :P [18:00] <divVerent> it really seems just a crude tool [18:01] <divVerent> possibly never written:intensity_colors [18:01] <divVerent> possibly constant :intensity_colors [18:01] <divVerent> e.g. it says that for the options [18:02] <durandal_1707> patcheck is mostly used for common typos you could not spot and style issues [18:02] <divVerent> yes [18:02] <divVerent> it did find accidental tabs in my code that I then fixed [18:05] <michaelni> Marco-123, you can add some av_log/printf into a neon function to be really sure if its used [18:08] <Compn> grumble [18:08] <Compn> where is mov_text in the documentation ? [18:08] <Compn> hey cehoyos :) [18:09] <Marco-123> michaelni: at the moment i am using a library that works with ffmpeg so I don't issue ffmpeg commands directly. However, for this purpose I guess I should! Would you elaborate more on what I can look for when issuing that command to tell if neon is being used? Im guessing that because its a neon function it wont even work if neon isnt being used. [18:09] <cehoyos> Hi [18:11] <cehoyos> Compn: [15:12] <Compn> durandal_1707 : yes, but is it fixed or is it a mis-duplicate tag ? :P <-- I don't understand this sentence... [18:15] <Marco-123> michaelni: thanks for your help, too. [18:18] <Compn> cehoyos : i just wanted someone to verify the bug was fixed [18:19] <Compn> since it was marked duplicate, that sentence is asking if it was a valid duplicate bug [18:19] <Compn> or a new bug [18:20] <Compn> interesting [18:21] <Compn> i couldnt get quicktime to open our ttext sample [18:22] <cone-686> ffmpeg.git 03Stefano Sabatini 07master:b868275ddbec: doc/muxers: fix level for the segment options subsection [18:22] <Compn> saste : where in the docs is the "3gpp timed text" subtitle option ? [18:23] <saste> Compn, why do you ask me? [18:23] <Compn> because you are active ? :) [18:23] <Compn> and i see you just commit some docs :) [18:23] <Compn> i'd like to fix it, if i can figure out if its missing. or it maybe that i am blind [18:23] <saste> Compn, i don't even know what "3gpp timed text" is [18:24] <Compn> mov text subs format [18:24] <saste> ubitxu? [18:24] <Compn> hes in japan [18:24] <Compn> soaking up all that radiation [18:24] <saste> i don't think it is documented at all [18:25] <saste> yes japan is cheap since then [18:25] <durandal_1707> saste: the lut rgb options and afade curve type options in doxy are displayed wrong what is correct syntax? [18:25] <saste> don't miss the opportunity [18:25] <Compn> so theres no list of subtitle formats and their options ? [18:25] <Compn> it would be cool to visit japan again [18:25] <saste> Compn, ffmpeg -h full may help [18:26] <Compn> saste : it doesnt list either [18:29] <saste> durandal_1707, item foo NL description [18:29] <saste> i'm going to fix lut [18:31] <Compn> http://ffmpeg.org/pipermail/ffmpeg-cvslog/2012-August/053256.html [18:31] <Compn> theres the commit [18:32] <cehoyos> Compn: Do you have any reason to assume it was not a duplicate? [18:34] <cehoyos> Compn: Muxing mov_text in mov is not QT-compatible, see ticket #1845 [18:34] <Compn> cehoyos : no reason. just curiosity [18:36] <Compn> ok thanks for bug 1845 [18:36] <Compn> we have mov_text samples [18:36] <Compn> that work in quicktime [18:36] <Compn> cehoyos : http://samples.ffmpeg.org/mov/subtitles-embedded/ [18:36] <Compn> at least http://samples.ffmpeg.org/mov/subtitles-embedded/BCDisassembly1RightSideTxt… works in quicktime, did not test the rest [18:38] <cehoyos> Compn: Which QT version did you test? [18:38] <Compn> 7.7.3 [18:39] <cehoyos> I ask because this file is around 10years older than the QT version mentioned in ticket 1845 [18:40] <cehoyos> Could you try ffmpeg -i BCDisassembly1RightSideTxt2.mov -codec copy -map 0 out.mov and test if out.mov shows the subtitles in QT? [18:40] <Compn> sure [18:41] <Compn> yes that works [18:41] <Compn> in quicktime [18:41] <cehoyos> Also with -vcodec mpeg4 -qscale 2 -strict -2 -acodec aac ? [18:42] <Compn> theres no -scodec listed [18:42] <Compn> do you want -scodec copy ? [18:42] <cehoyos> Could you test both -scodec copy and scodec mov_text ? [18:42] <Compn> sure [18:43] <Compn> -vcodec mpeg4 -qscale 2 -strict -2 -acodec aac -scodec copy works in quicktime [18:44] <Compn> -scodec mov_text shows weird things in qt [18:44] <Compn> like the text is 100 pt font stuck in a 20pt box, and color dark red maybe ? [18:44] <Compn> yes, if i fullscreen the video, i can see text [18:45] <Compn> kind of hard to explain :) [18:45] <cehoyos> You could try to enlarge the video radically: -s 1280x960 [18:47] <Compn> cehoyos : http://i.imgur.com/n7AeD4l.jpg [18:47] <Compn> thats with -scodec mov_text [18:48] <cehoyos> Thank you for the tests! [18:50] <Compn> in my screenshot, you can see that the subs are at the top with ffmpeg's reencode [18:50] <Compn> but in the original file, the subs are at the bottom... [18:51] <Compn> probably the alignment tag does nt get copied [18:51] <Compn> for some reason [18:51] <Compn> cehoyos : thank you for getting bug reports in order :) [18:52] <cehoyos> Regarding ticket #1840: Your suggestion for -c:s mov_text will not work in 50% of all cases... [18:52] <cehoyos> Especially not for the sample from ticket #756 [18:54] <Compn> if (textsubs) then [18:54] <Compn> i think dvdsub (vobsub?) in mp4/mov is rare [18:54] <cone-686> ffmpeg.git 03Stefano Sabatini 07master:902fa77fd716: doc/filters: fix syntax and description for the lut* options [18:54] <cone-686> ffmpeg.git 03Stefano Sabatini 07master:986540b65f6e: doc/filters: create a dedicated examples section for lut*, and itemize [18:54] <cone-686> ffmpeg.git 03Stefano Sabatini 07master:423856c4ae85: doc/filters: fix syntax and typo for the afade curve option [18:56] <Compn> oh that cpcdemo.mov is Stream #0:2(eng): Subtitle: eia_608 (c608 / 0x38303663) [18:57] <Compn> thats different than mov_text :\ [18:57] <cehoyos> (Apart from the fact that dvdsub, dvbsub, pgssub and maybe even xsub are not rare:) How is the rareness related to giving bad advice? [18:58] <Compn> cehoyos : you are right, how about "please use -scodec (supported mov subtitle list)" then ? [18:58] <cehoyos> Looks like ticket #846 [18:58] <cehoyos> I think the current error message is the best possible [19:01] <Compn> ok, then feel free to close 1840 bug as duplicate of 1845 [19:02] <cehoyos> But 1840 was never a duplicate of 1845 (you insisted yesterday in making it one) [19:02] <Compn> comment #3 pretty much makes it his problem [19:02] <Compn> makes it a duplicate, i mean [19:03] <Compn> ok mark it invalid then :) [19:03] <Compn> no worries [19:03] <Compn> cehoyos : new problem, where is mov_text listed ? [19:03] <Compn> i think its missing in documentation [19:03] <Compn> i cant find supported subtitle list either [19:03] <cehoyos> No, comment three changed the original report to something (more important but) unrelated [19:04] <cehoyos> I sometimes commit documentation fixes (saste: I hope you are ok with my changes an hour ago), but I don [19:04] <cehoyos> ... I don't feel responsible for documenting features I don't know much about. [19:05] <Compn> i dont mind documenting it myself [19:05] <Compn> i mean [19:05] <Compn> i want to fix it [19:05] <Compn> i just dont know where it should go :) [19:05] <cehoyos> Then send a patch [19:06] <Compn> curious why this mov_text does not work, no video stream ? http://samples.ffmpeg.org/mov/subtitles-embedded/nova_eu_3012c01_caps.mov-c… [19:07] <Compn> it works in quicktime i mean [19:07] <Compn> but not ffplay [19:12] <cehoyos> compn: I believe it would work with ffplay but ffplay does not support mov_text (or does it and I just don't know how?) [19:13] <Compn> cehoyos : oh, i forgot ffplay doesnt support mov_text [19:13] <Compn> then its my bad [19:18] <Compn> cehoyos : are there any bugs i should look at ? [19:18] <cehoyos> Compn: mov_text is documented afaict: http://ffmpeg.org/general.html#Subtitle-Formats [19:19] <Compn> i'm not good with programming .. [19:20] <Compn> cehoyos : yes, i found it , mov_text is listed with ffmpeg -codecs [19:21] <Compn> i was trying ffmpeg -formats ... i'm dumb :( [19:21] <Compn> is a subtitle a format or codec, who knows :D [19:25] <durandal_1707> no more comments on showspectrum? [19:26] <saste> durandal_1707, wait a day or so, so ubitux can comment [19:26] <durandal_1707> ubitux: seriously? [19:28] <saste> durandal_1707, he's the author of the filter [19:28] <saste> i can't have a look at it now, i'll try to do it tonight [19:32] <durandal_1707> saste: not listed in MAINTAINERS for that file ........................ [19:34] <nevcairiel> he is listed as author at the top of the file =p [19:39] <durandal_1707> nevcairiel: so i would need to wait for every single persona persona at the top of the file? [19:40] <nevcairiel> its only one person, and if that person is an active commiter to ffmpeg, yes, you should [19:41] <nevcairiel> its just polite to ask the author to review a patch to his code [19:41] <durandal_1707> well i will just add another persona at the top of the file which will simplify things a lot [19:41] <nevcairiel> that doesnt change who the author of the file is [19:45] <durandal_1707> even if code was so badly writen that it needs rewrite or adds more code that was already present .... [19:57] <saste> durandal_1707, i'm asking you to wait a day, not till the end of the world [19:58] <durandal_1707> saste: but end of world is near!!!! [19:58] <saste> durandal_1707, no it's already passed [19:59] <saste> can't say when it will be the next one [20:03] Action: durandal_1707 will ask ubitux for special authorisation code [20:04] <Compn> lol [20:04] <Compn> durandal_1707 : work on something else in the meantime :) [20:04] <Compn> i remember ffmpeg used to have much longer patch review times [20:04] <Compn> 37 patch resubmissions :D [20:05] <Compn> wonder what the highest one was [20:05] <saste> durandal_1707, the-matrix-effect filter [20:05] <Compn> try 42 ? [20:06] <saste> durandal_1707, to say the truth, i have an unfinished port of it, based on jwz code [20:06] <durandal_1707> of what? [20:07] <durandal_1707> matrix effect? it still not in? [20:07] Action: durandal_1707 sad [20:08] <burek> is "Subtitle: dvb_teletext" the same as "Subtitle: dvb_subtitle" ? [20:09] <Compn> no [20:09] <Compn> well [20:09] <Compn> i dont know :D [20:09] <Compn> i dont think so [20:09] <nevcairiel> no its not the same [20:09] <nevcairiel> dvb has two kind of subtitles [20:10] <burek> it seems that ffmpeg converts dvb_teletext to dvb_subtitle when -c copy is used [20:10] <nevcairiel> sounds like a bug [20:10] <burek> yeah, i guess i should report it [20:10] <burek> generally, should i wait and ask for people to report a bug [20:11] <burek> or if i notice in on the forum, should i just report it myself [20:11] <burek> i mean, do i somehow take away their credits or something? [20:11] <burek> because i'd not want to do that [20:12] <burek> notice it* [20:12] <nevcairiel> credit for reporting a bug? [20:12] <burek> well i dunno, i just ask.. :) [20:12] <nevcairiel> you could always mention the person and link to the forum [20:12] <nevcairiel> if you care [20:12] <burek> yeah, that could be ok i guess [20:13] <burek> ok, thanks :) back to reporting it then :) [20:39] <Marco-123> hi all. (my sys: ubuntu 10.04, arm proc, neon media processor). i am using a software library that uses FFMPEG. so i dont work with ffmpeg directly. however, i suspect that my hardware accelerators are not being used. so now I am to get into things. id like to write a minimal app, that will tell me (yes/no, conclusively) whether or not the hardware accelerators are being used by the ffmpeg libraries. could someone help me w [20:42] <Marco-123> i was thinking of having a sample videos, maybe encoding/decoding them with various codecs (h263plus or h264), audio and video, and trying to see if the hwaccel is being used along the way (??). or maybe there is a command that will tell me info directly. [20:42] <burek> does ffmpeg support hwaccel right now? [20:44] <cehoyos> Marco-123: Are you talking about hwaccel or asm / neon support? [20:52] <michaelni> nevcairiel, maybe you want to comment/review on the 2 alterantive [PATCH 1/3] vc1dec: the second field is written wrong to the picture / vc1dec: Fix destination pointers in context for fields [20:53] <michaelni> its better to have a second oppinon on which is better than me just pushing one in a few days [20:53] <nevcairiel> all i can do is check the bitstream parsing against the spec, because thats really all i know about how vc1 works :d [20:55] <michaelni> :( [20:56] <michaelni> sadly no bitstream parsing changes in the ones i pointed to, but thanks anyway [20:57] <nevcairiel> thats also the reason my patch was never finished, i wrote the bitstream parsing, and never had time or motivation to figure out how it fits into everything else [20:59] <Marco-123> cehoyos: hmm. maybe my understanding is off. i have a neon chip (media processor?), and i thought that this was the hardware accelerator. maybe my terminology and understanding of the thing is wrong. [20:59] <michaelni> nevcairiel, [PATCH 3/3] vc1dec: interlaced B-frames <----- contains some bitstream parsing in the added code [21:00] <nevcairiel> neon is not what we usually call a "hardware accelerator", neon is just "SIMD", what in our desktop CPUs is MMX/SSE [21:09] <Marco-123> nevcairiel: ah. ill read about hardware accelerators now. in the mean time, can an SIMD help my encoding/decoding performance? [21:09] <nevcairiel> yes, significantly [21:09] <nevcairiel> without SIMD, de/encoding is much slower [21:10] <Marco-123> if my board has a neon SIMD, might it also have a GPU? or is there an overlap of these two functionalities and only one should be found? [21:10] <nevcairiel> neon is part of the CPU, it has nothing to do with your GPU, so sure, it can have a GPU as well [21:10] <nevcairiel> although in a SoC stuff gets mixed :p [21:13] <Marco-123> alright i will look into my board's datasheet again for GPU details (it probably doesnt have one). so i guess my original question is: what is a good approach for conclusively knowing whether or not my ffmpeg encoding/decoding is acutally making use of the neon simd? [21:15] <wm4> are there fate tests for dvdsub? [21:16] <wm4> Marco-123: my guess is that if there's ARM specific asm in the source code, it uses neon [21:17] <nevcairiel> his question seems to be how to know if that asm is actually used =p [21:17] <Skyler_> see if neon: yes in configure? [21:17] <nevcairiel> isnt there a runtime check now too [21:17] <wm4> Marco-123: compile with asm disabled, and compare speed to normal configuration [21:18] <wm4> or rather, neon disabled, if there's a configure option [21:18] <Marco-123> nevcairiel: after much googling, im able to build ffmpeg so that the configure script reads "NEON enabled: yes" [21:18] <Marco-123> --enable-neon haha [21:18] <Marco-123> still took much googling! [21:18] <nevcairiel> i would've thought configure enables such things on its own by probing [21:18] <wm4> well then try with --disable-neon and compare speed, then you should get some conclusive results [21:19] <nevcairiel> but what do i know about arm [21:19] <Marco-123> wm4: yes this is a good plan. forgive my ignorance but I can't find on google what ASM in this context stands for [21:19] <Marco-123> assembler? [21:20] <wm4> yes [21:20] <nevcairiel> yes, assembly code, hand-optimized code that uses neon [21:20] <Marco-123> thank you [21:21] Last message repeated 1 time(s). [21:21] <Marco-123> so in your opinions: if the configure script reads "NEON enabled: yes", should I stop worrying about ffmpeg using it? [21:21] <Marco-123> is it a good indicator? [21:22] <wm4> sure, why not [21:24] <Marco-123> ok. to answer why not: im still seeing not so great encoding performance. granted my set up is embedded and thus not as fast as a desktop computer, so this might be the cause [21:26] <Marco-123> guys thanks for your help. you've cleared up a lot of fuzziness in my brain about this stuff [21:26] <Marco-123> :D [21:41] <Skyler_> If you want to be absolutely sure, disable NEON and see how much slower it is ;) [23:05] <cone-774> ffmpeg.git 03Matthieu Bouron 07master:5a67e30b1c71: ffmpeg: fix broken channel_layout option [23:10] <cone-774> ffmpeg.git 03Matthieu Bouron 07release/1.1:c8dace272829: ffmpeg: fix broken channel_layout option [23:29] <cone-774> ffmpeg.git 03Carl Eugen Hoyos 07master:6a50e8a190bc: Do not change codec in flv streams if the user has forced a codec. [23:42] <cone-774> ffmpeg.git 03Carl Eugen Hoyos 07release/1.0:299dc3baeaaf: Do not change codec in flv streams if the user has forced a codec. [23:42] <cone-774> ffmpeg.git 03Carl Eugen Hoyos 07release/1.1:3445bec6fc1d: Do not change codec in flv streams if the user has forced a codec. [23:58] <cone-774> ffmpeg.git 03Ronald S. Bultje 07master:8af915c21b03: vp56: remove clear_blocks call, and clear alpha plane U/V DC only. [00:00] --- Sat Feb 2 2013

1 0

ffmpeg.log.20130201
by burek 02 Feb '13

02 Feb '13

[02:51] <ulatekh> I accidentally encoded a .flv video with the wrong audio, and I'm trying to replace it, but "ffmpeg -i old.flv -an -vcodec copy -i new.mp3 -acodec copy -f flv new.flv" exits with "Unknown decoder: 'copy'". Does anyone know how do I do what I want? [02:51] <ulatekh> This is the latest git-master ffmpeg from about a week ago. [03:00] <ulatekh> Anyone awake? [03:05] <ulatekh> OK, one sec. [03:06] <ulatekh> http://pastebin.com/j8q7HW6s [03:07] <llogan> also paste the output of: ffmpeg -i old.flv -i new.mp3 [03:08] <ulatekh> OK, one more sec. [03:09] <ulatekh> http://pastebin.com/ip2AHAGS [03:09] <ulatekh> Thanks for looking into this for me, llogan. [03:09] <llogan> ffmpeg -i old.flv -i new.mp3 -map 0:0 -map 1:0 -c copy -shortest out.flv [03:10] <ulatekh> OK, I'll try that, one sec. [03:10] <llogan> note that your new audio stream is ~6 seconds shorter duration [03:10] <ulatekh> The old audio stream, being the wrong one, was too long. [03:11] <ulatekh> No errors...let me look at the result [03:12] <ulatekh> Looks like it worked! Awesome! I'll have to look up the meaning of "-c copy -shortest". [03:12] <ulatekh> Still...shouldn't "-vcodec copy" have worked? [03:13] <llogan> yes, but you were applying it to the decoder of new.mp3 [03:13] <llogan> in other words, you were using -vcodec copy as an input option [03:14] <ulatekh> Oh...I thought I had to put those options after the input file they related to. [03:15] <ulatekh> What would have been the correct command line with "-acodec copy" and "-vcodec copy"? [03:15] <llogan> -c copy does the same as -acodec copy and -vcodec copy (and more) [03:16] <llogan> An empty stream specifier matches all streams, for example -codec copy or -codec: copy would copy all the streams without reencoding. [03:16] <llogan> http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1 [03:16] <llogan> and -c is an alias for -codec [03:17] Action: llogan is going to be afk for a while [03:17] <ulatekh> Hmmm, looks like I have to take my knowledge of the ffmpeg command line up to the next level. Thank you very much for your help! [03:55] <sunjun_> http://www.beyond3d.com/content/articles/8/ [04:35] <temhawk> wow, why is the ffmpeg syntax so hard to read?? [04:36] <temhawk> take this for example: http://pastie.org/6003419 wtf!? [04:36] <temhawk> that's just for combining two videos ¬¬ [08:27] <ledil> Hi, Im getting this error: http://pastebin.com/64JFAwjM, Ive used ffmpeg 0.5 before and I have now switched to ffmpeg 1.1 ... [08:40] <Sunil_> for MPEG-ts file , wanted to change service name, provider . tried -metadata service_name="Some Channel" , it seems ti does not change service name [10:59] <buhman> wow, that's some dark magic there: http://pastie.org/6003419 [10:59] <buhman> I would have put them in mpg containers and simply concatenate [11:11] <samuele> Hi everyone, I have a little question on the LICENSE [11:14] <samuele> I found an android port of FFmpeg compiled with --enable-gpl free just for personal use. I can't bundle it in any developed new app. Is this a violation in terms of license, right? [11:16] <khali> samuele: definitely [11:16] <samuele> Thanks, it was just for confirmation. I'm about to say this over there. [11:17] <khali> actually I'm a bit too enthusiastic with "definitely" [11:17] <samuele> over thei website I mean [11:17] <khali> it probably depends on the other build flags [11:17] <Mavrik> wait [11:17] <samuele> I can paste them if you want. [11:17] <Mavrik> samuele: can't really parse your sentence. does it have "--enable-gpl" or not? [11:17] <samuele> yess sir [11:18] <Mavrik> does it have "--enable-nonfree"? [11:18] <khali> samuele: if includes other --enable flags which cause non-GPL code to be included then you must check the terms of these other licenses [11:18] <samuele> nope [11:19] <Mavrik> so, if you only have "--enable-gpl" without "--enable-nonfree" your ffmpeg binary is GPL licensed [11:19] <Mavrik> which means you can distribute it with your app, you can call it and everything [11:19] <samuele> OK: checked. "--enable-nonfree" not included. [11:19] <khali> thanks Mavrik for being more accurate that I was [11:20] <Mavrik> you may not link against ffmpeg libraries without making your code available under GPL license though [11:20] <Mavrik> khali: just being more specific, you were right the entire time ;) [11:20] <samuele> Mavrik: this is wat I meant with my question. [11:20] <samuele> so should I point this out to them? [11:20] <Mavrik> samuele: to who? [11:21] <Mavrik> and as I said... you can distirbute "ffmpeg.exe" and call it from other app to do encoding without violating GPL [11:21] <samuele> sorry: to who built this FFmpeg port to android and released NOT for free if I want to bundle it in an App of mine [11:22] <Mavrik> samuele: are you going to call ffmpeg binary (so use "runtime.exec("ffmpeg -i blahblah")) or are you going to call avcodec_decode... things manually? [11:23] <samuele> I don't really know, because the project is meant to be used as a library. I'm not an expert here, but I can go through further search [11:24] <Mavrik> ok [11:24] <Mavrik> in first case you may use it, in second your code has to be GPL as well [11:24] <Mavrik> that's the distinction [11:24] <samuele> do you want the links to the source code? [11:24] <Mavrik> I don't have time to really check it right now [11:25] <samuele> ah ok. anyway should be free to use. And my app is also GPLv3... [11:25] <samuele> ok. [12:08] <khali> is there a way to ask ffmpeg for the help/format of a specific filter? [12:08] <khali> -filters gives the list of filters but that is very succinct [12:12] <MessedUpHare> Hi all, I'm trying to figure out a way of programatically recording the output of -vf cropdetect - but i'm struggling to do so as the output is all on a single line [12:12] <MessedUpHare> is there something I'm missing here [12:13] <MessedUpHare> In an ideal world, I would like to end up with a text file which states times (or frame number) along with a detected change of "-vf crop" values [12:27] <MessedUpHare> not to worry, awk has saved the day [12:36] <someone-noone> Hello! Where can I read about how can I do Annex B->AVCC and AVCC->Annex B converting in h264 for dummies? :) I found lot information in internet but I there is smth I don't understand (I can't properly parse my h264 annex-b stream) [13:02] <relaxed> someone-noone: To demux from mp4 use ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264 [13:02] <someone-noone> relaxed, dude thanks, but I want to make my own demuxer and can't find sutiable specs for that [13:03] <someone-noone> relaxed, I've read ffmpeg code but I'd rather look at specs [13:04] <relaxed> I don't believe the h264 spec is free but that doesn't mean you can't find it for free. [13:05] <JEEB> it is [13:05] <relaxed> it is free? [13:05] <JEEB> they release it for free after six or so months after it's released [13:05] <JEEB> and the last updated spec was released in 12/01 [13:05] <relaxed> ok, maybe you can point him in the right direction. [13:05] <JEEB> I already did [13:05] <JEEB> I linked him the H.264 spec before [13:05] <someone-noone> I've already read 'bout avcc format [13:05] <someone-noone> it's clear for me [13:06] <JEEB> relaxed, if you don't believe it's free here you go http://www.itu.int/rec/T-REC-H.264-201201-I :) [13:06] <someone-noone> but I can't found any information about Annex B(actually, I can, but it doesn't seem to be true) [13:06] <relaxed> I have no reason to doubt you. [13:06] Action: relaxed sometimes makes wrong assumptions [13:06] <JEEB> ISO generally has its specifications locked [13:07] <JEEB> ITU-T seems to be more lenient with some specs [13:07] <JEEB> H.262 / MPEG-2 Video seems to be locked down, though [13:07] <JEEB> I wonder what they'll do with H.265, for adoption a similar protocol as they do with H.264 would be nice [13:14] <JEEB> also, no... seems like ISO actually has this specification available for free as well [13:14] <JEEB> http://standards.iso.org/ittf/PubliclyAvailableStandards/index.html [13:14] <JEEB> I always miss the "you can also get this specification for free from the ITTF" on the ISO spec page [13:27] <someone-noone> how can I produce avcc stream from ffmpeg(using libx264)? I tried to do this from .mp4: ffmpeg -i test.mp4 -c:v copy -an -f h264 test.h264 [13:28] <someone-noone> But it seems that output file has own mp4 format (first byte is not equal to 1) [13:28] <someone-noone> and also result is not demuxable with ffmpeg [13:29] <someone-noone> however, If I do ffmpeg -i test.mp4 -c:v libx264 -an -f h264 test.h264 [13:29] <someone-noone> the result is always in annex b [13:34] <brx_> my ffmpeg question http://pastebin.com/QHHtHL0K [13:44] <khali> brx_: I think your best chance is to add an empty audio channel to one.mp4 [13:45] <brx_> khali, do i do that durig the command where I create the video from jpg? [13:46] <brx_> i couldnt find how to do this when searching the docs [13:46] <brx_> ffmpeg -y -loop 1 -i blah.jpg -t 3 -vcodec libx264 one.mp4 [13:46] <khali> yes I'd do that in the first step [13:46] <brx_> what must i add? [13:46] <khali> I don't know, I never had to do that [13:47] <brx_> ok [13:48] <khali> brx_: I think you can pass -i twice [13:49] <khali> brx_: prepare a silent .wav file and pass it that way [13:49] <brx_> khali, ok i will try that [13:49] <brx_> thanks [13:50] <khali> you may have to pass -loop twice, but I'm not sure if it supports anything else than still pictures [13:50] <khali> brx_: so you may have to make your .wav file the right size it the first place [13:51] <brx_> ok [14:58] <digited> hi [14:59] <digited> are there any examples of encoding video with adding fancy effects from showwaves filter? [14:59] <digited> (also, ffplay example with showwaves/showspectrum doesn't work) [17:39] <Marco-123> hi all. I have a board with an ARM processor (and a NEON hardware accelerator), which is running Ubuntu Linux v10.04. I am building ffmpeg and my goal is to make sure it is taking full advantage of the hardware on the board. It's a freescale IMX53 btw. After much googling, I set a few flags in the configure script, and the script tells me "NEON enabled: yes" but the encoding performance is quite slow (same as when it says "NEON e [17:39] <Marco-123> can someone provide advice on how to build this project properly? [17:40] <Marco-123> or if there is a conclusive way to tell if the hardware accelerators are being used? another point is that the configure also tells me "enabled hwaccels: " and nothing is listed there. [17:53] <samuele> Hi. I have a question about compiling [17:54] <samuele> if I add "--enable-libmp3lame" I get the error "libmp3lame >= 3.98.3 not found" [17:54] <JEEB> see config.log [17:54] <JEEB> have you built lame yourself or trying to utilize a package in your OS? [17:55] <samuele> I downloaded the libmp3lame source, compiled and I have libs in /usr/local/libs [17:56] <samuele> I need mp3 in an ffmpeg for android. The build w/o libmp3lame works OK [17:57] <samuele> config.log : error: lame/lame.h: No such file or directory [17:57] <khali> samuele: odds are that /usr/local isn't looked at [17:58] <samuele> I also tried adding in the build script "export LD_LIBRARY_PATH=/usr/local/lib" [17:59] <khali> samuele: LD_LIBRARY_PATH is for runtime IIRC [17:59] <khali> won't help configure find includes [17:59] <samuele> :8 [17:59] <samuele> ok [17:59] <khali> samuele: did you try --extra-cflags=-I /usr/local/include ? [18:00] <khali> hmm, without space [18:00] <khali> --extra-cflags=-I/usr/local/include [18:00] <samuele> trying now. thanks. [18:00] <JEEB> if your compiler doesn't look there by default, you need both that [18:00] <JEEB> as well as --extra-ldflags=-L/usr/loca/lib [18:00] <khali> and --extra-ldflags=-L/usr/loca/lib [18:00] <samuele> ok [18:00] <JEEB> *local [18:00] <khali> ah, JEEB was faster ;) [18:01] <samuele> ok. call you back. [18:01] <JEEB> also PKG_CONFIG_PATH=/usr/local/lib/pkgconfig is useful [18:01] <JEEB> for some libs [18:21] <samuele> hei khali and JEEB [18:22] <samuele> giving "./configure --enable-libmp3lame" works [18:22] <samuele> but putting --enable-libmp3lame in the build.sh script provided with this project don't [18:22] <samuele> so I think it's the script that has to be checked [18:33] <samuele> http://pastebin.com/D3nqNhpd [18:35] <khali> samuele: ah, you're cross-compiling? [18:35] <samuele> yes, for android. [18:35] <samuele> I think this is bigger than what I can handle [18:36] <samuele> and I think this sounds realy bad in English... [18:37] <samuele> LoL [18:38] <samuele> "--sysroot=$SYSROOT" can be this? [18:40] <samuele> SYSROOT=$NDK/platforms/android-3/arch-arm [18:40] <samuele> NDK=${HOME}/android-ndk [18:41] <khali> samuele: I've never done that, sorry [18:42] <khali> samuele: first question is, how did you build libmp3lame? it must be cross-built too, otherwise this can't work [18:42] <samuele> not cross-built... [18:43] <khali> samuele: so start with this [18:44] <khali> samuele: and then you shouldn't install it to /usr/local, that would be for the host architecture [18:44] <khali> samuele: you must setup a tree for your android libs [18:46] <samuele> sorry. I missed the lines between "so start with this" and "and then you shouldn't..." [18:46] <samuele> I was reading above [18:48] <nilsge> hi. In Archlinux ffmpeg is built withou support for jack audio connection kit (-f jack or -f alsa -i jack both say no jack and ffmpeg -formats does not list it). [18:48] <nilsge> What is the configure switch to include it? --enable-jack or --enable-libjack are unknown [18:53] <nilsge> I built from git anyway and it lists "jack" under indev now, at configure stage, at leas [18:54] <nilsge> t [19:03] <creep> hi [19:03] <creep> why do people share TS and cam movies over net? ;/ [19:03] <creep> it is so much worse than going in a movie and watch it [19:38] <troy_s> Anyone familiar in here with calculating the exact bit width of a raw Y' / Cb' / Cr' data stream in the avframe? [19:38] <troy_s> YUVJ420P reports 12bpp, but the data is in fact only 8 bits wide. [19:42] <JEEBsv> the 12bit comes from the fact that since you have Cb and Cr 8+8 for 2x2 area, you then get 4 bits per one (luma resolution) sample [19:42] <troy_s> fflogger / durandal_1707 I'm using the API. [19:42] <JEEBsv> so yeh, that value does not really tell you the byte size of a single value [19:42] <troy_s> JEEBsv: Right. So is there a method to calculate the actual amount of data stored in a single unit of Cb / Cr? [19:43] <troy_s> JEEBsv: Case in point, if I am attempting to yank out the raw 10 or 12 bit data from a DNxHD or ProRes stream, is there a method to figure out the width of the unit? [19:43] <durandal_1707> isn't that stored in container/bistream [19:44] <troy_s> durandal_1707: Bit depth reports 12 as expected for YUVJ420P for example. I'm willing to admit I am missing some other variable to check. [19:44] <durandal_1707> only if you found out that all lower bits are zero [19:44] <troy_s> durandal_1707: I can't see that as being too effective given the nature that the data might be zero. [19:45] <JEEBsv> troy_s: if it's decoded already, it's probably in a format where a 16bit variable is used if it's >8bit , it depends on the PIX_FMT [19:45] <JEEBsv> YUV420P is 8bit [19:45] <troy_s> I'm thinking there is an obvious method to determine the width of a given unit of Cb / Cr (in the case of YUVJ420P, it's 1 byte per sample.) [19:45] <troy_s> JEEBsv: Yes. I'm just hoping that there is a variable to read. I'd had to make guesses based on PixFmt as to the width of the chroma. [19:46] <troy_s> JEEBsv: Similar to the bitwise shift for chroma widths and height for example. [19:48] <troy_s> Looks like there might be a pixel_shift value. Urgh. [19:49] <durandal_1707> so you actually want bits per pixel? [19:55] <troy_s> durandal_1707: I'm dumping the raw YCbCr [19:55] <troy_s> durandal_1707: So yes... I'm looking for the exact width of a Cb / Cr / Y value in the data, as opposed to me guessing. [19:59] <durandal_1707> troy_s: each pix fmt have its descriptor in libavutil/pixdesc.c [20:03] <troy_s> durandal_1707: Yes. I just found what MN replaced for the avg_bits_per_pixel - av_get_padded_bits_per_pixel. [20:03] <llogan> burek: i should stop opening an reply box and replying half an hour later...i didn't see your post until after. [20:03] <burek> it's ok :) [20:03] <burek> also, i didnt thank you for helping me there [20:03] <burek> so, thank you :) [20:04] <llogan> nice IMPORTANT box [20:04] <burek> i appreciate it :) [20:04] <durandal_1707> troy_s: that is how it is stored in memory i guess ... [20:04] <burek> well, i had to make it ridiculously big and red [20:04] <burek> because people just didnt read it at all.. :S [20:04] <llogan> now lets see how many people still miss it [20:04] <troy_s> durandal_1707: Yes... looks like those are the two tools. I was already referencing the log2_chroma_h and w for shifting, but I need to get the actual value. [20:04] <troy_s> durandal_1707: Thanks for your help by th wya. [20:08] <divVerent> stupid question, but what is the "proper" way to use the showspectrum filter with ffplay? [20:08] <divVerent> as in, is there anything simpler than ffplay -f lavfi 'amovie=filename.mp3, showspectrum' [20:08] <divVerent> because, THIS one can't seek [20:13] <saste> divVerent, no simple solution ATM [20:13] <saste> i'm working right now on -af in ffplay [20:13] <saste> but that doesn't fix the problem [20:13] <saste> another solution would be to port filter_complex to ffplay [20:13] <divVerent> saste: for the specific case BTW [20:13] <saste> or to add seek support in movie [20:14] <divVerent> in the long run, don't we want to get rid of ffplay's own visualization [20:14] <divVerent> and just use showspectrum? [20:14] <saste> divVerent, yes [20:15] <saste> but code doesn't write it itself (yet) [20:15] <divVerent> sure [20:15] <saste> which is a shame [20:15] <divVerent> and it doesn't seem trivial [20:15] <divVerent> also due to lack of -filter_complex :P [20:16] <divVerent> the alternative BTW would be making amovie able to seek [20:16] <divVerent> how likely is that? [20:16] <divVerent> (and movie too, of course) [20:19] <saste> divVerent, i can only say that is on my todo list, but that's a long list [20:20] <divVerent> hehe [20:20] <divVerent> it also doesn't seem trivial [20:20] <divVerent> just wishful thinking... but if it WERE done, it could become ffplay's ONLY input [20:20] <divVerent> (and other options remapped to using the movie source) [20:21] <divVerent> actually, ffmpeg's too... for ffmpeg it seems even more viable [20:21] <divVerent> now if only it wouldn't need weird filename escaping... ;) [20:47] <patbam> hi, how can i convert a NIST Sphere file to wav? I've tried & failed to find it in the docs [20:50] <burek> ffmpeg -i SOMETHING out.wav [20:50] <burek> where SOMETHING is your file [20:51] <patbam> i got this output: http://pastie.org/6019879 does it mean my file is borked? [20:51] <patbam> (& thanks) [20:52] <Gergov> guys how to create an flv file from an mp3 and a .jpg with decent quality ? [20:54] <Gergov> is there some standard terminal command that dont involve 1000 options [21:07] <llogan> Gergov: to what output format(s)? [21:09] <llogan> patbam: can you provide a sample file? [21:10] <Gergov> llogan, .flv [21:10] <Gergov> i want to upload a song to youtube [21:10] <patbam> llogan: i'm afraid i cant because the files are under copyright to a university :( [21:10] <sacarasc> Gergov: You don't need FLV to upload to youtube, you can do pretty much anything. [21:11] <Gergov> sacarasc, well then i dont care about the output format :) if any will do the work [21:11] <llogan> patbam: i found one and ffmpeg decodes it with no problem. [21:11] <patbam> llogan: hmm, did oyu just use ffmpeg -i foo.sph foo.wav? [21:11] <patbam> maybe my files are borked somehow [21:11] <llogan> no, i just did: ffplay speech.sph [21:12] <sacarasc> Gergov: ffmpeg -loop_input -i foo.jpg -i blah.mp3 -shortest -c:v libx264 -preset slow -crf 18 -c:a copy output.mkv [21:12] <sacarasc> I think that would work, but I am a little rusty. [21:12] <llogan> Gergov: https://ffmpeg.org/trac/ffmpeg/wiki/EncodeforYouTube [21:13] <llogan> s/-loop_input/-loop 1 [21:13] <Gergov> sacarasc, i get Unrecognized option 'c:v' Failed to set value 'libx264' for option 'c:v' [21:13] <sacarasc> Or just upgrade to a newer version. [21:14] <llogan> patbam: i guess your file is damaged, or your ffmpeg version is too old, or maybe there is a sph variant that ffmpeg doesn't understand [21:14] <patbam> llogan: this is the file header: http://pastie.org/6020822 i'm wondering if the "sample_coding -s27 ulaw,embedded-shorten-v2.00" bit is suspicious. i think it's just a variant of ulaw. is there a way i can force ffmpeg to try using ulaw to decode (demux?)? [21:14] <Gergov> sacarasc, ffmpeg version 0.8.5-6:0.8.5-0ubuntu0.12.10.1 is this old? [21:14] <sacarasc> That's probably not even ffmpeg. [21:14] <JEEBsv> that's libav, if you want to use libav, switch to the 'avconv' command [21:15] <Gergov> sacarasc, i typed aptitude install ffmpeg :) it should be it. or not ? [21:15] <JEEBsv> nah, debian/ubuntu use libav [21:15] <JEEBsv> (and gentoo now by default) [21:15] <Gergov> i see [21:16] <llogan> do they laso have a fake ffmpeg package? [21:16] <llogan> *also [21:16] <JEEBsv> the ffmpeg binary in libav is outdated (and was later removed), so 'avconv' is the command you'd want to use when using libav's stuff [21:16] <Gergov> thanks :) ill then install ffmpeg and use the command sacarasc gave me [21:17] <JEEBsv> llogan: gentoo? they have the nice thing of being able to recompile everything that depends on libav* so you can have both ffmpeg and libav there fully [21:20] <llogan> Gergov: you can follow the compile guide or one of the recent static builds if you want ffmpeg from FFmpeg [21:20] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [21:21] <llogan> patbam: yes, you can apply options to the decoder, but maybe your file contains shorten and not pcm. [21:22] <patbam> llogan: ah that sounds likely. do you happen to know know what shorten is? [21:23] <llogan> yet another (dead?) lossless audio format [21:23] <llogan> *losslessly compressed [21:28] <llogan> patbam: just guessing here. maybe try: ffmpeg -ar 8000 -ac 2 -f shn -i input [21:29] <patbam> llogan: hmm cool, it failed in a different way, that's progress :) http://pastie.org/6021755 [21:30] <llogan> i have no more ideas without a sample to monkey with [21:32] <patbam> ok llogan, thanks!! [21:33] <octe> i'm trying to follow this: http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20capture%20a%20lightning%20(th… to get a delayed stream [21:33] <octe> but the second instance just says udp://127.0.0.1:5678: could not find codec parameters [21:46] <llogan> burek made that guide, so maybe he'll see your output too [23:30] <nilsge> anyone used ffmpeg with jack input and got rid of the thousands of xruns? [00:00] --- Sat Feb 2 2013

1 0

ffmpeg-devel.log.20130131
by burek 01 Feb '13

01 Feb '13

[00:56] <cone-913> ffmpeg.git 03Piotr Bandurski 07master:9c50e69385e8: loco: fix rgba on big-endian [00:57] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:9df9420dea0f: interplayvideo: Free previous frames on resolution changes. [01:30] <shahriman> guys can anyone download the original source from here, i want to report a bug but to verify you need the source https://vimeo.com/58025358 [02:29] <Compn> shahriman : should be downloadable with youtube-dl or [02:43] <cone-913> ffmpeg.git 03Mirjana Vulin 07master:2b6a8187a6ae: mips: optimization for float aac decoder (core module) [03:53] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:fe6767f849d9: asfdec: fix integer overflow in packet_replic_size check [04:36] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:cdf0877bc341: h264/cabac: check loop index [04:39] <Compn> nevcairiel : sorry if i missed your answer, are you planning to review dxva patch ? [04:49] <highgod> Hi, all, I want to ask a question. Does ffmpeg has any filter to remove mosaic? Thanks [04:51] <Compn> uncensor filter? no , it does not [04:52] <Compn> that would be neat :) [04:52] <Compn> nevcairiel : highgod would like it if someone would review his dxva patch :) [06:06] <highgod> Thanks [07:22] <nevcairiel> Compn: i volunteer you then! [11:25] <durandal_1707> divVerent: you did anything with showspectrum? [11:28] <divVerent> not yet, can I see the code you currently have? [11:28] <divVerent> i.e. where is i [11:29] <divVerent> t [11:29] <durandal_1707> divVerent: you are not subscribed to ffmpeg-devel? [11:29] <divVerent> no, I look in the archives [11:29] <divVerent> ok then [11:30] <durandal_1707> well it should be in archives [11:30] <durandal_1707> search for showspectrum [11:30] <divVerent> http://ffmpeg.org/pipermail/ffmpeg-devel/2013-January/138172.html this is it probably [11:30] <durandal_1707> there is only one patch that adds new funcionality .... [11:31] <divVerent> right, this one keeps the old code there for compat though [11:33] <durandal_1707> well old code use same row for all channels [11:33] <durandal_1707> compat could be renamed to something better ... [11:33] <durandal_1707> if I know nice way to map 8 ch values into 24bit number [11:35] <divVerent> 24bit number? as in RGB? [11:37] <durandal_1707> yes, rgb24 [11:40] <divVerent> the way I'd prefer is defining channels by hue+ [11:40] <divVerent> it seems to be the best one can do, albeit far from perfect [11:42] <divVerent> durandal_1707: are there already HSV or HSL functions in ffmpeg? [11:43] <divVerent> hm... there is a "hue" filter [11:44] <divVerent> ah, this filter uses unit vectors in U,V space, scaled by Y... so I will try the same and see how it looks [11:48] <durandal_1707> divVerent: it uses rgb24 for output, but that can be changed to yuv444p [11:48] <divVerent> but then it's planar... I wanted to keep the code as close as possible to the original :P [11:48] <divVerent> too bad there is no packed yuv444 format yet [11:49] <divVerent> so I will do RGB for now [11:49] <durandal_1707> what original? [11:49] <durandal_1707> output format can be changed [11:49] <divVerent> the "compat" code :P [11:49] <durandal_1707> compat mode if legacy just for same row [11:49] <divVerent> basically, what I am doing, is adding a "combined" mode to your patch that is SORT of like compat [11:49] <divVerent> but supports multiple channels [11:50] <durandal_1707> there is no point to keep if for some reason like ffplay useless code [11:50] <durandal_1707> showspectrum is just port of ffplay one night hack session [11:51] <durandal_1707> there is no reason to keep such pointless legacy crude whatever code [11:51] <durandal_1707> it is just insane [11:51] <divVerent> sure [11:51] <durandal_1707> also ffplay when outputs rgb24 it fucks up color, everything is too bright [11:51] <divVerent> compiling now [11:52] <divVerent> hehe [11:52] <divVerent> well, trying out the concept on rgb24 now, if it works out, I'll change it to yuv444p [11:52] <divVerent> and then multiple channels will get equal brightness again [11:53] <durandal_1707> i have bunch of ideas to add to this filter to make it actuall useful [11:53] <divVerent> durandal_1707: another thing to notice BTW... the current (and old) code is quite slow... due to working by rows... I wonder if working by columns, and using existing SSE-optimized rotate code after it (if it exists) would be faster [11:54] <durandal_1707> i do not think rotation is needed ... [11:54] <divVerent> but OTOH [11:54] <divVerent> performance is probably ok anyway [11:54] <durandal_1707> code is slow perhaps it do not use something of x^2 +1 [11:54] <durandal_1707> or it is just writtend without performance in mind [11:55] <divVerent> the latter [11:55] <divVerent> probably not needed, it's just processing audio after all [11:55] <durandal_1707> though ffplay output if faster, so port is maybe culprit [11:56] <divVerent> hehe, the old code may indeed work faster, depending on how clever SDL_FillRect and SDL_UpdateRect are [12:14] <cone-469> ffmpeg.git 03Stefano Sabatini 07master:d99bc8475232: doc/eval: fix/extend documentation for root() function [12:14] <cone-469> ffmpeg.git 03Stefano Sabatini 07master:41e5e28daf0a: doc/eval: fix/extend documentation for taylor() function [12:14] <cone-469> ffmpeg.git 03Stefano Sabatini 07master:dcfbe1e0b67b: doc/eval: sort functions by name [12:15] <divVerent> durandal_1707: http://rm.sudo.rm-f.org/img/uploaded/b86c263c5a2942382417adf786567173.png http://rm.sudo.rm-f.org/img/uploaded/b07aa627f01447270acc377d9af60b02.png [12:15] <divVerent> is basically my idea [12:15] <divVerent> the views are the same, the combined one is just the colors of the two added [12:16] <divVerent> the colors themselves don't matter yet though, but they are chosen to be so they always all add up to grey [12:17] <divVerent> https://gist.github.com/4682205 - is my code at the moment [12:17] <divVerent> I suppose the new "combined" mode is close enough to the old "compat" so "compat" can be ignored and colorspace can be changed? [12:19] <durandal_1707> divVerent: fine. i just want to see >2 in combined mode [12:20] <divVerent> have no input file with more than 2 channels here, can you point me to one? [12:21] <divVerent> could always generate one though, but it looks more useful with real data [12:21] <durandal_1707> samples.ffmpeg.org/ lossless audio and wavpack ... [12:22] <divVerent> ah, thanks [12:24] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/0030d8c1a047a74683c5342464beac4f.png [12:24] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/5adb3378906a2b8b4c08bd3174e82ff6.png [12:25] <divVerent> 6 channel ID in the two modes (still RGB) [12:26] <durandal_1707> hmm you render one after another? [12:27] <divVerent> this audio file does that [12:27] <durandal_1707> 2nd non-combined one looks buggy [12:27] <divVerent> http://samples.ffmpeg.org/multichannel/6_Channel_ID.flac [12:28] <divVerent> durandal_1707: no, but the 6th channel is used weird [12:28] <durandal_1707> ahh then pick less retarded sample [12:28] <divVerent> yes [12:28] <divVerent> but first porting to YUV [12:28] <durandal_1707> eva something [12:28] <divVerent> so the color brightnesses will match [12:28] <divVerent> is there a file list of this whole server? [12:29] <divVerent> oh, stupid question, found it [12:29] <durandal_1707> yes each separate color for channel is good idea for separate option .... [12:29] <divVerent> ./A-codecs/lossless/wavpack/test_suite/num_channels/eva_2.22_6.1_16bit.wv [12:29] <divVerent> AH, should have looked in A-codecs/ [12:29] <durandal_1707> yes i have that sample for testing .... [12:32] <divVerent> haha, got integer overflows ;) [12:33] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/ca583f6309bf0323fef2e786752371bc.png separate [12:34] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/e42bb921dc915f39c369929c1e7bd63e.png combined (and divided by number of channels) [12:34] <durandal_1707> some kind of projecting X dimenstion to 2.3 is needed [12:34] <divVerent> 2.3? [12:34] <divVerent> you mean the different resolution due to different FFT sizes? [12:35] <durandal_1707> i mean best way to pick color from 8 chans [12:36] <divVerent> what I currently do, is YUV = 1, sin(angle), cos(angle) [12:36] <divVerent> oops, no, first cos, then sin [12:36] <divVerent> where angle is 2pi * channel / num_channels [12:36] <divVerent> that kind of alignment ensures that if all channels output the very same, you get grey in the sum [12:37] <durandal_1707> nice, that could be another option, where other one would be old boring monotonic for all channels.... [12:37] <divVerent> the combined one IS the old monotonic one :P [12:37] <divVerent> but the separate-channel one can easily be colored or uncolored [12:38] <durandal_1707> i mean in separate mode [12:38] <divVerent> now I am wondering... you are good in maths? [12:38] <divVerent> I know there is a mathematical theorem about the max of weighted sums of unit roots [12:38] <durandal_1707> and for separate mode i want something like sox does [12:38] <divVerent> yes, separate mode should IMHO have the choice between intensity-colors (sox) and channel-colors [12:39] <divVerent> intensity-colors being default [12:40] <cone-469> ffmpeg.git 03Giorgio Vazzana 07master:ff23b76899b2: lavd/v4l2: add list_standards option [12:40] <divVerent> basically, I need max real sum(i=0..n-1) omega^i * a_i, where 0 <= a_i <= 1 and omega = a complex unit root of order n [12:40] <divVerent> I THINK there is an easy expression for this [12:42] <divVerent> it SEEMS to be in the order of n/3 [13:05] <divVerent> ARGH, I HATE YOU, TV LEVELS, I HATE YOU [13:06] <divVerent> what is the proper YUV black level again [13:06] <wm4> can't you just output full range? [13:07] <divVerent> I currently do [13:08] <divVerent> and that means lower frequencies become all black [13:08] <durandal_1707> debends what color range you set or application you use think you are using.... [13:08] <divVerent> because converting to RGB doesn't assume full range [13:08] <divVerent> or can the filter specify it outputs full range? [13:08] <wm4> I'd expect that it can set it in the picture that is output [13:08] <durandal_1707> color_range is part of avctx [13:09] <durandal_1707> i dont think lavfi is aware of it at all [13:09] <durandal_1707> try YUVJ444P pix fmt [13:10] <durandal_1707> it is 2013 and there is still no proper way to handle it in code :( [13:11] <divVerent> yes [13:11] <divVerent> it's deprecated [13:11] <divVerent> but this works [13:11] <durandal_1707> swscale have some code that converts between diferent matrix crap [13:12] <divVerent> now wondering how to set the colors so it looks good [13:12] <divVerent> or rather, maybe should match the old code [13:12] <wm4> durandal_1707: swscale + levels works, for YUV [13:12] <divVerent> it used FF0080 and 00FF80 [13:12] <wm4> at least in some cases... [13:12] <divVerent> now what are these colors in YUV [13:13] <divVerent> if my CAS is right, these colors are roughly +U and -U [13:14] <durandal_1707> divVerent: you can find several nice distributions and make them configurable [13:14] <wm4> lol that ticket [13:14] <durandal_1707> Frodo lost his ring [13:21] <divVerent> durandal_1707: there is not many to pick from [13:21] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/c45edab2126015b59ff9aa6665d55427.png http://rm.sudo.rm-f.org/img/uploaded/ce6512b22561426517b5bb5728624d00.png [13:21] <divVerent> my first attempt to match the default colors of ffplay [13:22] <divVerent> could it be that the two filters use different windowing functions? [13:23] <divVerent> yes, indeed [13:23] <divVerent> your code with compat=1 also has less grey stuff around :P [13:23] <durandal_1707> why would they use different function? [13:23] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/e929fce848e6f0d09f841bad9183da10.png [13:23] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/c45edab2126015b59ff9aa6665d55427.png [13:23] <divVerent> first is compat=1 [13:23] <divVerent> second is my code [13:23] <divVerent> and the one before was ffplay [13:24] <divVerent> ffplay basically does SOME part of the RDFT differently [13:24] <divVerent> most likely the window function [13:25] <divVerent> ffplay uses a 1-x^2 window [13:25] <ubitux> yes i used a different windowing function [13:25] <durandal_1707> does it make sense? [13:25] <ubitux> (hello!) [13:25] <saste> ehi ubitux-san! [13:25] <divVerent> the avfilter function is better [13:25] <durandal_1707> x^2 + 1 should be used [13:25] <ubitux> the windowing function wasn't sexy so i used a hann one iirc [13:25] <divVerent> durandal_1707: sure not + 1 :P [13:25] <divVerent> yes [13:25] <divVerent> the issue you see on my screenshots :P [13:25] <divVerent> I tested with aevalsrc [13:26] <divVerent> and it's pure sine waves [13:26] <divVerent> ffplay shows LOTS of grey stuff around it [13:26] <divVerent> due to bad window function [13:26] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/ce6512b22561426517b5bb5728624d00.png current ffplay [13:26] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/e929fce848e6f0d09f841bad9183da10.png current showspectrum [13:26] <durandal_1707> ok, i prefer correct code [13:26] <divVerent> none is correct, but the better window function is less incorrect ;) [13:27] <durandal_1707> i'm against legacy [13:27] <divVerent> there is no way to do this correctly [13:27] <divVerent> but it's pretty good now [13:27] <durandal_1707> whatever, i just cant use current showspectrum as it is useless to me [13:27] <durandal_1707> so either I gonna improve it [13:27] <durandal_1707> or write new filter [13:29] <durandal_1707> so i will apply modified patch of mine for showspectrum, as it got no attention at all [13:29] <durandal_1707> and then will apply divVerent stuff [13:29] <divVerent> you can have my patch now BTW [13:29] <divVerent> what did you modify BTW, as I probably replaced all changed code by now [13:30] <divVerent> https://gist.github.com/4682536 [13:30] <divVerent> is my current changes vs yours [13:30] <divVerent> now all it needs is the intensity-coloring mode [13:30] <divVerent> (the "sox mode") [13:31] <durandal_1707> and then "rotated" mode, where slide goes up<->down [13:32] <durandal_1707> dont stop there, lets your imagination work out..... [13:32] <divVerent> I doubt it'd make sense to have both these modes... it'd duplicate too much code then probably [13:32] <divVerent> maybe better to decide on one mode, and use rotate filter for the other [13:32] <divVerent> but, rotated vs current display would likely be faster [13:33] <divVerent> e.g. that memmove mess can be all done as a single block then [13:33] <durandal_1707> duplicate? cant you just use functions? [13:34] <durandal_1707> yes, making bad code looks good is boring and daunting task.... [13:35] <divVerent> actually... I think I will solve that inefficiency now, which would also help with rotating [13:35] <divVerent> I will use an intermediate float buffer for each line [13:36] <divVerent> that way, you can in the regular display also bump saturation to like 100x [13:36] <divVerent> and then see channel differences better [13:36] <divVerent> in the current code, the clamping to 0..255 at each step would screw that up [13:39] <ubitux> saste: hey :) [13:39] <ubitux> do you miss me? [13:39] <cone-469> ffmpeg.git 03Paul B Mahol 07master:b70ea49ca32c: Port biquads filters from SoX [13:40] <ubitux> durandal_1707: bitexact + fate tests now? ;) [13:40] <durandal_1707> bitexact what? [13:40] <saste> ubitux: i'm feeling a bit annoyed since i have no new subtitles to try since a while [13:40] <ubitux> durandal_1707: integer arithmetic [13:41] <ubitux> saste: no new subtitles patches? [13:41] <durandal_1707> i still wait for that kerndeint improvements... [13:41] <saste> ah yes and that... [13:41] <durandal_1707> ubitux: i still do not get it what it means [13:41] <wm4> hey guys I could need a stable and useful API for lavfi [13:41] <ubitux> saste: nag beastd, he should be doing this [13:41] <durandal_1707> wm4: ask michaelni [13:41] <saste> beastd: I nag you! [13:42] <saste> wm4, you ask too much [13:42] <durandal_1707> ubitux: filters are bixexact with sox if you use aformat=samplefmts=dblp,bass...... [13:42] <saste> what about, unstable and useless? [13:42] <ubitux> durandal_1707: forget this, i though it wasn't supporting s16 etc [13:43] <wm4> e.g. recently I saw a commit that fixes seeking with lavfi and ff/avplay [13:43] <wm4> the solution was recreating the whole filter chain on seek [13:43] Action: ubitux doesn't want to leave this country [13:43] <ubitux> any ffmpeg work in japan for me? :( [13:43] <wm4> is that the proper lavfi solution? [13:49] <divVerent> durandal_1707: ok... https://gist.github.com/4682630 [13:50] <divVerent> now there's not much more of your patch left, so I diffed to master instead [13:50] <divVerent> local copy has the missing freep added [13:50] <divVerent> durandal_1707: the combined mode is now cool with like saturation=5 [13:53] <divVerent> so next part would be research as for how sox mode would work [13:53] <durandal_1707> do you have branch with several commits, it it makes sense to split them in smaller funcionality steps.. [13:54] <divVerent> no, don't have that [13:54] <divVerent> given I don't see many separate steps there [13:54] <durandal_1707> divVerent: you mean colors or output single frame for whole duration? [13:54] <divVerent> colors [13:54] <divVerent> single frame for whole duration can already be done, if calculating the width "right" ;) [13:55] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/257d3a6e6917db49fcf736ae9467da60.png with high saturation [13:55] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/ec4ffa6febfc860cbc9aaca340237a01.png - combined mode with high saturation [13:56] <divVerent> I can visually make out 4 channels there [13:58] <durandal_1707> using ch to loop over data planes confused me big time [13:58] <divVerent> I saw it as color channels [13:58] <divVerent> but the main reason was that p for plane was taken [13:58] <divVerent> by p for pointer ;) [13:58] <durandal_1707> :) [13:58] <divVerent> but sure, adding a plane var [14:00] <divVerent> do we use C99 BTW? [14:01] <durandal_1707> silly, but no [14:01] <wm4> ffmpeg requires C99 though [14:02] <divVerent> wm4: it does? [14:02] <divVerent> was mainly asking if I can replace this sqrt(re*re+im*im) by hypot(re,im) [14:02] <divVerent> ;) [14:02] <divVerent> hypot() is C99 though, and MS renamed it to _hypot [14:02] <wm4> divVerent: yes, why else do you think they wrote a c99-to-c89 converter just for libav? [14:02] <divVerent> hypot() is not C89, but existed before C99 on many systems [14:03] <divVerent> e.g. virtually any BSD [14:03] <divVerent> and glibc [14:03] <durandal_1707> i was refering to for (int plane ...) { ... [14:03] <divVerent> durandal_1707: hehe, yes... [14:03] <divVerent> I mainly am asking about hypot() [14:03] <divVerent> as I am now cleaning up the ugly macros [14:04] <durandal_1707> just send patch(es) to ffmpeg-devel when ready [14:04] <divVerent> based on master, I suppose? [14:05] <durandal_1707> yup [14:05] <divVerent> well, I mainly want to know before I do that [14:05] <divVerent> +#define CPLX_MAGNITUDE(re, im) sqrt((re)*(re) + (im)*(im)) /* is actually hypot(re, im) */ [14:05] <divVerent> do we NEED this macro at all then ;) [14:05] <divVerent> hypot() is not on every pre-C99 libc, but on most [14:05] <divVerent> MS is the exception, MS deprecated hypot() in favor of _hypot() in 2005 [14:05] <divVerent> so it'll warn in MSVC [14:05] <divVerent> but C99 added hypot() [14:05] <durandal_1707> do it in separate commit and other who carse will define hypot to something useful [14:06] <durandal_1707> for arch where it is not present... [14:06] <durandal_1707> *care [14:06] <divVerent> it probably exists on all [14:06] <divVerent> just warns on MSVC [14:06] <divVerent> just like fopen() ;) [14:07] <durandal_1707> dont worry, MSVC guys fill provide fix as soon something break.... [14:07] <divVerent> also, c99wrap SURE already fixes this [14:07] <divVerent> given this one can even be fixed by -Dhypot=_hypot [14:08] <divVerent> so hardest part now is cleaning up the history :P [14:08] <durandal_1707> is there someone on #ffmpeg that claims native aac encoder is good? [14:10] <saste> wm4: the "clean" solution, i guess, would be to implement a seek command in the various filters [14:10] <saste> then you simply send a seek to the relevant filters [14:11] <wm4> saste: I did that in mpv [14:11] <durandal_1707> and what seek would do in each filter? [14:11] <saste> we could even script the key-binding, so each user can choose her favored keystrokes [14:11] <wm4> durandal_1707: reset state [14:11] <saste> durandal_1707, it depends [14:11] <saste> some filters don't need to be reset (e.g. hflip, vflip, etc.) [14:11] <saste> movie could actually seek in the input [14:11] <wm4> though I actually wonder if there any filters which really benefit from a seek command, instead of recreating the graph [14:12] <wm4> my problem is more that recreating the graph is a complicated operation [14:12] <saste> wm4, ffplay_input could be changed to support a seek operation [14:13] <saste> again some scripting seems the right solution, so you can change/program the binding at the application level [14:13] <wm4> also, I've said it before, but string commands suck [14:13] <wm4> for both caller and filter implementations [14:16] <saste> wm4, willing to help in that area? [14:17] <wm4> saste: possibly... [14:18] <saste> wm4, first step would be to make all the filters parse the argument string with av_opt_ [14:18] <saste> at that point we can switch and factorize initialization [14:19] <saste> i don't think there are many filters left [14:19] <saste> then we could provide a secondary init() interface, which accepts options as an AVDictionary or whatever [14:20] <cone-469> ffmpeg.git 03Paul B Mahol 07master:402ea625ab7b: lavfi/biquads: remove pointless casts [14:20] <saste> this could simplify scripting, since you would need no more escaping hell since each option can be specified in a separate argument [14:21] <saste> (or you could just set options on the filter) [14:27] <saste> is ffmpeg going to compete with sox? [14:28] <saste> otoh sox doesn't seem to be very active [14:29] <wm4> can't ffmpeg just use sox instead? [14:29] <saste> wm4, yes [14:29] <saste> and ladspa as well [14:32] <durandal_1707> ffmpeg biquads are faster than sox ones (planar<->interleaved) sox is missing optimizations [14:45] <Compn> someone (carl !! you there? :) want to retest if https://ffmpeg.org/trac/ffmpeg/ticket/1957 is fixed now? [14:56] <durandal_1707> Compn: it is duplicate of ticked which is fixed [15:05] <durandal_1707> divVerent: i like your patch [15:06] <divVerent> will soon post it [15:07] <divVerent> I divided it back up into a few commits, but the big change of how your method works can't really be separated out [15:07] <divVerent> also, did I mention that I hate that YUV444P is not actually YUV444P but YVU444P? ;) [15:12] <Compn> durandal_1707 : yes, but is it fixed or is it a mis-duplicate tag ? :P [15:15] Action: Compn going to upset carl by reopening random bugs and turning them into feature requests :D [15:32] <saste> Compn, we should tag tickets with a difficulty level [15:32] <saste> so newbee contributors can pick the easy ones [15:34] <cone-469> ffmpeg.git 03Giorgio Vazzana 07master:514216d8a928: lavd/v4l2: read the correct time per frame from devices that support a standard [15:34] <cone-469> ffmpeg.git 03Stefano Sabatini 07master:a842dc632eca: doc/indevs: apply misc fixes to the v4l2 documentation [15:35] <durandal_1707> saste: can that be added to trac? [15:35] <saste> durandal_1707, sure [15:35] <saste> vlc trac has it at least [15:35] <saste> a tag cloud would be fun as well [15:36] <durandal_1707> than lets add it [15:36] <saste> who's the designated maintainer for trac? [15:41] <Compn> no idea [15:41] <Compn> saste : good idea re difficulty level [15:45] <Compn> i dont think i'm an admin on trac. [15:45] <Compn> probably michaelni [16:00] <durandal_1707> hmm amovie cant use http:// .... ? [16:01] <wm4> durandal_1707: AFAIK that should work? [16:02] <durandal_1707> amovie cant use http:// .... ? [16:03] <ubitux> backquote the ':' [16:03] <ubitux> it's an option separator [16:03] <wm4> d'oh [16:11] <Compn> that info in the docs? [16:11] <Compn> example with http url be nice :) [16:11] Action: Compn gone [16:12] <durandal_1707> what i should put before : ? [16:14] <durandal_1707> i tried / and ` [16:14] <ubitux> \ obviously [16:15] <durandal_1707> it does not work [16:15] <ubitux> the \ must be present, is your shell interpreting it? [16:15] <durandal_1707> \\ works [16:16] <ubitux> yes because otherwise the shell "escape" the :, basically removes the \ [16:16] <durandal_1707> was waste of time, lavfi cant handle mp3 stream [16:18] <saste> durandal_1707, no? [16:19] <durandal_1707> i get spam: [16:19] <durandal_1707> [mp3 @ 0x29878400] Header missing [16:19] <durandal_1707> [Parsed_amovie_0 @ 0x29809140] Decode error: Invalid data found when processing input [16:20] <durandal_1707> even if i download such file [16:42] <saste> Maximum number of registered filters 128 reached, [16:42] <saste> uh... [16:44] <durandal_1707> lol, really? [16:46] <durandal_1707> indeed [16:46] <durandal_1707> saste: how you trigered it? [16:46] <saste> durandal_1707, i just updated my ffmpeg [16:47] <saste> it should be that i enabled more filters than you [16:47] <durandal_1707> exactly [16:47] <saste> patch to raise the max number incoming [16:47] <durandal_1707> isn't it part of ABI/API , we need to bump major [16:48] <durandal_1707> lavf does not have such limit? [16:48] <saste> durandal_1707, no, it's internal [16:49] <durandal_1707> it should raise on its own, so no need to increase number each time limit is reached [16:51] <durandal_1707> i want to do this with ffmpeg: sox -n o.sox synth pl G2 pl B2 pl D3 pl G3 pl D4 pl G4 delay 0 .05 .1 .15 .2 .25 remix - fade 0 4 .1 [16:53] <saste> yes it's weird, i don't remember why we did filter registration that way [16:56] <saste> durandal_1707, can't reproduce the mp3 issue with amovie [16:57] <cone-469> ffmpeg.git 03Stefano Sabatini 07master:e47114d5459a: lavfi: increment max number of registered filters from 128 to 256 [17:01] <divVerent> durandal_1707: http://rm.sudo.rm-f.org/img/uploaded/9db2af3387cdae336f6d831702da5f6e.png [17:01] <divVerent> this is with stolen code from sox though ;) [17:03] <durandal_1707> this is palette or? [17:03] <divVerent> nope [17:03] <divVerent> https://gist.github.com/4683934 [17:03] <divVerent> this is the code [17:03] <divVerent> stolen from sox's palette generation [17:04] <durandal_1707> good, it would be nice to make it generic so other shemes can be added... [17:04] <divVerent> I don't like this though [17:04] <divVerent> it's slow, and nasty (YUV conversion...) [17:04] <divVerent> ideally I'd like to replicate the same curves but in YUV space [17:05] <divVerent> or rather, very similar ones [17:05] <divVerent> I tried to do it, but yellow became pink [17:05] <divVerent> rest works [17:05] <divVerent> wonder if I should just make it an array of (value, y, u, v) items [17:05] <divVerent> and it'd linearily interpolate [17:06] <divVerent> and I'd predefine it for sox's colors at all "critical" points [17:06] <divVerent> i.e. all intensity values mentioned there, and 0.30 (middle of the sin() curve for b) [17:06] <durandal_1707> well having sox as some kind of palette is extra feature [17:07] <divVerent> also, as you see, i also made logarithmic scale for this one [17:07] <durandal_1707> it uses 256 colors only, which i find too low [17:08] <durandal_1707> but it can be used in sox mode, if one wants to have same output as sox does.... [17:09] <divVerent> well, I rather would prefer it to be a table with linear interpolation [17:10] <divVerent> as that CAN be more accurate in the end [17:12] <durandal_1707> sure [17:17] <divVerent> yes, works [17:23] <cone-469> ffmpeg.git 03Paul B Mahol 07master:22fad4e2b8c7: doc/filters: fix afade cuve types listing [17:24] <divVerent> durandal_1707: http://rm.sudo.rm-f.org/img/uploaded/180081251f2e54275685c44b505b66bb.png sox colors with square root scale (old ffplay) [17:24] <divVerent> http://rm.sudo.rm-f.org/img/uploaded/d7fd80be0fd8e92c8f7941f12a87dafb.png sox colors with logarithmic scale (sox) [17:24] <divVerent> both look useful, actually [17:28] <durandal_1707> divVerent: yes [17:31] <durandal_1707> shit, typo [17:31] <durandal_1707> i hate typos [17:39] <divVerent> https://gist.github.com/4684199 for you to test, will post it on the ML later [17:40] <divVerent> I recommend the combinations: [17:40] <divVerent> - showspectrum (separate, otherwise ffplay-like) [17:40] <divVerent> - showspectrum=combined=1 (ffplay replication) [17:40] <divVerent> - showspectrum=intensity=1:logscale=1 (sox) [17:40] <divVerent> - showspectrum=intensity=1 (soxlike but ffplay's scale) [17:41] <divVerent> - showspectrum=saturation=0 (what your original code did) [17:42] <divVerent> saturation=-1 yields an alternate color scheme ;) [17:42] <durandal_1707> my original code was just to not ignore other channels .... [17:42] <divVerent> yes [17:42] <divVerent> I mainly like using saturation in combined mode [17:42] <divVerent> to exaggerate channel differences [17:43] <durandal_1707> does this one still depends on old patch of mine? [17:43] <divVerent> the series includes it [17:43] <divVerent> but I could squash the first two, but didn't to keep your credit [17:43] <durandal_1707> it is splitted? [17:44] <divVerent> my patch has been split, but the first change that basically rewrote yours is not split [17:44] <durandal_1707> divVerent: i dont care for my credits [17:44] <divVerent> then I'd squash the first two [17:47] <divVerent> https://gist.github.com/4684260 would be the same with the first two changes squashed and explained as a single commit [17:48] <divVerent> but for finally committing it, it probably would be best to have all changes in a single commit in the end [17:48] <divVerent> for reviewing it may be better this way though [17:50] <durandal_1707> i'm fine with whatever combination [17:51] <divVerent> so am I... will play around with it a bit more, but probably won't change it much any more [17:51] <divVerent> as for the one-image-output feature, not sure... I also want to look into the Y scale though [17:52] <divVerent> as in, where it comes from [17:52] <divVerent> whether top is nyquist, or something else [17:53] <divVerent> it SHOULD be nyquist ;) but I don't trust the code regarding that yet [18:04] <cone-469> ffmpeg.git 03Stefano Sabatini 07master:7e0fc1a24fa6: CREDITS: redirect to Git log, remove current outdated content [18:09] <divVerent> yes indeed... nyquist is ONLY at the top when the height is a power of two (if not combined, multiplied by channel count) [18:09] <divVerent> does FFT size HAVE to be a power of two? if yes, it's difficult to fix [18:10] <divVerent> compare output with s=512x256 to s=512x257 ;) [18:10] <durandal_1707> divVerent: first commit/ my code have some flaws, see av_size_mult() - this is supposed to check for overflows [18:10] <durandal_1707> but i did not updated it ... [18:12] <durandal_1707> also you should use av_realloc_f or otherwise leaks happens when memory alloc fails [18:15] <divVerent> ok, committed fix for these two locally [18:16] <divVerent> what do you suggest about the nyquist issue? [18:16] <divVerent> the thing is, APPARENTLY only power-of-two FFT sizes are supported [18:17] <divVerent> so if you have (evil case) 7 channels [18:17] <divVerent> you only see the full frequency range if the output size is 7*poweroftwo [18:17] <divVerent> issue existed before too, but then only applied to the full size [18:17] <divVerent> default height of 480 was bad for the same reason :P [18:18] <divVerent> ignore it, like before - but set the output height to 512 so by default, you do see the full spectrum for 1ch and 2ch at least? [18:19] <divVerent> the only alternative would be using a scaling filter to trim the size down [18:19] <divVerent> I'd rather not [19:17] <cone-469> ffmpeg.git 03Ronald S. Bultje 07master:a5ba798c16d0: dsputil: remove unused functions copy_block{2, 4, 8, 16}. [19:17] <cone-469> ffmpeg.git 03Michael Niedermayer 07master:ed341549bbc2: Merge commit 'a5ba798c16d0614d982a76755fdd72b37d437170' [19:27] <cone-469> ffmpeg.git 03Martin Storsjö 07master:a65f965c04bf: mpegvideo: Do REBASE_PICTURE with byte pointers [19:27] <cone-469> ffmpeg.git 03Michael Niedermayer 07master:324498f82470: Merge commit 'a65f965c04bfa27adedc0409c14cc05903f483d0' [19:29] <saste> what happened to the ffmpeg -i option? [19:30] <michaelni> saste, ? [19:31] <saste> michaelni, I'm trying to set -user_agent, and failing [19:31] <saste> also I can't see the "i" entry in the options array (ffmpeg_opt.h) no more [19:31] <saste> ffmpeg_opt.c [19:31] <michaelni> the parsing code was redesigned a few times [19:32] <michaelni> its more convoluted now [19:33] <saste> so now is more "evil" [19:34] <saste> but, it seems I can't set avio options [19:34] <saste> is that a regression or what? [19:34] <michaelni> did it work before ? [19:35] <saste> dunno, but I'm trying an example: ffplay -user-agent foobar URL and it seems to fail with unknown option [19:35] <saste> the example was taken from the docs, so I thought about a regression [19:37] <saste> but maybe it was never possible [19:37] <saste> wm4, how can you reproduce the http cookies issue? [19:39] <saste> in other words, how can I pass protocol options to avformat_open_input()? [19:39] <saste> is that possible? [19:42] <saste> wow, why all these MP3 hacks in avformat_open_input()? [19:42] <michaelni> btw did you try -user-agent or user_agent ? [19:43] <michaelni> ffmpeg -h full has it with - not _ [19:49] <saste> michaelni, i'm debugging the issue [19:52] <saste> it is setting the option, but then for some "evil" reason it fails later in ffmpeg_opt.c code [20:02] <cone-469> ffmpeg.git 03Diego Biurrun 07master:0f5b0b4178b8: avisynth: Change demuxer name to avoid conflicts with AVS [20:02] <cone-469> ffmpeg.git 03Diego Biurrun 07master:29f1fa74225e: configure: Move newlib libc check before mingw libc check [20:02] <cone-469> ffmpeg.git 03Michael Niedermayer 07master:01d012ba3db9: Merge commit '29f1fa74225e9fdbab6d590857edbecb51cf74da' [20:08] <cone-469> ffmpeg.git 03Diego Biurrun 07master:197252f1c58b: configure: Add a comment indicating why uclibc is checked before glibc [20:08] <cone-469> ffmpeg.git 03Diego Biurrun 07master:52acd79165e7: x86: hpel: Move {avg,put}_pixels16_sse2 to hpeldsp [20:08] <cone-469> ffmpeg.git 03Michael Niedermayer 07master:dd87d4a318b3: Merge remote-tracking branch 'qatar/master' [20:20] <saste> michaelni, ffplay -user-agent foobar http://ffmpeg.org/ffmpeg-logo.png [20:24] <saste> oh wait it's a bit of a mess [20:25] <saste> lavf see that the url is an image file, so it doesn't open avio [20:25] <saste> so it doesn't consume the option -> error [20:26] <saste> so i can't see no way to pass the url options here [20:26] <saste> also avformat_open_input() has no way to discriminate avio options from demuxer options [21:00] <Daemon404> does lavfi have a way to generate blank input easily? [21:00] <Daemon404> i.e. i need somthing good to test encoding threading [21:11] <llogan> Daemon404: what do you mean by blank input? [21:11] <Daemon404> llogan, i used /dev/zero. nevermind. [21:12] <saste_> Daemon404, color=c=bikeshed [21:12] <llogan> and "aevalsrc=0" [21:15] <cone-469> ffmpeg.git 03Nicolas George 07master:6560625f0aec: lavfi/bufferqueue: add ff_bufqueue_is_full(). [21:15] <cone-469> ffmpeg.git 03Nicolas George 07master:ae14887ee7f2: lavfi/af_amerge: check for buffer queue overflows. [21:15] <cone-469> ffmpeg.git 03Nicolas George 07master:dcce43340f45: lavfi/af_amerge: set outbuf->audio->channels. [21:15] <cone-469> ffmpeg.git 03Nicolas George 07master:d2ccab5f8a5f: lavfi/buffersrc: forward filter_frame errors. [21:23] Action: Compn wonders why saste has a hardcoded max filter number [21:23] <Compn> its like hardcoding a max codec number. a foolish thing! :P [21:30] <durandal_1707> divVerent: just send patches, and size could be set for channel height or rounded from window size so some part of window is black.... [21:31] <durandal_1707> also sox have alternative pallete, if you want to do it too.... [22:14] <durandal_1707> can somewhere help this guy who keep asking user questions in community on ffmpeg? [22:16] <saste_> durandal_1707, what guy? [22:16] <durandal_1707> g+ guy [22:17] <cone-469> ffmpeg.git 03Stefano Sabatini 07master:349e7f423f19: doc/indevs: itemize examples for the v4l2 device [22:17] <cone-469> ffmpeg.git 03Stefano Sabatini 07master:b999774f0b11: doc/indevs: document v4l2 options [22:22] <durandal_1707> so to pad start of audio you use aresample, there really should be example in doc... [22:27] <llogan> durandal_1707: link? [22:36] <durandal_1707> llogan: what link? [22:39] <llogan> nevermind. i see what you're referring to. [22:40] <durandal_1707> why are there two cone- bots? [22:41] <llogan> you already told him to get help at the mailing list. not much else to say. [23:10] <kierank> shahriman: did you see the vc-1 interlaced patch [23:10] <shahriman> no i did not. where is it? [23:10] <shahriman> kierank: ^ [23:10] <kierank> on the ml [23:11] <kierank> j-b: ^ [23:11] <shahriman> kierank: can you please give me the link? [23:11] <kierank> http://ffmpeg.org/pipermail/ffmpeg-devel/2013-January/138302.html [23:15] <shahriman> kierank: i forgot almost everything i knew about that code, but the patch looks ok to me [00:00] --- Fri Feb 1 2013

1 0

ffmpeg.log.20130131
by burek 01 Feb '13

01 Feb '13

[00:18] <ska> oops. sorry. [00:21] <sacarasc> You should be sorry! [00:21] <sacarasc> What for? [00:37] <K-Rich> Hi all [00:37] <K-Rich> Cananyone tell me why this doesn't work? ffmpeg -y -acodec pcm_s16le -f alsa -i pulse -s 640x360 -qscale 1 -r 30 -vcodec rawvideo -f video4linux2 -i /dev/video0 -f yuv4mpegpipe | tee test.avi | mplayer -cache 1024 -nosound - [00:38] <K-Rich> sorry ffmpeg -y -acodec pcm_s16le -f alsa -i pulse -s 640x360 -qscale 1 -r 30 -vcodec rawvideo -f video4linux2 -i /dev/video0 -f yuv4mpegpipe - | tee test.avi | mplayer -cache 1024 -nosound - [00:45] <K-Rich> anyone? [00:45] <K-Rich> sorry ffmpeg -y -acodec pcm_s16le -f alsa -i pulse -s 640x360 -qscale 1 -r 30 -vcodec rawvideo -f video4linux2 -i /dev/video0 - | tee test.avi | mplayer -cache 1024 -nosound - [00:45] <K-Rich> gives errors [00:47] <K-Rich> http://pastie.org/5976041 [00:57] <K-Rich> and still noone [00:57] <K-Rich> heh [01:10] <klaxa> K-Rich: two things [01:11] <klaxa> actually three [01:11] <K-Rich> ok [01:11] <klaxa> a) you are using avconv, this is the wrong channel unfortunately [01:11] <klaxa> b) apparently you are not allowed to access /dev/video0 [01:11] <klaxa> c) why do you bother to capture audio if you don't play it back with mplayer? [01:11] <K-Rich> (mind you i "just" tried to change to real ffmpeg, with this result http://pastie.org/5976323 )) [01:12] <klaxa> ah didn't see the tee pipe [01:12] <K-Rich> i want the audio captured in the file but want to see the video, if the displayed video had sound it would cause a feedback loop with the recording [01:13] <klaxa> i'm not sure, but i think when playing back raw audio (pcm_s16le) and raw video you will somehow have to tell mplayer that it is raw data [01:13] <K-Rich> also i am in the group video and 'crw-rw----+ 1 root video 81, 0 Jan 30 15:18 /dev/video0' [01:13] <klaxa> also the order of your parameters is a bit weird [01:15] <K-Rich> i'm just learning this, trying to make an applet for screen/webcam recording [01:15] <klaxa> i thnik general practice is ffmpeg -f <in_format> [in_options e.g. codec, filters] -i <in_stream> [-f container format] <out_file> [01:15] <klaxa> *think [01:17] <K-Rich> if put a filename (capture.avi) rather than '-| tee test.avi | mplayer -' it works, just doesn't display [01:18] <klaxa> ah wait... [01:18] <klaxa> right... [01:18] <klaxa> ffmpeg <all your stuff> -f avi pipe: | tee somefile.avi | mplayer - [01:19] <K-Rich> /dev/video0: Operation not permitted STILL ugh, even when i do it with sudo [01:20] <klaxa> move the -i /dev/video0 before -vcodec even [01:22] <K-Rich> http://pastie.org/5976457 [01:22] <K-Rich> same friggin access denied crap [01:22] <K-Rich> err operation not permitted [01:23] <K-Rich> hmmmmmmmm [01:24] <K-Rich> one sec BRB [01:24] <klaxa> try also adding something like -f video4linux2 before -i /dev/video0 [01:24] <K-Rich> i wonder if all the trying somehow left it open [01:24] <K-Rich> ok [01:35] <K-Rich> getting closer [01:35] <K-Rich> i can see it [01:36] <K-Rich> but the vid plays fast and the sound is empty (static only) [01:36] <K-Rich> ffmpeg -acodec pcm_s16le -f alsa -i pulse -s 640x360 -qscale 1 -r 30 -f video4linux2 -i /dev/video0 -vcodec rawvideo -f avi pipe: | tee test.avi | mplayer -cache 1024 -nosound - [01:38] <K-Rich> klaxa: you still here? [01:38] <klaxa> yeah [01:38] <klaxa> um... about the audio thing, have you selected the correct recording device in pulse? [01:39] <klaxa> with the video thing, i'm not sure rawvideo has data about framerates [01:40] <K-Rich> okay using mjpeg rather than rawvideo works sorta, way delayed and still no sound [01:41] <klaxa> open pulseaudio volume control and check you have the correct device set for the recording stream [01:45] <K-Rich> okay lol, got sound...... [01:45] <K-Rich> how do i know what encoders i have, i need one with yuv [01:45] <K-Rich> still have the delay though [01:56] <klaxa> K-Rich: ffmpeg -codecs [02:04] <K-Rich> klaxa: woo hoo (sorta) [02:04] <K-Rich> ffmpeg -acodec pcm_s16le -f alsa -i pulse -s 640x360 -qscale 1 -r 30 -f video4linux2 -i /dev/video0 -vcodec huffyuv -f avi - | tee test.avi | mplayer -nosound -cache 1024 - [02:05] <K-Rich> works great, except there is a delay on the video... any idea on that? [02:05] <klaxa> mmhh no idea actually, maybe huffyuv is straining your cpu? although it shouldn't really [02:06] <K-Rich> the only one without a delay was rawvideo, but it played at like double speed, though the audio was normal speen (the image pauses half way through) [02:15] <K-Rich> grrrrrr [02:15] <K-Rich> updatung the ffmpeg made my deskrecord script look like ass (pixilated) [02:15] <K-Rich> it was clear before [02:18] <K-Rich> fow is it a command that worked with avconv doesn't work with ffmpeg ? [02:18] <K-Rich> it's become all blocky (pixilated) [02:19] <klaxa> sounds like a quality setting issue [02:21] <K-Rich> ffmpeg -acodec pcm_s16le -f alsa -i pulse -s 1280x800 -qscale 1 -r 30 -vcodec rawvideo -f x11grab -i $DISPLAY -s 640x356 -vf pad=640:360:0:2:000000 ~/Desktop/Video_$(date +%F_%T).avi [02:23] <klaxa> hmmm that should work though [02:25] <K-Rich> it looks like it's an animated jpeg with low quality, all blocky [02:25] <K-Rich> it worked great with ubuntus avconv ffmpeg whatever [02:26] <K-Rich> Stream mapping: [02:26] <K-Rich> Stream #1:0 -> #0:0 (rawvideo -> mpeg4) [02:26] <K-Rich> Stream #0:0 -> #0:1 (pcm_s16le -> libmp3lame) [02:26] <K-Rich> mpeg4 wtf? [02:27] <K-Rich> ffmpeg version 0.10.6-6:0.10.6-0ubuntu0jon1~precise1 [02:29] <K-Rich> Mwahahahaa got it [02:29] <K-Rich> added -sameq [02:29] <K-Rich> works great now :) [02:33] <K-Rich> heh... made the other command better as well [02:33] <K-Rich> the output at least [02:33] <K-Rich> though there is still delay :/ [02:42] <K-Rich> ffmpeg -acodec pcm_s16le -f alsa -i pulse -s 640x360 -qscale 1 -sameq -r 30 -f video4linux2 -i /dev/video0 -vcodec huffyuv -f avi - | tee test.avi | ffplay -an -window_title Webcam\ Recording - [02:42] <K-Rich> hmmmmmmm still a delay :/ [05:35] <Tronic> Strange problem with my own software... Decoding of MP3 and Ogg (didn't try other formats yet) produces highly distorted sound (lots of white noise etc), but the actual audio can be heard thru. [05:36] <Tronic> Same source code has worked with libav and older ffmpeg versions. Also, ffplay, with current version, decodes audio just fine. Video decoding in my application is also working properly. [05:37] <Tronic> Any suggestions other than trying to go thru my own code and ffplay line by line, seeking for diffferences? [05:40] <Tronic> I was also afraid that different library versions might be mixing up but uninstalling everything but the latest didn't change anything. I am running current git version at the moment. [05:54] <Tronic> With WAV format audio there is no distortion. [09:34] <praveenmarkandu> hi. if a source file is already h.264 and aac and when i transcode it again and use -c:v libx264 -c:a libfaac [09:34] <praveenmarkandu> will it be slower than just using -c:v copy [09:37] <praveenmarkandu> what i actually want to do is reduce the bitrate and resolution of the video [09:37] <praveenmarkandu> will copy make a speed difference? [10:11] <cbsrobot> praveenmarkandu: copy is faster [10:11] <praveenmarkandu> cbsrobot, i just tried copy with my full command and the -vf scale didnt seem to work [10:12] <cbsrobot> yeah stream copy disables the filters [10:13] <praveenmarkandu> oh okay [10:13] <praveenmarkandu> and -s:v is the same as -vf scale [10:13] <praveenmarkandu> so it wont work either right? [10:15] <praveenmarkandu> and it seems it also disables the video bitrate setting [10:15] <praveenmarkandu> hah. okay. should have expected [11:43] <lemonjelly> Hello. With ffmpeg, is it possible to offset all timestamps in the *output* streams by a certain amount? [11:45] <lemonjelly> I've been experimenting with the -itsoffset and -copyts options but I can't get ffmpeg to behave as I need [11:47] <lemonjelly> Basically I'm trying to output mov fragments of an ism stream, but I'm using the -ss <offset> option to start partway through the input stream, and I want the first fragment I output to have a frag_start value matching <offset>. But it is always 0. [13:14] <Zeeflo> burek, are you around? You got a bug of some sort in your build [13:14] <Zeeflo> (which has just set me back 2 days!! :/) [13:35] <someone-noone> Hello! I'm getting crash(from time to time) in avcodec_decode_audio4()->aac_decode_frame()->av_packet_get_side_data() with next code: http://ideone.com/uxeYwz [13:35] <someone-noone> I can't figure out what can be a problem. Sometimes problem occur and sometimes no [13:35] <someone-noone> Any help? [13:39] <Zeeflo> Any ideas? fontconfig: Selected font is not the requested one: 'DejaVu Sans' != 'Arial'its/s dup=1 drop=0 [13:41] <Mavrik> someone-noone: it seems something goes wrong when you're poping the packet [13:41] <Mavrik> are you copying all fields? [13:42] <Mavrik> is your buffer padded with at least FF_BUFFER_PADDING (or what's it) bytes? [13:43] <someone-noone> Mavrik, one momen [13:44] <someone-noone> Mavrik, http://ideone.com/VBliwr this is how I'm reading packets. I'm porting my code from old version of libav. [13:45] <someone-noone> Mavrik, and buffer& you're talking about AVFrame's one? [13:53] <ledil> Hello, Ive installed ffmpeg from jon-severinsson ubuntu ppa repository, but now Im getting "Unknown encoder 'libfaac'". libfaac-dev and libfaac-0 are installed ... any hints ? [13:54] <ledil> this is my version: 0.10.6-6:0.10.6-0ubuntu0jon1~lucid2 [13:54] <ledil> when executing ffmpeg, i cant see a --enable-faac ... is faac deprecated ? [13:54] <someone-noone> ledil, ffmpeg codecs [13:54] <someone-noone> do you see libfaac? [13:55] <Zeeflo> ledil, try the daily build for ubuntu [13:55] <Zeeflo> ah you are [13:55] <Zeeflo> hmm [13:56] <ledil> http://pastebin.com/8xCYWaTc [13:56] <ledil> here is the ouput [13:56] <ledil> of ffmpeg -codecs and faac is missing ... [13:57] <ledil> dont know what to do ? because when Im using the mediaxxx I cant install libavcodec-extras-52 because it conflicts with ffmpeg ... when using jon-severinsson package, im getting "libfaac" unknown ... [13:57] <ledil> I cant find a good ffmpeg package, at the moment :( [13:57] <JEEB> because faac is nonfree and thus impossible to enable if you're making a binary for others [13:58] <ledil> how to solve this ? [13:58] <JEEB> I think even debian stopped linking to it after that was found [13:58] <Zeeflo> nope [13:58] <Zeeflo> works fine here [13:58] <Zeeflo> wheezy 3.2 kernel [13:58] <ledil> my server is lucid [13:59] <JEEB> not sure but in any case I think very recent builds no longer have faac enabled as it's nonfree, methinks [13:59] <JEEB> ledil, 1) switch to fdk-aac, as that is of better quality for AAC 2) compile yourself [13:59] <JEEB> that's the only way [13:59] <Zeeflo> im using Bureks builds.. faac is working here [13:59] <JEEB> both faac and fdk-aac are nonfree [13:59] <JEEB> o_O [13:59] <JEEB> burek has faac enabled? [13:59] <JEEB> Zeeflo, paste ./ffmpeg output [13:59] <Zeeflo> im using the aac libs in my encodings, so yea, id guess so! [13:59] <JEEB> pastebin [13:59] <Zeeflo> sure [14:00] <JEEB> it should have the configure line too [14:00] <someone-noone> ledil, I'd just install ffmpeg from sources [14:00] <Zeeflo> http://pastebin.com/KnJHXz6L [14:01] <JEEB> with fdk-aac instead of faac [14:01] <JEEB> because fdk-aac is fraunhofer's encoder [14:01] <Zeeflo> and this is my input: -c:a aac -strict experimental -ac 2 -b:a 128k [14:01] <JEEB> oh [14:01] <JEEB> that uses the internal one [14:01] <JEEB> not faac [14:01] <Zeeflo> ah [14:01] <JEEB> that'd be called ffaac (ff-something is the internal thing) [14:01] <Zeeflo> but aac is aac right ;) [14:01] <JEEB> yes, but that is experimental and barely as good as vo-aacenc [14:01] <JEEB> (or somewhat better) [14:02] <Zeeflo> it works perfectly well [14:02] <JEEB> but still worse per the same rate than faac or fdk-aac [14:02] <JEEB> and faac isn't exactly great either [14:02] <ledil> Zeeflo: you are using bureks builds ? what ppa repository is that ? i cant switch to fdk-aac at the moment, I need to find a repository with a ffmpeg that has faac support [14:02] <JEEB> there are none [14:02] <JEEB> because faac is nonfree [14:02] <Zeeflo> ledil, you need just the aac [14:02] <JEEB> as I already said, Zeeflo is not using faac [14:02] <ledil> ah,ok [14:02] <JEEB> well, yes -- the "experimental" status mostly comes from the fact that it's not optimal, not that it doesn't work Zeeflo [14:03] <Zeeflo> those two does the same thing [14:03] <Zeeflo> JEEB, I am aware of that, but as long as it works, it works? [14:03] <JEEB> -_- [14:03] <Zeeflo> for my stereo 128k it works like I need it to [14:03] <JEEB> as soon as you hit something that goes below the comfort level of the libavcodec encoder it will be derp [14:03] <JEEB> at least it no longer adds random noise [14:03] <JEEB> it used to do that like 1.5 years ago [14:04] <Zeeflo> dunno about that :) [14:04] <Zeeflo> It works for me :) [14:04] <JEEB> well, if you don't care about audio quality or don't hear the difference then the internal one is OK [14:04] <Zeeflo> you cant hear any difference at stereo level anyways [14:04] <JEEB> you can [14:04] <Zeeflo> u got super ears then [14:04] <JEEB> no [14:04] <JEEB> I've got bad ears [14:05] <Zeeflo> well, im not trying to create THX 7.1 lossless sound here.. [14:05] <JEEB> I have good eyesight for video compression artifacts and bad ears, and I can still easily hear that when you happen to drop below the comfort level of libavcodec's aac encoder, it's worse than fdk-aac here [14:05] <JEEB> and the comfort level is probably somewhere around 128k [14:06] <JEEB> although when I had to use the libavcodec encoder I kept to 192k just in case [14:06] <Zeeflo> you couldnt hear the difference in 128k to 160 or 192k [14:06] <Zeeflo> no one can [14:06] <JEEB> oh yes you can with the libavcodec encoder [14:06] <Zeeflo> dude come on.. [14:06] <JEEB> it's better than the libavcodec vorbis encoder, that's for sure [14:07] <JEEB> probably better than the libavcodec wma encoder, too [14:07] <JEEB> I know that there are encoders where at those rates you really shouldn't be able to distinguish stuff, but the libavcodec's encoder is not one of those [14:08] <Zeeflo> then you probablay got Monster cables with 24k gold all the way through, with oxydized this and that priced at $1000 pr inch on a $200.000 home stereo [14:08] <JEEB> ... [14:08] <Zeeflo> the human ear cannot hear the difference [14:08] <ledil> JEEB, fdk-aac, do I need an additional package to install ? [14:08] <JEEB> yes, you would have to compile it as well [14:08] <Zeeflo> you gotta compile it [14:08] <ledil> or is it only "-acodec fdk-aac" ? [14:09] <JEEB> it is nonfree as well because of the crappy license fraunhofer put on the source [14:09] <ledil> JEEB, is there binary package ? [14:09] <JEEB> so it cannot be distributed either [14:09] <JEEB> in binary [14:09] <JEEB> (linked with (L)GPL at least) [14:09] <Zeeflo> Whats your suggestion instead of bureks aac then? [14:09] <Zeeflo> ill make a sound test [14:09] <Zeeflo> a clinic sound test [14:10] <JEEB> fdk-aac with afterburner on, start around 96k and go up from there with various samples [14:10] <Zeeflo> @ constant bitrate [14:10] <Zeeflo> @128k [14:10] <Zeeflo> vs aac [14:10] <Zeeflo> ill do that [14:11] <JEEB> that's a bit on the high'ish side for fdk-aac probably, but with many sources the libavcodec aac encoder should start showing its skin there [14:11] <Zeeflo> and ill show you the results [14:12] <Zeeflo> crap its cold here [14:12] <JEEB> also please do not compare me with the so-called "audiophiles", I'm all for valid tests and such, and I am just saying that the libavcodec encoder just sucks at its job since no-one finished it >_> [14:12] <Zeeflo> im freezing [14:12] <JEEB> if someone would have the time to put into the libavcodec encoder it'd be great [14:12] <JEEB> unfortunately, it's simpler in most cases for people to just compile fdk-aac and use it locally [14:13] <Zeeflo> people probably just think like me: It works well enough, leave it at that. [14:13] <Zeeflo> also [14:13] <Zeeflo> its the downside of GPL [14:13] <Zeeflo> no-one can force any-one to finish :D [14:13] <JEEB> more like open source [14:13] <JEEB> not specific to GPL in any way [14:14] <JEEB> or all software to be honest [14:14] <Zeeflo> i stand corrected, but you know what I mean [14:14] <JEEB> even with payware there are cases where a developer has stopped developing the app and no matter how much the clients derp he won't finish it [14:14] <JEEB> ("finish" as in fix the bugs etc.) [14:14] <JEEB> also, if you really want a laugh [14:15] <JEEB> try the "ogg" encoder [14:15] <JEEB> not libvorbis, make sure it uses the internal one [14:15] <JEEB> I guarantee you will have a laugh :) [14:15] <JEEB> so compared to that, the aac encoder is much better [14:18] <Zeeflo> h264 wont work with ogg would it? [14:19] <Zeeflo> in a mp4 container [14:19] <JEEB> yes, because it's a container that doesn't specify a way to mux vorbis in it. It has nothing to do with H.264 :) [14:19] <Zeeflo> i use libx264 with aac for mp4 containers, for streaming (rtmp) [14:19] <Zeeflo> flash and html5 fallback [14:20] <Zeeflo> i dont think ogg would work with that? [14:20] <JEEB> I'm not saying you should use it [14:20] <Zeeflo> would flash eat that? [14:20] <JEEB> I just said that if you were going to try out aac and fdk-aac, you might as well quickly encode the same thing with the vorbis encoder [14:20] <Zeeflo> well. that would depend on the player.. [14:20] <JEEB> just for laughs :P [14:20] <JEEB> also rtmp(e) is flv [14:20] <JEEB> not mp4 [14:21] <Zeeflo> nah, im using mp4 [14:21] <JEEB> of course your rtmp(e) service can take in already encoded mp4 [14:21] <JEEB> and remux that on the fly [14:21] <JEEB> into flv [14:21] <JEEB> just that live streaming doesn't fly with that [14:21] <JEEB> unless your mp4 muxer suddenly supports movie fragments [14:21] <Zeeflo> it does [14:21] <Zeeflo> -movflags [14:22] <Zeeflo> +faststart [14:22] <JEEB> uhhh [14:22] <JEEB> that's not it [14:22] <JEEB> that's just putting the index in the front /after/ encoding [14:22] <Zeeflo> do you mean somthing like qtindexing then? [14:23] <Zeeflo> moov atoms? [14:23] <JEEB> no, I mean like something like L-SMASH movie fragments feature. the libavformat demuxer supports it, but muxer AFAIK doesn't [14:23] <JEEB> moov atom moving is moving the index [14:23] <JEEB> can only be done /after/ encoding [14:23] <Zeeflo> i dont know about movie fragtments.. [14:23] <Zeeflo> My setup works.. [14:23] <JEEB> yes, because your thing most probably just reads the mp4 and remuxes it to flv on the fly when feeding it to rtmp(e) [14:23] <Zeeflo> MP4's on amazons CDN (cloudfront) played with JW player 6.1 [14:24] <JEEB> I'm just telling you that rtmp(e) by itself is flv [14:24] <muken> JEEB: ? libavformat muxer supports movie fragments [14:24] <JEEB> muken, oh? [14:24] <JEEB> I've just never seen them used [14:24] <JEEB> lol [14:32] <JEEB> Zeeflo, btw if you really want to see what the CDN is doing to your content, you can point ffmpeg to the rtmp(e) URL the player reads (ffmpeg -i rtmp://url:port ) and see what format etc. it sees coming towards you [14:32] <JEEB> I would guess it just remuxes your mp4 files into flv and transports them like that [14:33] <Zeeflo> i would have to make the file public in the bucket then wouldnt i? [14:33] <JEEB> nope [14:33] <JEEB> at least I have no idea what you're talking about [14:33] <Zeeflo> heh [14:33] <Zeeflo> i know I would have to [14:33] <Zeeflo> :) [14:33] <JEEB> if you are talking about pastebinning, then no -- you can clear out the rtmp(e) url or whatever from the command line and so forth [14:34] <JEEB> and the rtmpe handshake [14:34] <Zeeflo> nono, I am using signed and timed urls for my content [14:35] <JEEB> anyways, just trying to tell you that it pretty surely isn't mp4 when it gets pushed through rtmp(e) [14:35] <Zeeflo> ill make a file public and see :) [14:36] <JEEB> sure, ffmpeg should be able to read what's coming up from the tube with an rtmp:// url as input [14:36] <Zeeflo> BTW [14:37] <Zeeflo> bureks build has a bug [14:37] <Zeeflo> how do I submit it? [14:37] <JEEB> depends on if it's a bug in the build or bug in ffmpeg itself [14:37] <Zeeflo> i dont know.. [14:37] <JEEB> if you still get the bug with building the same revision yourself f.ex. then you poke bug burek [14:38] <JEEB> otherwise your poke the ffmpeg trac issue tracker [14:38] <Zeeflo> when you select subtitles=blabla.srt, if theres no fonts on the server, it doesnt warn you [14:38] <Zeeflo> server=system [14:39] <Zeeflo> it did that on my old server as well [14:39] <Zeeflo> I had to get the ms core fonts [14:39] <Zeeflo> and install fontconfig [14:39] <Zeeflo> and then I had subs in my stream [14:39] <JEEB> so if the system lacks any fonts of any description it just fails to render anything instead of failing with an error? [14:39] <Zeeflo> before I did that, it acted like it was creating a new stream with subs in it, but it wasnt [14:39] <JEEB> uhh [14:40] <Zeeflo> an no warning/error messages [14:41] <Zeeflo> with everything else, ffmpeg is pretty good at warning you or even halting [14:41] <Zeeflo> just not with this! [14:41] <Zeeflo> atleast for the debian build [14:43] <JEEB> that sounds like a ffmpeg bug, but you might want to check how it handles itself in case you build a build with current ffmpeg with libass etc. yourself, and then try it out like that. Although I feel in most cases like that the build would then depend on shared libraries and would actually derp at you for trying to start it up without f.ex. fontconfig to begin with (Before ffmpeg's code actually runs) [14:44] <JEEB> burek's build should have everything in it static methinks, so even if you don't have the stuff around, it would still kind of go forward [14:44] <JEEB> naturally lacking any font cache or anything you wouldn't really get anything [14:45] <JEEB> so I guess that some checks might be looked into, but depending on how "dumb" the checks would have to be done, or where they would have to be done (ffmpeg? or fontconfig? or something else?) it could probably take time to see it get completely "fixed" [14:46] <Zeeflo> yea [14:46] <JEEB> for example, if it's just a case of lax checking in fontconfig [14:46] <JEEB> even if you fix it right now [14:46] <JEEB> it won't fix machines already out there [14:46] <Zeeflo> one just gotta remember to install fonts on his/her new server if they want subs in their streams! Otherwise they might gotta start all over! [14:46] <Zeeflo> <--- like he had to! [14:47] <Zeeflo> :D [14:47] <JEEB> I generally build my stuff myself so I generally grab all the related libraries for that [14:47] <Zeeflo> i tried that as well [14:48] <Zeeflo> but I couldnt get it to support SRT subs [14:48] <Zeeflo> I had to convert everything to ass [14:48] <Zeeflo> untill I got Bureks build [14:48] <JEEB> sure you were building everything new enough? [14:48] <Zeeflo> Besides, I dont think building it myself would make it any bit better! [14:48] <Zeeflo> a wise man said, if it works dont fuck with it [14:49] <Zeeflo> and his build works [14:49] <Zeeflo> atleast for me [14:50] <JEEB> well, it would at least make you not forget fontconfig, fonts you'd still have to grab somewhere but I'd guess in most cases those might be grabbed as a dependency? [14:50] <JEEB> also I'm just used to building ffmpeg and friends by now :V [14:51] <JEEB> I can see why burek's builds are quite useful for many people tho [14:51] <JEEB> they offer an easy way out for those who don't know their ways to compile the stuff they need [14:51] <Zeeflo> i just install it directly [14:52] <Zeeflo> fontconfig isnt a dependency of msttcorefonts [14:52] <Zeeflo> but its needed for ffmpeg [14:52] <JEEB> I know it isn't :| [14:53] <JEEB> also I was talking of it the other way, generally grabbing fontconfig/freetype might end you up with some fonts too in many cases -- I might be completely incorrect tho naturally [14:54] <Zeeflo> im still missing fonts though [14:54] <Zeeflo> ffmpeg is still complaining, but atleast it can fallback to arial [14:54] <Zeeflo> it want Deva Vu [14:55] <Zeeflo> im perfectly fine with arial [14:59] <Zeeflo> JEEB, is there also a command to level the volume fixed? [14:59] <Zeeflo> ive noticed some video files has high volume [14:59] <Zeeflo> some has moderate [14:59] <Zeeflo> some almost has no volume :D [15:04] <durandal_1707> volume , volumedetect filter [15:05] <durandal_1707> they are all listed in filter documentation available on web [15:07] <Zeeflo> i must have missed something i wrote earlier [15:07] <Zeeflo> ;) [15:11] <ubitux> ebur128 is likely more effective than volumedetect [15:12] <ubitux> (but slower) [15:28] <_kud> Hello [15:29] <_kud> I'm trying to do a video with some png; i've got this script= https://gist.github.com/1e0fdf400961e7e5d779 [15:29] <_kud> but it seems not to loop it 20 times [15:29] <_kud> do you have any idea how to make a loop properly? [15:51] <undercash> hello [15:52] <undercash> i m trying to compile ffmpeg on ubuntu precise 64 and i get a ERROR: libx264 not found [15:52] <undercash> which didnt happen for a very long time :) [15:52] <undercash> oops 32bit sorry, testing on laptop [15:53] <JEEB> libx264-dev package is not installed? [15:53] <JEEB> if you want to use the package that is [15:53] <JEEB> if you built libx264 yourself then make sure you had either --enable-static or --enable-shared set [15:53] <undercash> yea it is installed [15:54] <JEEB> check config.log then? [15:54] <undercash> ok let's put it [15:54] <hackeron> hey, I'm trying to record from a granstream video server, I'm doing: ffmpeg -i rtsp://admin:admin@192.168.0.214:554 -analyzeduration 0 -codec:v copy -codec:a libfaac -flags +qscale -global_quality 1 -afterburner 1 test.mkv --- that gives me broken metalic sound. If I switch libfaac to libmp3lame then I get good audio but it is not in sync with the video. Any ideas? [15:54] <JEEB> lol [15:54] <JEEB> so which is it? [15:54] <undercash> the --enable-static i mean [15:54] <undercash> but note that it wasn't necessary like 2 weeks ago [15:55] <JEEB> undercash, so which is it -- are you building with a pre-built binary package of libx264 or are you building it yourself? [15:55] <undercash> myself [15:55] <JEEB> if you aren't building it yourself you don't add that [15:55] <JEEB> oh [15:55] <undercash> following the well known topic, i want to add libass [15:55] <JEEB> so you had neither --enable-static or --enable-shared on your x264 configure line? [15:55] <undercash> seems interesting [15:55] <undercash> neither yes, but before it wasn't a problem [15:55] <undercash> if i remember well [15:55] <JEEB> that was quite a long time ago then [15:55] <JEEB> not weeks [15:56] <undercash> 2 weeks ago [15:56] <JEEB> no, that wasn't the up-to-date x264 then [15:56] <JEEB> I know for at least a year if not more x264 has needed one of those [15:56] <JEEB> or it would not install the libraries [15:56] <undercash> i guess it was since i had to update yasm too ;) [15:56] <JEEB> no [15:56] <JEEB> yasm update to 1.2 was new [15:56] <JEEB> that was added for the possible addition of haswell asm [15:56] <undercash> anyway thx you i got to recompile x264 [15:57] <undercash> not really useful to debate if it s a recent update or not.. [15:57] <JEEB> you might as well do --disable-cli too [15:57] <XATRIX> Guys, i need some advices... I'm trying to catenate a few mp3 files and, result file is not readable on some players... I'm interested in JS players, and chrome player. Can someone help me to debug ? [15:57] <JEEB> because you only need the library for ffmpeg [15:57] <JEEB> not the command line encoder [15:58] <undercash> i came here coz i thought maybe something was broken in the ffmpeg git [15:59] <undercash> ok JEEB thx, i m recompiling [15:59] <undercash> yes i never use x264 as standalone [16:00] <hackeron> My input audio is pcm_mulaw, 8000 Hz, mono, s16, 64 kb/s from RTSP - any ideas why libfaac would generate broken, metalic sounding audio while libmp3lame sounds fine? - Anything I can do in ffmpeg like maybe force a higher sample rate for the output audio? [16:02] <Mavrik> hackeron: it's quite possible libfaac doesn't support that sample rate [16:02] <Mavrik> hackeron: you can try using libfdk_aac, which is pretty much currently the best ffmpeg supported AAC encoder [16:02] <Mavrik> hackeron: or try resampling the audio to 22050 or some such [16:04] <JEEB> undercash, http://git.videolan.org/?p=x264.git;a=commit;h=c1e60b9032196d204db8dce77051… [16:04] <JEEB> it was may 2011 [16:04] <JEEB> where it changed [16:04] <JEEB> everything from this commit onwards needed --enable-static or --enable-shared [16:06] <hackeron> Mavrik: I tried: ffmpeg -i rtsp://admin:admin@192.168.0.214:554 -analyzeduration 0 -ar 44100 -codec:v copy -codec:a libfdk_aac -async 1 -flags +qscale -global_quality 1 -afterburner 1 test.mkv -- now it sounds OK but it is out of sync with the video [16:06] <hackeron> Mavrik: it stays in sync if I use libmp3lame though :/ [16:07] <Mavrik> *grumbl* :) [16:07] <Mavrik> hackeron: try without resampling when u sing fdk_aac [16:07] <hackeron> Mavrik: then it dies saying: [libfdk_aac @ 0x7f8ce88b2c00] Unable to encode frame: Encoding error [16:07] <hackeron> Audio encoding failed (avcodec_encode_audio2) [16:08] <undercash> did the job, thx jeeb [16:11] <JEEB> np [16:14] <antonello_> Can I obtain an decoder without reading the file header? [16:14] <hackeron> Mavrik: any ideas? [16:14] <Mavrik> not really, shouldn't go out of sync :\ [16:15] <Mavrik> does your input get probed properly (your analyzeduration is awfully short?) [16:15] <Mavrik> antonello_: yep [16:16] <hackeron> Mavrik: hmm, not sure - but changing libfdk_aac to libmp3lame in the above command I get audio/video sync [16:16] <hackeron> and changing the libfaac I get broken metalic sound [16:17] <Mavrik> antonello_: just call avcodec_find_encoder with right encoder ID [16:18] <antonello_> i do this ... [16:19] <Mavrik> good. [16:20] <antonello_> I obtain a AVCodecContext ... and after i setting the codec parameters [16:21] <antonello_> channels sample_rate bit_rate sample_fmt channel_layoyt [16:21] <antonello_> my error is [16:21] <antonello_> [aac @ 0x10bd800] Not evaluating a further program_config_element as this construct is dubious at best. [16:21] <antonello_> [aac @ 0x10bd800] channel element 1.7 is not allocated [16:22] <antonello_> I don't understand where i miss . [16:23] <hackeron> Mavrik: same story with libvo_aacenc - it seems to be about 1.5 seconds out of sync just like libfdk_aac - but perfect sync with libmp3lame. Does ffmpeg calculate duration different with libmp3lame? [16:23] <Mavrik> hackeron: hmm... wouldn't know, but duration has little to do with it :\ [16:24] <antonello_> this is my code http://pastebin.com/iUGmCq6J line 184 -190 [16:25] <Mavrik> antonello_: um [16:26] <hackeron> Mavrik: any ideas why libmp3lame would be in sync and libfdk_aac and libvo_aacenc audio is around 1.5 seconds behind the video? [16:26] <Mavrik> antonello_: AVFormatContext won't have its fields filled out if you don't read the file header [16:26] <Mavrik> antonello_: which also means your ids and other data will be wrong [16:26] <Mavrik> hackeron: no idea really... that shouldn't happen :\ [16:26] <Mavrik> hackeron: did you try using -async 22050 or something like that [16:28] <antonello_> then.... what should I use? [16:29] <Mavrik> either probe the file to fill out those fields [16:30] <Mavrik> or don't read them at all [16:30] <Macey> Hi All, who looks after the API? [16:31] <Mavrik> O.o [16:31] <hackeron> Mavrik: just tried: ffmpeg -i rtsp://admin:admin@192.168.0.214:554 -analyzeduration 0 -ar 44100 -async 44100 -codec:v copy -codec:a libfdk_aac -flags +qscale -global_quality 1 -afterburner 1 -y test.mkv -- no change, audio is the same amount behind video - libmp3lame still in sync :/ [16:34] <Diogo> hi i need to capture sound from a blackmagic card using ffmpeg, i don't have sound input...anyone with the same problem? [16:46] <antonello_> Can I obtain an header to add an head of my streeam ? [16:49] <hackeron> I'm reading audio/video from an RTSP source. I have -ar 44100 -async 1 and -flags +qscale - when I use -codec:a libmp3lame - audio and video are in sync, however when I switch to libfdk_aac or libvo_aacenc the video is about a second ahead of the audio. Any ideas? [16:53] <hackeron> ah wait, I tried to create an mp4 file and now the libmp3lame is out of sync by the same amount, hmm [17:06] <hackeron> ah, I had to move -async 1 to be after -codec:a libmp3lame - then I get audio/video sync with libmp3lame - but not with libfdk_aac or libvo_aacenc -- is -async 1 not supported with those? [17:30] <Youka> I'm a bit confused about avcodec_decode_video2 and the auto allocated bitmap inside AVFrame. In the examples, AVFrame bitmap is never freed after decoding, so is it a memory leak (which is ignored because it's small and the program exits after it)? [17:30] <teratorn> Youka: run some tests under valgrind [17:31] <jeje34> hi to all ;-) [17:33] <jeje34> I'm using the last release version of FFMPEG (1.1.1) and it seems the function av_lockmgr_addref and av_lockmgr_release doesn't exist anymore. Before, I was using AVCodec 5.2, and these functions were existing [17:35] <jeje34> sorry I was disconnected [17:35] <jeje34> I use ffmpeg to decode h264 video in a multithread appication [17:40] <jeje34> and in time, I've the error : Insufficient thread locking around avcodec_open/close() whith a crash of my application [17:43] <jeje34> when I was using avcodec52.dll, i call if (av_lockmgr_addref() == 1) [17:43] <jeje34> { av_lockmgr_register(&ff_lockmgr); [17:43] <jeje34> } [17:44] <jeje34> with static int ff_lockmgr(void **mutex, enum AVLockOp op) and casting mutex to CRITICAL_SECTION (I'm on windows envirronment) [17:45] <jeje34> but with the last version, the av_lockmgr_addref doesn't exist [18:03] <jeje34> what happen if several thread call av_lockmgr_register(&ff_lockmgr) [18:03] <jeje34> ? [19:19] <JoeyJoeJo> I have an mkv file that won't play and I'm not sure why. Can ffmpeg repair the file or at least let me know what the problem is? [19:20] <klaxa> probably, try running it through ffprobe [19:20] <klaxa> also some more information on the exact way you can't play it would be great [19:20] <klaxa> i.e. what player and what error message [19:21] <JoeyJoeJo> ffprobe seemed to work. It said "Invalid data found when processing input" [19:21] <klaxa> if you torrented the file, verify your local data, maybe the header is corrupted, i have similar problems from time to time [19:22] <klaxa> mplayer2/libavformat interprets the file as DV Video and starts playing back garbage [19:22] <JoeyJoeJo> I'm trying to play it with VLC and I don't see any picture or hear any sound. The seek bar doesn't even count. And I can't find any errors either [19:23] <JoeyJoeJo> I did torrent it. so I try verifying my local data [21:10] <llogan> burek: perhaps the IMPORTANT note on gusari forum should be even bigger. [21:26] <bparker> is there a way to embed exact pixel colors at certain positions within a compressed video? For example I have a compressed video and I want the first pixel of the video to be 100% blue. It doesn't matter if the video has to be re-encoded, that's fine. Any idea if that's possible? [21:27] <bparker> I'm worried about the compression messing with the color values themselves [21:28] <klaxa> depends on the codec i guess, but yeah compression will probably mess with that stuff [21:29] <klaxa> lossly compression that is [21:35] <bparker> klaxa: right, that's what I'm trying to work with unfortunately [21:35] <bparker> I need some markers placed on each corner of a compressed video [21:35] <bparker> with specific pixel values because I'm essentially encoding auxillary data there [21:36] <klaxa> can't you work with approximate values? [21:36] <bparker> sortof like steganography, but really small bits of data [21:36] <bparker> it's possible, but I'm worried about false positives [21:36] <bparker> if the video happens to contain a color in the range I'm looking for, that's bad [21:37] <klaxa> mh yeah [21:37] <bparker> and I don't want to visibly alter the video enough that it's distracting to users [21:37] <bparker> becaue the displays can see 100% of the pixels [21:37] <bparker> what I'm basically doing is [21:38] <bparker> taking a video signal in from a device, capturing it and doing image analysis on it [21:38] <bparker> then based on that analysis, doing different kinds of post-processing on the image and outputting it to a TV [21:38] <bparker> analysis being some bits I want set that tell me what to do with the video [21:39] <bparker> and the only way I have to communicate that is with the actual video data (pixels) itself [21:40] <klaxa> i don't think you'll be able to achieve that unless you use lossless codecs or settle with approximation [21:40] <klaxa> or establish a second independent data channel of some sort [21:40] <bparker> that's what I thought [21:41] <bparker> unless I can come up with some kind of scheme/algorithm that is resistant to the compression, that's really my only option [21:42] <klaxa> and you have to use the videostream? why not try to code it in binary at the top or bottom and analyze that within some tolerance range? [21:42] <klaxa> like black and white pixels [21:42] <klaxa> you should be able to differenciate black and white even when compressed [21:43] <klaxa> although it may put strain on the codec because hard edges are hard to compress [00:00] --- Fri Feb 1 2013

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