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January 2013

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ffmpeg.log.20130105
by burek 16 Jan '13

16 Jan '13

[00:00] <sesam> Mavrik, same here but in avid [00:00] <Mavrik> t4nk637: audio? x264? [00:00] <llogan> t4nk637: mp4 is a container format that can contain various video and audio formats [00:00] <t4nk637> no audio [00:00] <Mavrik> sesam: sadly, haven't found a decently cheap way of doing Premiere renders on a server :\ [00:01] <Mavrik> t4nk637: so, you'll need "image2, mp4" muxers, "libx264" encoder, "mjpeg" decoder [00:01] <sesam> mavrik: isn't there a distrubuted render app that you can install on other computers? i know after effects does [00:01] <llogan> Mavrik: i tend to use a frameserver from premiere to ffmpeg because i hate AME [00:01] <Mavrik> t4nk637: disable ffserver, ffplay, ffprobe [00:01] <Mavrik> t4nk637: and all devices and hwaccels [00:02] <Mavrik> sesam: aftereffects yes, premiere no :\ [00:02] <llogan> --disable-everything --enable-foo for an extreme method [00:02] <Mavrik> t4nk637: what llogan said, start with "--disable-everything" and enable just those muxers, encoder and decoder [00:02] <t4nk637> ...yup, that sounds great... I'd seen the --disable-everything --enable-goodstuffonly ... [00:02] <llogan> there is --enable-small if size is a concern, but i've never really looked at that [00:03] <t4nk637> awesome, the list you showed helps [00:03] <Mavrik> --enable-small is ok if you need a small binary [00:03] <Mavrik> but that will come at performance cost though [00:03] <Mavrik> I suggest you leave that out if you're on a slow device and can afford a little flash. [00:03] <t4nk637> processing will happen at night, so performance can be sacrificed.... cool [00:03] <llogan> ...since you mentioned "embedded" [00:04] <t4nk637> Kirkwood [00:04] <t4nk637> = sortaslow [00:04] <Mavrik> llogan: frameserver? [00:04] <luc4> Mavrik: I see in the AVStream structure of my AVFormatContext there is a AVCodec, but that is not initialized. Do you know why I need to init that manually? [00:04] <llogan> Mavrik: https://ffmpeg.org/trac/ffmpeg/wiki/FFmpegPremierePro [00:04] <llogan> might work for avid too [00:05] <sesam> llogan: that i will look up.... [00:05] <Mavrik> luc4: hmm, you should have AVStream's in the AVFormatContext and then AVCodecs attached to each stream [00:05] <Mavrik> luc4: and you have to open the codec for each of those streams [00:05] <Mavrik> llogan: oh, that looks interesting [00:06] <Mavrik> thanks :) [00:06] <luc4> Mavrik: oh, so maybe I just have to open? In the sample codes I see I also have to allocate so I was asking myself why. [00:06] <Mavrik> hmm [00:06] <t4nk637> thnx #@ llogan & Mavrik ; byebye [00:06] <llogan> Mavrik: feel free to edit/update the wiki page if something is not clear/correct [00:06] <Mavrik> will do [00:07] <llogan> i'm not usually the editor at my office, so i don't actually end up using premiere that much [00:07] <llogan> thankfully [00:07] <sesam> llogan: =) [00:07] <klaxa> btw Mavrik, just checked with my logs, it was you i talked about the stuff with splitting h264 streams and all the shit :P [00:07] <llogan> which reminds me i need to go get some more b-roll before it gets too dark [00:08] <sesam> llogan: good luck [00:08] <llogan> thanks. you too. [00:08] <klaxa> and in the end i just took the I frames output by mkvtoolnix which seem to be IDR frames after all [00:08] <klaxa> and i pass muxed files [00:09] <llogan> Mavrik: oh, i noticed that one project file no longer shows the render queue...possibly related to debugmode frameserver...possibly not, so try with a backup first. [00:09] <llogan> whatever the cause, it's terribly annoying. [00:10] <llogan> i'll blame adobe [00:10] <Mavrik> ^^ [00:10] <Mavrik> blaming Adobe is rarely wrong [00:10] <llogan> "render queue" being the "export video" or whatever it's called. [00:12] <llogan> i've also seen some audio/video sync issues, but i just ended up muxing them separately. i didn't get to investigate much. [00:13] Action: llogan leaves for real [01:20] <verdoc> say I've got "videofile.mp4" and it's encoded exactly how I want... what's the best way to create a "loop-version.mp4" that is the same video just looped X amount of times? [01:22] <Mavrik> probably by a tool like mp4box [01:22] <Mavrik> to concat it several times [01:22] <verdoc> I tried that actually [01:23] <verdoc> that output file didn't seem to have an updated length [01:23] <seece> are -slice-max-size and -keyint options deprecated? [01:24] <Mavrik> verdoc: hmm, possible [01:24] <Mavrik> mp4 format isn't really built for easy concatenation :\ [01:25] <verdoc> I'm beginning to pick up on that [01:25] <verdoc> ;) [01:25] <verdoc> I tried this approach - http://revision3.com/forum/showthread.php?t=37586 [01:25] <Mavrik> the ugly and annoying hack is to remux to .ts, concat that and remux to mp4 :) [01:25] <Mavrik> but mp4box should really be able to do that :\ [01:26] <klaxa> shouldn't ffmpeg itself have a loop functionality? [01:26] <Mavrik> verdoc: changing to mpg will reencode the file [01:26] <klaxa> it's referenced in the online manpage, but nowhere actually mentioned [01:26] <Mavrik> it was deprecated or something [01:26] <Mavrik> iirc [01:27] <verdoc> do you have sample of the remux hack? [01:27] <klaxa> -loop_input and -loop_output are marked as deprecated and one should use -loop instead, but -loop isn't explained anywhere [01:27] <verdoc> yeah, I saw that in the man page too klaxa [01:27] <verdoc> it looks like it's used to create the single image youtube music "video" songs [01:27] <Mavrik> verdoc: ffmpeg -i <test.mp4> -codec copy -bsf:v h264_mp4toannexb output.ts [01:27] <Mavrik> then concat files in any way [01:28] <Mavrik> and do ffmpeg -i <test.ts> -codec copy output.mp4 [01:29] <verdoc> it didn't like that [01:29] <Mavrik> no? :) [01:29] <verdoc> I got dep warnings [01:29] <verdoc> and it said -codec wasn't valid [01:30] <Mavrik> just how old is your ffmpeg? :) [01:31] <verdoc> ffmpeg 0.8.4-4:0.8.4-0ubuntu0.12.04.1 [01:31] <Mavrik> that would explain the errors [01:31] <verdoc> *** THIS PROGRAM IS DEPRECATED *** [01:31] <verdoc> This program is only provided for compatibility and will be removed in a future release. Please use avconv instead. [01:31] <Mavrik> use "-vcodec copy -acodec copy" instead of "-codec copy" [01:31] <JEEB> switch to the command 'avconv' in that case [01:31] <JEEB> that's the updated binary with libav [01:31] <JEEB> :) [01:31] <verdoc> same syntax? [01:31] <JEEB> yes [01:32] <JEEB> (mostly) [01:33] <JEEB> the biggest difference I know is how you set libx264 the more fine-grained specific settings (-x264opts vs specific -settings) [01:33] <verdoc> `avconv -i <test.mp4> -codec copy -bsf:v h264_mp4toannexb output.ts` seems to be doing something [01:38] <llogan> verdoc: -loop only works with image inputs and formats that support looping such as animated GIF [01:46] <verdoc> that worked [01:46] <verdoc> thanks guys [02:26] <luc4> Hi! Can I somehow decode a 6 channels AAC stream to 2 channels? [02:39] <pods_guest_974> hi what is the best way to detect a anime/porn movie ...by colors, histogram of colour?? ffmpeg can help to do this? [02:46] <Tjoppen> those are two separate questions [03:20] <limpc> probably means hentai, and theres no way to do that [03:21] <limpc> if its not in the file info, you cant detect by a frame [03:21] <limpc> cant tell the difference between Legend of the Overfiend, and Naruto :P [03:40] <p4plus2> Is there any way to specify which jack ports for ffmpeg to record from? Ideally selecting three channels (left, right, and a mono) and having them mix down to two channels(left+mono, right+mono) [03:41] <p4plus2> (left and right generally being system or program sounds, mono being from a microphone) [03:44] <p4plus2> actually I may have a guess as to how to make it work -- i'll try it out [03:46] <p4plus2> hmm nothing [03:47] <p4plus2> ok so it seems "ffmpeg -f jack -i name_here" creates a port for hooking ffmpeg up to [03:48] <p4plus2> but is there a way to redirect sounds there through ffmpeg? [03:52] <p4plus2> maybe I'll try jack capture and just merge the two sources later [03:52] <p4plus2> last question then would be is there any way to pause a recording? [03:54] <p4plus2> actually maybe STOP/CONT will work -- if that is even a safe action for ffmpeg... [04:00] <raket> what about ffmpeg -f alsa -i plughw? or is your 1) soundcard uttercrap 2) you run a soundserver ? :-) [04:03] <p4plus2> well I'd rather use jack so I can capture explicit programs and not some others [04:05] <p4plus2> alsa, at least to my knowledge, doesn't have a simple way of allowing that (maybe something more complicated with loopback devices could be done, but that requires all of my programs to allow me to easily use said device) [04:05] <raket> it works, if you have a real soundcard with more then 1 channel [04:06] <p4plus2> My sound card is (for now anyways) just a on board sound card with very limited options [04:06] <raket> but it's very uncommon with such soundcards and no one seems to have documentet but it does in fact exist and does work. i've used this long long time ago [04:07] <p4plus2> Ah, I'll try and dig something up in a bit -- gotta go eat dinner real quick though [04:07] <p4plus2> Thanks for the info [04:07] <raket> hope jack works! :-) [04:07] <raket> (i have the same problem as you describe) [04:09] <raket> anyone here? i need to know a video codec where i can force 8bit video [04:29] <luc4> Hi! I'm trying to decode a AAC audio stream with S32 sample format, 6 channels 48000 samples per second, but this is what I get: ftp://luc4.homeip.net/output.pcm. I used this as a reference: http://ffmpeg.org/doxygen/trunk/decoding__encoding_8c_source.html#l00238. Any idea why that is the result? [04:37] <DJJeff> can ffmpeg do mp4 to swf ? [04:38] <DJJeff> I see mp4 to flv using winff [04:39] <luc4> Seems possible. [04:44] <DJJeff> well I got my video into flv just need it in swf heh [04:44] <DJJeff> im impressed :-) [04:46] <p4plus2> DJJeff: via google I found "ffmpeg -i input.flv -acodec copy -vcodec copy output.swf" [04:47] <DJJeff> yes that worked.... lost quite a bit of quality [04:48] <DJJeff> my mp4 file was 9MB the swf was only 1MB [04:50] <DJJeff> just tried from mp4 to swf its complaining about sample rate or something [04:55] <p4plus2> Hmm [04:55] <p4plus2> maybe play with the settings where you converted to flv? [04:59] <DJJeff> might work [04:59] <DJJeff> going from mp4 to swf the filesize is actually getting large 30MB+ [05:00] <p4plus2> How was the quality? [05:04] <DJJeff> dunno it ummm gave me warnings and I pressed ctrl c [05:44] <DJJeff> seems like pushing the bitrate up to about 3000 in winff seems to maintain quality [05:44] <DJJeff> from mp4 to flv [05:44] <DJJeff> resulting in a 7MB flv from my 9MB mp4 [05:45] <DJJeff> and about a 3.5MB swf [05:46] <DJJeff> alot of quality is lost from flv to swf :-( [05:47] <DJJeff> oh I see the bitrate dropped to 1500 [06:02] <DJJeff> cant seem to get the bitrate from flv to swf to go higher then 1500 [06:02] <DJJeff> seems to ignore the -minrate all together [11:38] <misterno> hey would i get better quality if i use source with certain resolution and then compress that or get source with higher resolution then lower it to same resolution as in previous example and then compress? [11:39] <misterno> i'm talking about a game that can be resized [11:39] <misterno> so in both example i would end up with equal resolution [11:41] <Mavrik> misterno: resizing will be done before encoding in any case [11:41] <Mavrik> so unless you use a noticably worse resizing algorithm there will be no difference [11:41] <misterno> wouldn't game do better resizing itself? [11:42] <misterno> likea 3d game [11:42] <misterno> but your point is good so no detail will be taken from higher resolution source, it would be resized first [11:43] <misterno> i'm just looking into how to record games in best possible quality using little amount of space [11:45] <misterno> i see some streams on youtube at 320 or 480 that have quite good quality, then i see streams that are way too blurry at those resolutions [11:46] <Mavrik> that's usually just differences in set bitrate of video [11:46] <Mavrik> the resizing algorithm doesn't really make a noticable difference in content like games [11:46] <misterno> maybe codec used too? [11:49] <Mavrik> usually yes, but I doubt people use anything other than h.264 nowdays [11:50] <misterno> i use afterburner to record video and it doesnt have h.264 [11:50] <misterno> it offers uncompressed, RTV1, MJPG and VFW compression [11:52] <Fjorgynn> lol [11:52] <misterno> do you know better app than afterburner? [11:53] <BtbN> DXtory + vlc [11:53] <BtbN> at least on windows [11:53] <misterno> hmm never heard of dxtory i will check it out thanks [11:54] <BtbN> it mapps screen capture to a virtual DirectShow device which VLC can use as source [11:55] <Mavrik> misterno: hm, yea, uncompressed is usually the best for quality then [11:55] <Mavrik> misterno: MJPG with really really relaxed settings is ok as well [11:55] <misterno> oh wait i was wrong [11:56] <misterno> because im new to this i didnt know i can setup VFW and then choose one of many codecs one which is x.264vfw [11:56] <misterno> so it does support it in afterburner [11:57] <misterno> uncompressed is unfortunatelly too taxing for my system which doesnt allow me to play game properly [11:57] <misterno> so i have to use some kind of compression [11:57] <misterno> also when i record 30 fps it just doesnt look nice [11:57] <misterno> i prefer at 60 when it looks smooth [11:59] <misterno> is there difference between H264 and h264? [12:00] <sacarasc> The shift key. [12:02] <misterno> why i have both offered in x264vfw ffmpeg configuration under "fourcc" ? [12:02] <misterno> same goes for x264 and X264 [12:16] <Yulth> Hi everyone! [12:17] <Yulth> Is there any way to store the mp3 id3 data into an mp4 container with HE-AAC audio and, furthermore, with accurate estimation of duration? [12:18] <Yulth> I mean, the command on this pastebin, avoiding the lines reported also on the pastebin :) http://pastebin.com/8cGVqmWi [12:19] <Yulth> any ideas? [12:19] <durandal_1707> you tried and it did not work? [12:21] <relaxed> Yulth: Isn't '-f adts' raw aac, and not mp4? [12:22] <durandal_1707> and why mp3 is in output at all, please paste full and uncut output [12:22] <Yulth> durandal_1707: it produces an mp4 (HE-AAC) audio file with no metadata from the mp3 file [12:22] <Yulth> ok, let me 1 minute! :D [12:22] <relaxed> remove -f adts [12:23] <Yulth> relaxed: ffmpeg doesn't work if '-f adts' is removed, because doesn't recognize the format of the output audio... [12:26] <Yulth> the pastebin with full output [12:26] <Yulth> http://pastebin.com/ApWVFQAB [12:28] <durandal_1707> remove -f adts [12:30] <Yulth> so, here the error: http://pastebin.com/a3yUSUfM [12:31] <durandal_1707> that is cover, if you do not care about cover, add -vn [12:33] <Yulth> let me try it [12:35] <Yulth> wooa!! it worked!! [12:35] <Yulth> :) [12:35] <Yulth> and any solution to solve the warning message: "Estimating duration from bitrate, this may be inaccurate" ? :D [12:37] <durandal_1707> that is from demuxer level and does not affect muxing in any way [12:38] <cytrinox> hi [12:38] <durandal_1707> in other words mp4 duration should be exact to what is duration of decoded mp3 stream [12:39] <Yulth> durandal_1707: but when it is converted and streamed in real time, Cannot produce errors when decoding by clients? [12:40] <durandal_1707> no, decoding should be fine [12:40] <cytrinox> simple question: is it possible to set the display width/height for matroska output files? something like -s hd1080 --display-size-foo=hd720? [12:41] <Yulth> durandal_1707: ok, and it Cannot produces seek errors, isn't it? [12:42] <durandal_1707> cytrinox: size of video or display size which define pixel real size? [12:43] <durandal_1707> Yulth: what protocol for streaming, not all support seeking [12:43] <Yulth> html [12:43] <cytrinox> durandal_1707: I mean display size [12:44] <Yulth> some audio players supports hatml header "Accept-Ranges: bytes" [12:44] <Yulth> *html [12:44] <cytrinox> | + Display width: 1920 [12:44] <cytrinox> | + Display height: 1040 [12:44] <cytrinox> these are properties of the output video track [12:44] <durandal_1707> Yulth: html is not protocol for streaming [12:45] <durandal_1707> cytrinox: and what is size of video? [12:45] <cytrinox> the question is if it is possible to instruct ffmpeg to encode the video as something link hd1080, but set the display size to something different [12:45] <cytrinox> | + Video track [12:45] <cytrinox> | + Pixel width: 1920 [12:45] <cytrinox> | + Pixel height: 1040 [12:46] <cytrinox> s/link/like [12:46] <Yulth> durandal_1707: What alternative do you suggest for? [12:46] <Yulth> rtsp? rtmp? [12:46] <durandal_1707> if Display width/height is part of container than it should be supported [12:46] <durandal_1707> Yulth: http, https, and so on, but not html [12:47] <Yulth> durandal_1707: ok [12:47] <luc4> Hi! I'm using the samples in the documentation to decode an aac stream, but I get something like this: ftp://luc4.homeip.net/output.pcm. I copy-pasted the code in the documentation and this is the result. Anyone who has any advice? [12:47] <cytrinox> durandal_1707: yes, display width/height properties are part of matroska. but I have not found any switch how to set these information manually [12:47] <Yulth> durandal_1707: really really grateful for your help. Thanks once again! See you! :D [12:49] <durandal_1707> cytrinox: this is just metadata or there is special handling in mkv? [12:51] <cytrinox> uh, I don't known how this is internally handled by matroska, but it can simply modified via mkvpropedit video.mkv --edit track:1 --set display-height=x [12:51] <cytrinox> but my intention was to automatically produce the correct file with ffmpeg [12:52] <durandal_1707> you mean the size was different from one in video stream? [12:52] <cytrinox> eh? [12:53] <durandal_1707> mkv have one video stream or several video streams, no? [12:53] <cytrinox> I mean that ffmpeg always write the same display size into mkv as the real pixel size [12:54] <cytrinox> and I want to have different (smaller) displays sizes [12:54] <cytrinox> afaik other containers supports different display sizes, too [12:55] <durandal_1707> paste ffmpeg output [12:57] <luc4> Anyone who knows whether there is some known issue in AAC decoder maybe? [12:58] <cytrinox> durandal_1707: what output? It's a theoretical question how to set different values for the display width/height properties directly with ffmpeg [13:00] <Mavrik> luc4: shouldn't be, make sure you have correct samplerate and channel number (+ layout) set when playing (and creating) the pcm file [13:02] <luc4> Mavrik: samplerate should be correct, 48000, there are 6 channels here, and this is correct, but for the moment I'm trying to write the PCM samples only from the first. I used the decoding_encoding.c sample. Maybe that is not ok for this kind of stream? [13:03] <durandal_1707> cytrinox: it is hardcoded, you could open bug report to make it configurable [13:04] <luc4> Mavrik: to be clear, I simply copy-pasted the decode sample method from that sample code and used on my aac stream. [13:04] <Mavrik> that should work perfectly [13:04] <Mavrik> I can't listen to your stream sadly (too slow download) [13:05] <Mavrik> but usually the problem is with wrong samplerate and/or channel layout [13:05] <luc4> Mavrik: I see this fwrite(decoded_frame->data[0], 1, data_size, outfile); [13:06] <luc4> Mavrik: shouldn't it be fwrite(decoded_frame->data[0], 1, decoded_frame->linesize[0], outfile); ? [13:08] <Mavrik> not in case of audio it seems [13:08] <Mavrik> the call "av_samples_get_buffer_size" tells you how many bytes the passed number of samples at passed parrameters take [13:09] <luc4> Mavrik: it is my understanding data_size is the size of the entire frame, but the frame should contain all the channels. [13:09] <luc4> Mavrik: anyway not working either. I can send you the decoded sample somehow, it is less than 1MB. [13:09] <Mavrik> data_size is the size of nb_samples samples when taking into account number of channels and sample format [13:10] <luc4> Mavrik: oh ok, so that is not the issue... [13:11] <luc4> Mavrik: that is the start of bug buck bunny from an aac stream. I can hear that audio more or less, but it is all messed up. [13:12] <durandal_1707> luc4: what is not working, what you get as output? [13:12] <luc4> durandal_1707: I placed a sample here ftp://luc4.homeip.net/output.pcm. I didn't know how to provide it. [13:13] <durandal_1707> aac decoder is changed to use planar sample fmt output [13:13] <durandal_1707> so if you use only data[0] you will get only 1st channel and it will be all messed up if you thing it is > 2 ch file [13:13] <durandal_1707> *think [13:13] <luc4> durandal_1707: oh... [13:14] <durandal_1707> you need to manualy interleave [13:14] <luc4> durandal_1707: oh... I'm sorry, it was my understand each channel was ok by itself... [13:15] <luc4> durandal_1707: so I can't take a 6 channels aac and play only 1 channel? [13:16] <durandal_1707> luc4: each channels is in separate plane for aac [13:17] <durandal_1707> so using only data[0] you should have only one channel if you are using enough recent library [13:18] <luc4> durandal_1707: yes, and forgive my ignorance :-) if I played only samples coming from data[0] as mono audio, should I expect a "correct" audio or not? [13:19] <durandal_1707> you should expect correct audio if decoder is using planar sample format [13:20] <luc4> durandal_1707: to be cleared, I used that sample code from the documentation to output to file only data[0]. Then I played that with aplay, specifying 48000Hz, 1 channel. Audio is like the one I provided. [13:20] <durandal_1707> aplay? [13:21] <luc4> durandal_1707: yes, I see that sample format for ffmpeg is float_planar: AV_SAMPLE_FMT_FLTP. [13:21] <luc4> durandal_1707: yes, alsa player. [13:22] <luc4> durandal_1707: the only thing that I'm not sure is that aplay is playing the samples as signed 32 bit little endian. ffmpeg instead outputs float. [13:22] <durandal_1707> that is the problem [13:23] <luc4> durandal_1707: ah ok! But do you have any advice on how to convert float to signed 32? [13:24] <luc4> durandal_1707: my complete code should output the audio using OpenMAX on embedded devices. But I don't seem to be able to play float with that, so I suppose I have to convert or ask ffmpeg to output something different. Is it possible? [13:25] <durandal_1707> yes, ffmpeg use aresample filter which use libswresample [13:26] <luc4> durandal_1707: I just tried to play the file using aplay output_ok.pcm -f FLOAT_LE -r 48000 -c 1. More or less same messed up audio. [13:28] <luc4> durandal_1707: it seems that with float only the volume is different. [13:29] <durandal_1707> how many channels original aac have? [13:29] <luc4> durandal_1707: 6 [13:31] <luc4> durandal_1707: it is simply a stream extracted from the big buck bunny video sample using ffmpeg. [13:32] <durandal_1707> how? / paste log output [13:32] <luc4> durandal_1707: of the extraction? [13:33] <luc4> durandal_1707: the aac stream seems correct. I can play it with other players and it is perfect. [13:35] <luc4> durandal_1707: this is the extraction output http://pastebin.com/BCZQdsZ7 [13:37] <durandal_1707> than you are either removing some samples or writing uninitialized data [13:38] <durandal_1707> perhaps you need to paste code you use [13:39] <luc4> durandal_1707: I'm surely doing something wrong, this is the second day I use ffmpeg :-) but I just copy-pasted the decoding_encoding.c function... [13:40] <luc4> durandal_1707: that sample was the output of the code from decoding_encoding.c. [13:42] <durandal_1707> luc4: make sure that you write exactly what you get [13:43] <durandal_1707> if you write only one channel samples than data_size should be divided by 6 [13:44] <luc4> durandal_1707: that is what I asked before to Mavrik... [13:45] <p131> Hi everyone ! [13:45] <p131> I'm playing with concat filter with no luck :( [13:46] <p131> ffmpeg -y -i 1264 -i "Sans nom-1.avi" -s 320x240 -filter_complex "[0:0],scale=320:240[first];[1:0],scale=320:240[second];[first][second]concat=n=2:v=1 [v]" -s 320x240 -map "[v]" out.mpg [13:46] <p131> => no such filter '' [13:46] <p131> I know it's a simple syntax issue, but i cannot get into it [13:46] <durandal_1707> ffmpeg -filters list concat? [13:47] <p131> yepo [13:47] <p131> The issue came from the moment i tried to chain the scale filter [13:47] <p131> (cause concat need files to be at the same res) [13:49] <p131> I must have made a dummy syntax error, but i dont understand where [13:49] <smooki> hi [13:50] <smooki> can ffmpeg do dts connect ? [13:50] <smooki> or dolby digital live ? [13:54] <luc4> durandal_1707: seems to work! But shouldn't decoded_frame->linesize[0] be equal to data_size/6.0? [13:55] <durandal_1707> it is not? [13:56] <luc4> durandal_1707: let me check the values precisely. [13:57] <smooki> anyone know maybe if any software can do DTS Connect please ? [13:59] <luc4> durandal_1707: linesize[0] is data_size/3.0. [14:00] <luc4> durandal_1707: also, since that function is writing down only the first channel, shouldn't it be changed to data_size/6.0? [14:01] <durandal_1707> luc4: example is broken/ needs some love to work with recent changes.... [14:03] <luc4> durandal_1707: ok, but what about linesize[0]? Shouldn't it be equal to data_size/6? [14:04] <durandal_1707> luc4: what if you use len/6 instead? [14:04] <luc4> durandal_1707: do you mean data_size/6? [14:04] <durandal_1707> no [14:05] <luc4> durandal_1707: ah the value returned by the decode function. [14:06] <luc4> durandal_1707: len == 1272. [14:08] <luc4> durandal_1707: I will use data_size/channels, but from the documentation it is my understanding linesize[0] should be equal to data_size/6. [14:18] <Fjorgynn> can the mp4 container contain subtitles? [14:18] <JEEB> 3gpp timed text, yes [14:19] <JEEB> or well, mpeg-4 timed text which is IIRC based on 3gpp timed text [14:36] <hub2> Hi,I made a program in C++ which encodes files to AAC but I got error "The encoder 'aac' is experimental but experimental codecs are not enabled, add '-strict -2' if you want to use it." and I don't know what to do. Have you any ideas how to solve it? [14:37] <Fjorgynn> libvo_aacenc [14:37] <Fjorgynn> hub2: that's the encoder [14:38] <Fjorgynn> codec whatever [14:41] <hub2> Fjorgynn:Yes I know that's about encoder but i don't know how to "add '-strict -2' " or solve it any other way? [14:42] <Fjorgynn> if you use -c:a libvo_aacenc you don't have to add some strict whatever [14:42] <Fjorgynn> why not add -strict -2 in the line and try otherwise? [14:43] <durandal_1707> it is C++ program [14:43] <Fjorgynn> so it's not using ffmpeg? [14:45] <hub2> in what line i should add it,it's a C++ program? [14:50] <Mavrik> hub2: if I understand correctly, you have a C++ encoder which calls ffmpeg libraries directly? [14:52] <hub2> Yes [14:53] <Mavrik> hub2: basically, the catch is AAC encoder in ffmpeg is still pretty experimental, that's why it's usually better to use fdk_aac, libvo_aacenc or libfaac [14:53] <Mavrik> hub2: the "proper" way to fix your issue is to recompile ffmpeg libraries with libfdk_aac enabled and then recompile the C++ program [14:54] <Mavrik> this will make it choose the external encoder [14:54] <hub2> Thanks i'll try that [14:55] <Fjorgynn> is libfaac better than libvo_aacenc, and how can I get it in Windows? :( [14:56] <Mavrik> hub2: the other option is to set your AVCodecContext->strict_std_compliance to FF_COMPLIANCE_EXPERIMENTAL [14:56] <Mavrik> Fjorgynn: no, it's not really [14:56] <Mavrik> Fjorgynn: use libfdk_aac if you can [14:56] <Mavrik> it's by far the best one [14:57] <Fjorgynn> aha [14:58] <Fjorgynn> yay, managed to get the size down to 5-6 MB /minute. [14:59] <Fjorgynn> Mavrik: okay, I have no idea what the difference is really [14:59] <Fjorgynn> between libfdk_aac and libvo_aacenc [15:00] <Mavrik> Fjorgynn: HE-AACv2 support, multi-channel support and the encoder itself it noticably better (better audio quality) [15:01] <Fjorgynn> okay [15:21] <Fjorgynn> seems like changing the sample rate from 48 to 44.1 doesn't make a lot of difference [15:27] <luc4> Hi! I'm currently playing a stream by reading the file (single stream) and passing data to the decoder. Can I simply extract encoded data so that I can pass that to a hardware decoder? [15:55] <JEEB> <Fjorgynn> if you use -c:a libvo_aacenc you don't have to add some strict whatever <- switch to fdk-aac if you only ca [15:55] <JEEB> *can [15:55] <JEEB> vo_aacenc isn't really better than the internal aac encoder [15:56] <JEEB> and yeah, I see Mavrik already noted that fdk-aac is preferred if you're going to use some library [16:01] <Fjorgynn> JEEB: so how can I do it on Windows? [16:01] <Fjorgynn> if I don't want to compile anything [16:02] <Fjorgynn> got the latest ffmpeg [16:03] <espr3ss0> hi again [16:03] <Fjorgynn> Mavrik: [16:14] <fatpony> i have a general question about two pass x264 encoding [16:15] <fatpony> should i include all options in the first pass? [16:15] <fatpony> like ref and subme? [16:16] <JEEB> Fjorgynn, fdk-aac seems to be not LGPL or GPL compliant license-wise so you'd indeed have to compile [16:16] <JEEB> (patent clause in the license) [16:18] <Fjorgynn> I also has besweet [16:18] <espr3ss0> Q: does anyone know if this works ? ... "http://letsneverdie.net/blog/?p=75" [16:18] <Fjorgynn> fatpony: what about 2 pass? [16:18] <JEEB> fatpony, uhh why are you setting specific x264 settings? [16:19] <JEEB> presets should be fine in like 90%+ of all cases [16:19] <fatpony> JEEB: i want a specific filesize for the output [16:19] <Fjorgynn> JEEB: and how do I compile it on Windows? [16:19] <JEEB> ...that is setting an average bit rate [16:19] <JEEB> those settings have NOTHING TO DO WITH FILE SIZE [16:19] <JEEB> (which you mentioned) [16:20] <fatpony> let's say i'm using two passes anyway [16:20] <fatpony> should i set the subme and so on parameters on the first pass? [16:20] <JEEB> well yes, that is just -preset yourpreset -pass 1 -b:v yourbitrate and then -preset yourpreset -pass 2 -b:v yourbitrate [16:21] <fatpony> okay [16:21] <JEEB> use preset and -pass 1 and -pass 2 [16:21] <JEEB> that way you get automated "fast first pass settings" with the first pass [16:21] <JEEB> while keeping everything else the same [16:21] <JEEB> :V [16:21] <fatpony> i've seen examples where both presets and ref/subme are specified [16:21] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset [16:21] <JEEB> list of presets for libx264 [16:22] <Fjorgynn> seems like I found something [16:22] <JEEB> fatpony, some people want to use specific values with specific presets, that's their choice -- I'm pretty sure you have no reason to blindly follow [16:22] <JEEB> setting ref can help with ffmpeg because libx264 doesn't limit reference frames to a certain level if you are setting it [16:23] <JEEB> so if you do -level 40 it will set level 4.0 in the header, but not actually limit the reference frame count [16:23] <JEEB> that's the only case I really think where setting some setting manually really matters [16:23] <JEEB> but if you are not setting a level [16:23] <JEEB> x264 will auto-decide based on your general settings [16:24] <fatpony> yeah i changed ref to respect level 4 [16:24] <fatpony> .1 [16:24] <JEEB> Fjorgynn, grab the latest git, autoreconf -fiv, see the configure script generated and then ./configure --enable-static --disable-shared --prefix=/your/prefix [16:24] <JEEB> yes, that makes sense [16:24] <JEEB> too bad libx264 doesn't do that automatically [16:24] <JEEB> only x264cli does [16:24] <fatpony> where are these presets stored? [16:24] <JEEB> inside libx264 [16:24] <JEEB> you access them with -preset [16:24] <fatpony> ah ok [16:24] <fatpony> so it's ok to use -preset slow -x264opts ref=4? [16:25] <JEEB> yes [16:25] <JEEB> preset sets the defaults [16:25] <JEEB> then you limit ref [16:25] <JEEB> just make sure you set the level then too I guess [16:25] <fatpony> oh yeah it's in my commandline [16:26] <JEEB> so yeah, it'd be something like -c:v libx264 -pass 1 -preset slow -level 41 -x264opts ref=4 and -c:v libx264 -pass 2 -preset slow -level 41 -x264opts ref=4 [16:27] <JEEB> the -pass 1 sets "fast first pass settings" [16:27] <fatpony> oh you set level through ffmpeg? [16:27] <fatpony> i added it to x264opts [16:28] <JEEB> yes, there's a level setting in ffmpeg as far as I know [16:28] <fatpony> i'll use that then [16:28] <JEEB> if I'm incorrect do note [16:29] <Fjorgynn> JEEB: easy for you to say [16:29] <espr3ss0> Q: anyone know what FAM is ? [16:29] <JEEB> Fjorgynn, possibly [16:30] <JEEB> you could use qtaac or neroaacenc via command line if you want [16:30] <JEEB> *qaac was it? [16:30] <JEEB> first needs QuickTime's certain components installed [16:30] <JEEB> and then mux the AAC stream into whatever you want [16:31] <fatpony> well thanks for your help JEEB, i'll let it encode now :) [16:56] <Fjorgynn> JEEB: can't do that [17:03] <JEEB> well then you don't encode to AAC :D [17:03] <JEEB> because all of your alternatives atm are crap [17:04] <Fjorgynn> JEEB: then why are there no aac alternatives already compiled? [17:04] <Fjorgynn> "home made" [17:04] <JEEB> because most of the good alternatives available are either not GPL or LGPL compatible, so you can't distribute builds with them [17:05] <JEEB> and the internal one lacks an active developer [17:05] <JEEB> if you want to sponsor the internal one's development, feel free. You'll just have to find an active maintainer for it [17:07] <JEEB> right now if you can't compile fdk-aac the easiest way is either to use QT's encoder via qaac or use nero's encoder via neroaacenc [17:07] <JEEB> you can pipe raw audio into both as far as I know [17:09] <Scorcerer> hello [17:14] <Scorcerer> is there a linux tool to softcode *.srt subtitles into an .avi container ? [17:15] <Scorcerer> i can use aviaddxsubs on windows, but it takes time, GUI, and can't be automated easily [17:16] <Fjorgynn> installed msys but yeah [17:17] <JEEB> that isn't really supported widely so I recommend you'd use a better container if you need to have subtitles as data in there Scorcerer [17:17] <Fjorgynn> read something about the ass filter [17:17] <JEEB> that's hardsubbing [17:17] <Fjorgynn> aha [17:17] <JEEB> he wanted to put the srt into the container [17:17] <JEEB> separate track [17:17] <Scorcerer> i only need it for PS3 [17:18] <JEEB> yeah, the PS3 seems to support it in some weird way [17:18] <Scorcerer> roght now in a directory i have an .avi movie, .txt subs and .rst subs [17:18] <Scorcerer> but thse separate files aren't streamed over DLNA [17:19] <Scorcerer> and it doesn't support .mkv [17:19] <Fjorgynn> mp4? [17:19] <JEEB> mp4 with timed text would work, but I have no idea if his solution works with that [17:19] <Scorcerer> it doesn't matter what codec i use, if its .mkv container it won't play [17:20] <Scorcerer> but i'm not 100% sure now [17:20] <JEEB> or if PS3 won't support timed text [17:20] <JEEB> also ffmpeg cannot convert other formats into timed text so you'd have to use GPAC (mp4box) for it [17:20] <JEEB> (other subtitle formats) [17:20] <Scorcerer> it's ok for now, as my ps3 is in repair, but i have another question [17:21] <Scorcerer> i have bunch of movies encoded in divx, and to play them on RaspberryPI i need to transode them into x264 [17:22] <Scorcerer> one episode weights ~350MB [17:22] <Scorcerer> is there easy way to transcode that ? [17:22] <Fjorgynn> git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg [17:22] <Fjorgynn> can't get this to work [17:22] <JEEB> you need git basically [17:22] <JEEB> you should be able to grab msysgit from its project [17:22] <JEEB> I usually grab the 7z [17:22] <JEEB> and extract it, and then add the /bin of it to the end of the PATH [17:22] <JEEB> (of msys shell) [17:23] <JEEB> (or you might just grab the installer and it'll handle global PATH addition for it) [17:23] <JEEB> http://code.google.com/p/msysgit/downloads/list [17:23] <Scorcerer> i have ffmpeg 0.10.6 installed on my media server (which runs on linux) [17:23] <JEEB> the ones that don't say msysgit and netinstall are what you want [17:23] <Scorcerer> isn't it enough ? [17:24] <JEEB> Scorcerer, if it has libx264 linked in, yes [17:24] <JEEB> also the format is H.264 [17:24] <Scorcerer> yes, it does [17:24] <JEEB> the encoder is called x264 [17:24] <Scorcerer> i also read i have to use 2-pass encoding if i want to getacceptable quality [17:25] <Scorcerer> but do i need to specify all options like HV size and others ? [17:25] <JEEB> eh [17:25] <JEEB> if you want good quality you EITHER use crf (quality-based) OR 2pass average bit rate [17:25] <JEEB> crf gives you somewhat stable set quality [17:26] <JEEB> average bit rate gives you an average bit rate over the whole video [17:26] <JEEB> both work from the same algorithm [17:26] <JEEB> so if you don't need specific size [17:26] <JEEB> you set crf [17:26] <JEEB> and one pass needed [17:26] <JEEB> also what do you mean with "HV size and others?" [17:27] <Scorcerer> like, horizontal/vertical size of input/output video stream [17:27] <JEEB> output is the same as input by default [17:28] <JEEB> unless you need to change that, there is no need to scale anything [17:29] <Scorcerer> ah, so it's ok to jus use x264 and set crf to medium, for, like, one episode, and if i'll find quality acceptable, run it on all others ? [17:29] <Scorcerer> wondering how long will it take to transcode one episode :) [17:29] <JEEB> just use -ss and -t [17:29] <JEEB> to seek to a certain part and then encode X seconds [17:30] <JEEB> 25-30 seconds of material should be fine for a test [17:30] <JEEB> do first like this: ffmpeg -i input -ss <Amount to seek> -t <Time to encode> -c:v libx264 -crf 23 -an out.mp4 [17:31] <JEEB> then if that looks good, push the crf value higher [17:31] <JEEB> if it looks bad, push it lower [17:31] <JEEB> by one or so [17:31] <JEEB> then just keep doing that until you have the highest crf value that still looks good to you [17:31] <klaxa> -ss is pretty slow though :S [17:31] <JEEB> oh right [17:31] <JEEB> you could set that before -i [17:31] <JEEB> it's faster that way [17:31] <JEEB> not exact, but faster [17:32] <klaxa> oh... if only i new that earlier :P [17:32] <JEEB> after that you can start testing -preset values [17:32] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset [17:32] <JEEB> medium is the default [17:32] <JEEB> find the slowest that is still fast enough for you [17:33] <JEEB> (everything but ultrafast and placebo should be "OK" to use) [17:33] <JEEB> (placebo and ultrafast are usually for benchmarks) [17:33] <Scorcerer> the truth is, i would want to keep quality as-is, but get similar or smaller filesize - for now, each episode has 350MB [17:33] <JEEB> then crf mode is the best for that [17:34] <JEEB> because it lets you select the highest crf value that still looks good [17:34] <JEEB> that way you can compress exactly as much as you can while keeping the quality you want [17:34] <Scorcerer> would like to try crf, as it's easier to batch-transcode many files, and i thuink it would be faster than 2-pass [17:34] <Scorcerer> ah, ok [17:34] <JEEB> average bit rate mode is made for exactly that -- when file size is more important than the quality [17:34] <Scorcerer> on to the tests then, thanks for help :) [17:35] <JEEB> crf is when you want to set the "quality" [17:35] <JEEB> and then the presets will help you tweak it [17:35] <JEEB> (compression level) [17:35] <JEEB> for the PS3 you also would end up using -level 41 and limit reference frames most probably [17:35] <JEEB> unfortunately libx264 doesn't do that automatically [17:36] <JEEB> the x264 command line encoder sets reference frame counts according to the resolution :< [17:36] <JEEB> (and level you set) [17:37] <Scorcerer> i think that's too much for me now, gotta do some trial-and error tests and then i'll be ready to learn some more :) [17:37] <JEEB> well, for now [17:37] <JEEB> just do what I noted up in the log [17:38] <JEEB> encode 25-30 seconds of content [17:38] <JEEB> with various crf values set [17:38] <JEEB> look at the results, see if it looks good enough [17:38] <JEEB> adjust crf value until you've found the highest that looks good enough for you [17:41] <Fjorgynn> If I download this source http://ffmpeg.zeranoe.com/builds/ [17:42] <JEEB> IIRC that hasn't been updated for quite a while [17:42] <JEEB> oh, it has [17:42] <JEEB> someone poked zeranoe [17:42] <Fjorgynn> 2013-01-04? [17:42] <JEEB> although I would still get the ffmpeg sources from git [17:42] <Fjorgynn> ;) [17:43] <Fjorgynn> JEEB: but then I must download all stuff also, not only x264 and aac, right? [17:43] <JEEB> just install/setup msysgit one way or another and be done with it :P [17:43] <JEEB> Fjorgynn, you'd have to in any other case too afaik [17:43] <JEEB> uhh, what [17:43] <JEEB> what exactly do you want to do with ffmpeg [17:44] <JEEB> if you just want to encode H.264 and AAC (and quite a few other formats), you only need yasm -> x264 -> fdk-aac -> ffmpeg [17:44] <Fjorgynn> convert stuff to H.264 with aac. [17:44] <JEEB> that's all you need then [17:44] <JEEB> yasm for assembler stuff [17:44] <JEEB> x264 for H.264 encoding [17:44] <JEEB> fdk-aac for AAC audio encoding [17:44] <JEEB> ffmpeg for... you know, ffmpeg :P [17:44] <Fjorgynn> yasm... hmm [17:44] <JEEB> yasm is simple to grab from the main site of theirs [17:45] <JEEB> you just need the yasm.exe to be in PATH [17:45] <JEEB> so you can call it from msys with 'yasm' [17:45] <Fjorgynn> never heard of yasm before [17:45] <JEEB> it's a rewrite of nasm [17:45] <Fjorgynn> never heard of that either [17:45] <JEEB> basically, assembly code to binaries [17:45] <JEEB> both x264 and ffmpeg EXTENSIVELY use assembly code [17:45] <JEEB> so you will want it [17:45] <JEEB> unless you wan to be slow [17:45] <Fjorgynn> and how do I include it in the ffmpeg? [17:46] <JEEB> just have yasm available from the msys prompt [17:46] <JEEB> put yasm.exe into the msys PATH [17:46] <JEEB> that's it [17:47] <JEEB> it's available from yasm's site [17:47] <JEEB> http://yasm.tortall.net/Download.html [17:47] <JEEB> <JEEB> you just need the yasm.exe to be in PATH [17:47] <JEEB> <JEEB> so you can call it from msys with 'yasm' [17:47] <Fjorgynn> and how do I put it in the PATH? I don't know something like that. Should be more documentation about it [17:48] <JEEB> ... [17:48] <JEEB> PATH is not something msys-specific [17:48] <JEEB> neither *nix specific [17:49] <JEEB> try 'echo $PATH' [17:49] <Fjorgynn> http://ffmpeg.org/trac/ffmpeg/wiki/MingwCompilationGuide [17:49] <JEEB> in msys's shell [17:49] <JEEB> it will show the PATH of your current msys shell [17:49] <JEEB> PATH is where the system (in this case msys shell) looks for binaries to run [17:50] <Fjorgynn> there are already yasm in the mingw msys stuff I am using :o [17:50] <JEEB> well that's great then [17:50] <JEEB> check yasm --version [17:50] <JEEB> to check the version [17:51] <Fjorgynn> 1.2.0 [17:51] <JEEB> great [17:51] <JEEB> up-to-date too [17:51] <Fjorgynn> http://pastebin.com/EdEsMUuS [17:51] <Fjorgynn> using this script thingy [17:51] <Fjorgynn> http://ffmpeg.org/trac/ffmpeg/wiki/MingwCompilationGuide [17:51] <Fjorgynn> and this [17:52] <Fjorgynn> or yeah not much that last but [17:52] <JEEB> ugh [17:52] <JEEB> forgets about --prefix [17:52] <Fjorgynn> googled it :/ [17:52] <JEEB> also memalign hack is no longer neede [17:52] <JEEB> *needed [17:52] <JEEB> w32threads are default now [17:52] <JEEB> can you pastebin what you get when you run 'ls -alh /mingw' ? [17:53] <JEEB> not paste on the channel, but pastebin and link here [17:53] <Fjorgynn> http://pastebin.ca/2299627 [17:54] <JEEB> I will guess that if you do gcc -v the default gcc will be i686-w64-mingw32, right? [17:55] <Scorcerer> what can i do when audio is going out-of sync with video (i'm using c:a copy right now)? [17:55] <Fjorgynn> something like that JEEB [17:55] <JEEB> ok [17:56] <Fjorgynn> Target: i686-w64-mingw32 [17:56] <JEEB> Fjorgynn, thus your default --prefix should be --prefix=/mingw/i686-w64ming32 [17:56] <JEEB> argh [17:56] <JEEB> forgot the second dash [17:56] <JEEB> --prefix=/mingw/i686-w64-mingw32 [17:57] <JEEB> now first of all grab x264 from git with git clone git://git.videolan.org/x264.git and it should create a folder called 'x264' into the current folder and start putting stuff into it [17:57] <JEEB> that is, if you now finally have git around :P [17:58] <Fjorgynn> libiconv-2.dll is missing [17:58] <JEEB> you failed to set up git properly [17:58] <Fjorgynn> will look after it [17:59] <JEEB> basically when you get x264's sources around, you do ./configure --enable-static --prefix=/mingw/i686-w64-mingw32 (see the i686-w64-mingw32 subfolder under /mingw , that is the folder for mingw stuff for that architecture, we will follow that) [18:00] <JEEB> oh right, you should also disable cli [18:00] <JEEB> ./configure --disable-cli --enable-static --prefix=/mingw/i686-w64-mingw32 [18:00] <JEEB> should configure x264 without the command line encoder, with a static library and install it into i686-w64-mingw32 [18:00] <JEEB> then you do 'make' [18:01] <JEEB> and then if that finishes correctly, 'make install' to install it into the prefix [18:02] <Fjorgynn> because now I have several folders [18:02] <Fjorgynn> directories [18:13] <Fjorgynn> so [18:22] <Fjorgynn> JEEB ERROR: libx264 not found [18:24] <Fjorgynn> will try again [18:25] <Scorcerer> JEEB: transcoding didn't stop after full movie length, it it problem with file or i messed something up? [18:25] <Scorcerer> i'll paste cmd, sec [18:26] <Fjorgynn> ah, seems to be git that was the problem there [18:26] <Scorcerer> ffmpeg -i movie.avi -c:v libx264 -preset slow -crf 22 -c:a copy test_22.avi [18:27] <Fjorgynn> Scorcerer: what's the problem with that? [18:28] <Scorcerer> quality was nice, filesize was really small (need to try with crf 18) audio was totally out-of-sync and it didn't stop transcoding after 1:36:xx which was end of movie... [18:29] <Fjorgynn> :/ [18:29] <Scorcerer> i used mplayer to check how long it is, and before i pushed q it was 1:52:21 :/ [18:34] <Fjorgynn> should work [18:34] <Fjorgynn> I've got a problem compiling ffmpeg myself [18:36] <Scorcerer> i don't think it's a problem witch ffmpeg itself, rather badly encoded file :/ [18:36] <JEEB> Fjorgynn, if you used the correct prefix with everything, then --extra-cflags="-I/mingw/i686-w64-mingw32" and --extra-ldflags="-L/mingw-/i686-w64-mingw32" should work for x264 [18:37] <JEEB> although if you have pkg-config around it'd be simpler to just set PKG_CONFIG_PATH=/mingw/i686-w64-mingw32/lib/pkgconfig before ./configure on the line (PKG_CONFIG_PATH=... ./configure --options) [18:39] <Fjorgynn> now I can configure ffmpeg atleast [18:39] <DJJeff> anyone know how to ffmpeg from .flv to .swf and keep same quality ? [18:40] <Fjorgynn> lol flash [18:40] <DJJeff> ? [18:40] <Fjorgynn> ffmpeg -i input.flv output.swf [18:40] <Fjorgynn> doesn't it work? [18:41] <DJJeff> it does but I loose alot of quality [18:41] <DJJeff> my 7MB flv becomes 1MB swf [18:41] <DJJeff> the flv bitrate was 5000 the swf cant go higher then 1500 [18:42] <DJJeff> not sure why [18:42] <Fjorgynn> $ make [18:42] <Fjorgynn> common.mak:138: *** missing separator. Stop. [18:42] <Fjorgynn> JEEB: :( [18:42] <JEEB> hmm [18:42] <JEEB> inb4 CRLF makefiles [18:42] <JEEB> turn off autocrlf in git and re-clone [18:42] <Fjorgynn> what? [18:43] <Fjorgynn> it is ffmpeg [18:43] <JEEB> yes [18:43] <JEEB> but that sounds like a usual problem conceived by having the sources cloned with Windows CRLF end lines [18:43] <JEEB> and not just LF *nix end lines [18:43] <JEEB> I'm currently trying to grab the github help page regarding it from my browser's history [18:43] <JEEB> https://help.github.com/articles/dealing-with-line-endings#platform-windows [18:44] <JEEB> there [18:44] <JEEB> you want to set autocrlf to 'false' [18:44] <JEEB> not true [18:44] <Fjorgynn> how? [18:44] <JEEB> open the link I provided? [18:44] <JEEB> it's the first command line noted in there [18:44] <JEEB> as far as I can see [18:45] <Fjorgynn> git config --global core.autocrlf false [18:45] <JEEB> yes [18:45] <JEEB> Then see the "Re-normalizing a repo" [18:45] <JEEB> part [18:45] <JEEB> so you don't have to re-clone [18:46] <Fjorgynn> okay [18:47] <JEEB> except for the commit part of course [18:47] <JEEB> you don't need to do that [18:47] <JEEB> you just need to do the first two commands there [18:47] <Fjorgynn> okay [18:48] <Fjorgynn> now I'll configure and make again [18:48] <Fjorgynn> DJJeff: can't you set the -b:v 5M ? [18:50] <Fjorgynn> And we have a winner! [18:53] <Scorcerer> what does that mean: Switching to NI mode, due to poor interleaving ? [18:53] <JEEB> uh-oh [18:53] <JEEB> can you pastebin the whole log of that file onto a pastebin provider, and then link it here? [18:54] <Scorcerer> sure, one sec [18:55] <Scorcerer> http://pastebin.ca/2299655 [18:56] <JEEB> welp [18:56] <Scorcerer> that is for other file, audio desync persists here, dunno about going beyond EOF [18:57] <JEEB> it isn't going beyond EOF [18:57] <JEEB> that file is just... [18:57] <JEEB> you might want to pass it through virtualdub or so in the stream copy mode for both video and audio [18:57] <Scorcerer> so it's broken ok :) [18:58] <JEEB> the first hint is [18:58] <JEEB> [avi @ 0x634fb0] scale/rate is 0/0 which is invalid. (This file has been generated by broken software.) [18:58] <JEEB> then you've set the audio bit rate way too low [18:58] <JEEB> remember that bit rate is set in bits in ffmpeg [18:58] <JEEB> not in kilobits [18:59] <JEEB> you were setting it to 192 while I think you meant 192k [18:59] <Scorcerer> didn't know that, thanks [18:59] <Scorcerer> yup [18:59] <Scorcerer> anyway [18:59] <Scorcerer> do i have to use virtualdub, or is there linux-cmd tool to do this ? [18:59] <JEEB> anyways, you might want to pass such files through something like vdub or possibly nandub depending on how broken they are [19:00] <JEEB> well, 1) vdub has command line interface and 2) it works under wine [19:00] <JEEB> so you can use it under linux [19:00] <JEEB> if you want [19:00] <hotwings> is ffmpeg git down at the moment? [19:00] <Scorcerer> don't have windows here, except for VM [19:00] <Scorcerer> ah, ok [19:00] <Scorcerer> i'll try, thanks [19:00] <JEEB> see the virtualdub documentation basically I guess [19:00] <JEEB> hotwings, seems up for me http://git.videolan.org/?p=ffmpeg.git [19:00] <JEEB> at least the http part of it [19:01] <Fjorgynn> ... [19:01] <hotwings> getting fatal: unable to connect to source.ffmpeg.org: [19:01] <hotwings> with git clone [19:01] <JEEB> hotwings, that domain leads to the videolan repo anyways [19:01] <JEEB> so use the videolan url [19:02] <JEEB> probably borked DNS entry or something [19:02] <Fjorgynn> now I must have libx264-129.dll in the same directory [19:02] <Fjorgynn> but whatever [19:02] <hotwings> ok thanks JEEB [19:02] Action: Fjorgynn hugs JEEB *thanks* [19:02] <JEEB> Fjorgynn, you did --enable-shared and not static did you not? [19:02] <JEEB> with x264's configure [19:03] <JEEB> or you already had a libx264.dll.a in the library lookup path [19:03] <JEEB> which is the shared library [19:03] <JEEB> if you have libx264.a and libx264.dll.a in the same folder, remove the .dll.a one [19:03] <JEEB> to make sure it's not used [19:05] <Fjorgynn> aha [19:09] <Fjorgynn> I see [19:10] <Fjorgynn> JEEB: is it presets and the ffmpeg.exe I need? [19:10] <Fjorgynn> or just the ffmpeg.exe? [19:10] <JEEB> usually you don't even need the ffpreset files [19:10] <Fjorgynn> :) [19:10] <JEEB> because libx264's presets are inside it [19:10] <JEEB> so they're within the binary [19:11] <Fjorgynn> and with this build I can convert avi mpeg4 to x264 (h.264) ? [19:11] <Fjorgynn> now I must learn how to use the fdk-aac :o [19:12] <Fjorgynn> now it compiles my new binaries :D [19:14] <JEEB> yes, most if not all decoders are in ffmpeg itself [19:14] <JEEB> and you just need to encode H.264 [19:14] <JEEB> and you have libx264 for that [19:14] <JEEB> and fdk-aac for aac [19:17] <Fjorgynn> is it -q:a 1-5 instead of -b:a 128k I should use now or what? [19:17] <JEEB> whichever you want [19:17] <Fjorgynn> I see [19:17] <Fjorgynn> I've noticed that youtube and vbr isn't friends :( [19:20] <JEEB> shouldn't have anything to do with that [19:21] <Fjorgynn> hmm it still complains about that file... [19:22] <Fjorgynn> but I recompiled x264 and ffmpeg [19:22] <Fjorgynn> ./configure --disable-cli --enable-static --prefix=/mingw/i686-w64-mingw32 [19:22] <JEEB> that's fine, now you should have libx264.a in /mingw/i686-w64-mingw32/lib [19:23] <JEEB> and x264.h in /mingw/i686-w64-mingw32/include [19:23] <DJJeff> im really having a hard time with flv to swf quality here is my output [19:23] <DJJeff> http://pastie.org/pastes/5627484/text [19:23] <Fjorgynn> JEEB: and? [19:24] <JEEB> do you have any other libx264.something files in either /mingw/lib or /mingw/i686-w64-mingw32/lib? [19:24] <JEEB> ending with .a [19:24] <JEEB> esp. libx264.dll.a [19:25] <Fjorgynn> yes in the i686-w6... [19:26] <Fjorgynn> not any more [19:27] <JEEB> you should only have your own libx264.a which is the static library around, basically [19:31] <Scorcerer> i'll forgo virtyaldub and try use avidemux :) [19:31] <JEEB> avidemux... I've seen nothing good from it [19:31] <JEEB> if you just need to remux [19:31] <JEEB> use vdub or possibly nandub [19:32] <JEEB> it'll take care of most of those hacky files if it is possible [19:34] <Scorcerer> do i need decompressor for usign direct stream copy mode then ? [19:34] <Scorcerer> or just load file, set mode to direct stream copy andt save ? [19:38] <Fjorgynn> JEEB: one problem [19:38] <Fjorgynn> ./configure --disable-cli --enable-static --prefix=/mingw/i686-w64-mingw32 [19:38] <Fjorgynn> if I do that (redownloaded x264) I get an error saying that libx264 isn't foud [19:38] <Fjorgynn> when I try to configure ffmpeg [19:39] <Fjorgynn> so yeah [19:40] <JEEB> Fjorgynn, how do you configure ffmpeg :P [19:41] <JEEB> your x264 configuration seems just fine [19:41] <DJJeff> ffmpeg has hard time going from flv to swf is there other programs that work better in ubuntu/debian ? [19:41] <Fjorgynn> ./configure --enable-libfdk-aac --enable-libx264 --enable-gpl --enable-version3 --enable-nonfree [19:42] <Fjorgynn> http://pastebin.ca/2299663 [19:42] <Fjorgynn> x264 [19:42] <JEEB> PKG_CONFIG_PATH=/mingw/i686-w64-mingw32/lib/pkgconfig ./configure --enable-libfdk-aac --enable-libx264 --enable-gpl --enable-version3 --enable-nonfree --extra-cflags="-I/mingw/i686-w64-mingw32/include" --extra-ldflags="-L/mingw/i686-w64-mingw32/lib" [19:42] <JEEB> try this [19:43] <sacarasc> --disable-cli? :o [19:43] <JEEB> he doesn't need the x264 cli [19:43] <JEEB> only the library [19:44] <sacarasc> Oh, that's x264's configure. [19:44] <sacarasc> I got confused, sorry. [19:45] <Scorcerer> JEEB: looks like VirtualDub worked (at least on one file - testing in ffmpeg now) - no warnings in ffmpeg so far. [19:47] <Scorcerer> but audio is still out of sync :/ [19:48] <Fjorgynn> same problem JEEB [19:48] <JEEB> pastebin config.log [19:51] <Scorcerer> JEEB: Streams in files are numbered from 0 or from 1 ? [19:51] <Fjorgynn> JEEB: http://pastebin.ca/2299666 [19:51] <JEEB> Scorcerer, yes [19:51] <JEEB> from zero [19:51] <JEEB> both open inputs and streams in inputs [19:52] <Scorcerer> ok, so then the error i got from VirD is from subtitle stream, nothing to worry about then :) [19:52] <Scorcerer> "AVI: Stream 2 has an invalid sample rate. Substituting 15 samples/sec as a placeholder [19:53] <Scorcerer> " [19:53] <JEEB> Fjorgynn, ugh seems like x264 chose pthreads and not win32threads... since it was available. [19:53] <JEEB> --enable-win32thread [19:53] <JEEB> add this to x264's configure [19:54] <JEEB> to make it use standard win32 threads [19:54] <JEEB> for some reason if x264 can find pthreads it will go there instead of just keeping to win32threads by default [20:01] Action: Fjorgynn gives JEEB a big kiss [20:02] <Fjorgynn> ffmpeg version N-48482-g6717d1a Copyright (c) 2000-2013 the FFmpeg developers built on Jan 5 2013 19:19:10 with gcc 4.7.2 (GCC) configuration: --enable-libfdk-aac --enable-libx264 --enable-gpl --enable-vers [20:02] <Fjorgynn> ion3 --enable-nonfree [20:04] <DJJeff> avi to mp4 is pretty impressive in terms of quality and filesize [20:07] <Fjorgynn> yeah [20:07] <Scorcerer> am i wrong, or avi is just a container in which you can store anything you want (like x264)? [20:08] <sacarasc> Yes, that is correct. [20:08] <sacarasc> Well, not anything, but some things. [20:09] <Fjorgynn> why are people using libx264 with a i container? [20:09] <Fjorgynn> AVI [20:09] <Fjorgynn> that is [20:10] <Scorcerer> for me, it's just that i have every movie in an avi [20:10] <Scorcerer> what container it should be stored in, then? mp4 ? [20:11] <sacarasc> Whatever you want, but don't reencode for reencoding's sake. [20:12] <Fjorgynn> all new stuff I make are mp4 [20:17] <JEEB> AVI + B-frames is a very bad combo (be it MPEG-4 Part 2 or H.264) [20:17] <JEEB> for MPEG-4 Part 2 (divx/xvid etc.) AVI ended up being the "de facto" container [20:18] <JEEB> but at least with H.264 people switched to other containers [20:18] <Scorcerer> so the test encoding ended, it looked like this : http://pastebin.ca/2299669 [20:19] <Scorcerer> i had to manually stop it after i checked it's legth which was 2:10:xx, instead of stopping after, like, 1:36:xx [20:19] <Scorcerer> :( [20:20] <Scorcerer> audio was not only not synced, but also fast played from beginning after every skip (forward/backward) [20:20] <Scorcerer> but video was great :) [20:20] <Scorcerer> not sure what to do now :/ [20:27] <Fjorgynn> Scorcerer: copy the audio stream? [20:28] <Fjorgynn> I dunno [20:29] <Scorcerer> tried that, didn't work :( [20:30] <Scorcerer> for now, i'll use virtualDub on all files [20:30] <Scorcerer> and then i'll think what to do :) [20:34] <Fjorgynn> btw flac seems to cut wav files in hal [20:34] <Fjorgynn> half [20:40] <Scorcerer> still won't stop after normal end of video :/ [20:41] <JEEB> it is probably getting the packets and how long the whole thing is supposed to be wrong [20:42] <JEEB> nandub /might/ be needed for VBR audio in avi [20:42] <JEEB> (welcome to the garden of hacks that is "normal stuff in AVI" [20:42] <Scorcerer> that's why i'm migrating wrom this divx %#*%^ [20:46] <Fjorgynn> is there a way to se what -q:a I should have? [20:46] <Fjorgynn> for aac? [20:49] <JEEB> "I should have"? [20:51] <espr3ss0> Q: i saw when looking through some web-pages but have forgotten what ... do i need to with 'make' "install-lib / install-man" for any of the following ? ... "gettext / glib / gpac / libiconv / pkg-config / x264" [20:52] <Fjorgynn> JEEB: what I understand there is 1-5, what does that mean and how do I find out about that? [20:53] <JEEB> no idea (Ž [20:53] <Fjorgynn> -q:a seems not to work at all [20:55] <Scorcerer> nandub didn't help :( [20:55] <Scorcerer> damn [20:55] <JEEB> can you extract the audio track from the avi? [20:56] <JEEB> avconv -i derp.avi -vn -c:a copy out.ac3 if it's ac3 [20:57] <Scorcerer> i don't have avconv yet, sec [20:57] <JEEB> argh [20:57] <JEEB> ffmpeg I mean [20:57] <JEEB> sorry, getting tired :P [20:57] <JEEB> (as well as multitasking) [20:57] <Scorcerer> sure, no problem :) [20:58] <Scorcerer> that's strange [20:58] <Fjorgynn> -vbr seems to be good [20:58] <Scorcerer> now time= was equal to total length of movie [20:59] <Scorcerer> and earlier (when transcoding all at once) it wasn't [20:59] <Scorcerer> it extracted ok [21:00] <Scorcerer> maybe i should extract video stream too ? [21:04] <JEEB> nah [21:04] <JEEB> just encode the video (-an is "audio none") [21:05] <JEEB> or wait [21:05] <Scorcerer> i'm doing it now, into out.avi [21:05] <Scorcerer> done [21:05] <Scorcerer> and time wal also good (time=01:36:37) [21:05] <Scorcerer> was* [21:05] <JEEB> ffmpeg -i input.avi -i out.ac3 -map 0:1 -map 1:1 -c copy out.mp4 [21:06] <JEEB> ugh [21:06] <JEEB> -map 0:0 and -map 1:0 [21:06] <JEEB> I meant [21:06] <Scorcerer> one sec [21:06] <JEEB> that should pick the video from the avi, and audio from the ac3 file [21:08] <Scorcerer> [mp4 @ 0x645630] track 0: could not find tag, codec not currently supported in container [21:08] <Scorcerer> shouldn't it be 0:0 and 0:1 ? [21:08] <JEEB> pastebin full log [21:08] <JEEB> and no [21:08] <JEEB> because you have two inputs [21:09] <Scorcerer> ah, right [21:09] <JEEB> so basically the first track from inputs 0 (probably the video) and 1 (the ac3 file) [21:09] <Scorcerer> http://pastebin.ca/2299681 [21:10] <JEEB> ok [21:10] <JEEB> msmpeg4 is not ok into mp4 [21:10] <JEEB> try dot-mkv [21:10] <Scorcerer> ok, now it works [21:11] <Scorcerer> ok, done with right time again [21:11] <JEEB> yeah, taht thing just meant that the format X can't be put into mp4 [21:11] <JEEB> *that [21:11] <Scorcerer> now transcode in x264 ? [21:11] <Scorcerer> to* [21:11] <JEEB> if that mkv is fine, you can transcode with those two inputs :) [21:13] <Scorcerer> no video in mplayer [21:13] <Scorcerer> one sec [21:13] <developer> can ffmpeg encode video with srt or ass subs from mkv ? [21:16] <sacarasc> developer: Yes. [21:16] <sacarasc> -vf ass, IIRC. [21:16] <JEEB> not sure if that could load the subtitles straight from the file [21:17] <JEEB> (as in, muxed stream from input) [21:17] <JEEB> if yes, great [21:18] <Scorcerer> JEEB: no video output from .mkv, but i'm transcoding anyway, mplayer spits out smth like this: http://pastebin.ca/2299684 [21:19] <developer> would be really good to just encode with the subs already from mkv instead of muxin out the tracks and remuxin :) [21:19] <JEEB> Scorcerer, looks like mplayer just fails at playing it :P [21:19] <Scorcerer> yeah, i think so too [21:20] <Scorcerer> one more question, what does "time=xx:xx:xx.xx" mean? place in file where ffmpeg is now ? [21:26] <espr3ss0> Q: can mp4v2 be used instead of gpac ? [21:27] <sacarasc> I don't think you need either for ffmpeg. [21:29] <espr3ss0> ok [21:29] <espr3ss0> thanks [21:37] <spaam> ubitux: does your HDS demuxer handle encrypted streams? [21:42] <durandal_1707> HDS? [21:42] <spaam> durandal_1707: HTTP Dynamic Streaming [21:42] <spaam> durandal_1707: Adobe thing :) [21:47] <durandal_1707> first it is not demuxer and second it is not ubitux demuxer [21:52] <durandal_1707> and ubitux did not do much with it because company for which he worked lost interest for hds [21:53] <spaam> ok :) [22:19] <doublehp> i can fully read a file with mplayer and vlc, including ST, but ffmpeg complains: Decoder (codec text) not found for input stream #0:3 [22:22] <doublehp> http://pastebin.ca/2299687 [22:24] <doublehp> how is it possible that both players can read it fine, but ffmpeg won't find the codec ??? [22:25] <doublehp> it may be a broken dep in Gentoo, but, i don't know which one [22:25] <JEEB> I'm /pretty/ sure you didn't mean to re-encode there [22:26] <JEEB> and that just means that it can't convert from srt to ass subtitles [22:27] <doublehp> JEEB i did not start yet to work on the recoding aspect; i do want to recode, to make file smaller; any help welcome :) I wanna keep the 2 audios and the subs [22:28] <JEEB> -map 0 maps all tracks from input 0 (the first -i you have), otherwise only the "best" track will be grabbed of both video/audio/subtitles into output. If you want to copy audio and subtitles, set -c:a copy -c:s copy [22:28] <JEEB> and I don't really know what you mean with "make the file smaller". The input already seems to be an x264-encoded H.264 stream [22:29] <JEEB> I mean, you won't get it much smaller without losing quality unless you resize it down [22:30] <doublehp> input is 16GB; i want output to be around 1GB [22:33] <JEEB> How that will look will depend on what kind of content there is in that movie then (its compression complexity), as well as how well it was originally already compressed. The x264 wording in there means that someone most probably already once has encoded that from a source with x264 [22:33] <JEEB> I recommend you use -ss and -t to seek to a certain part that actually has content, and then encode 25-30 seconds worth of content with -c:v libx264 -crf 23 [22:33] <JEEB> crf 23 is also the default for libx264 [22:34] <JEEB> (-ss is seek, and -t is "how much to encode time-wise, basically") [22:34] <JEEB> if the output looks good, you make the crf value higher [22:34] <JEEB> if it looks bad, make it somewhat lower [22:35] <JEEB> do that for a while and you will find the highest crf value that still gives you quality that you like [22:35] <JEEB> after that you can start setting up the preset, all of them are listed http://mewiki.project357.com/wiki/X264_Settings#preset <- here, and they are set with the -preset setting [22:35] <JEEB> medium is the default [22:36] <JEEB> slower presets = better compression [22:36] <JEEB> you can then try a few and see what is the slowest that is still fast enough for you [22:36] <JEEB> after that you can try encoding with those settings [22:36] <JEEB> if it will still end up too big, then you have to resize [22:37] <JEEB> but that's it so far [22:38] <JEEB> (you could just set the bit rate to a certain size, but that would prioritize "I want this size!" instead of quality [22:38] <JEEB> (and for bit rate based encoding you would need two passes) [22:38] <doublehp> JEEB it added a second new sub .... [22:38] <JEEB> haha, do show the log [22:39] <JEEB> (with the command as well as output) [22:40] <JEEB> also btw, I do recommend you build a newer ffmpeg :) Esp. if you're in an environment like gentoo where you have to build your stuff yourself anyways [22:41] <doublehp> http://pastebin.ca/2299689 [22:41] <Scorcerer> JEEB: it works now, splitting audio and video seems to do the trick [22:41] <JEEB> Scorcerer, seems like the audio track demuxing barfed and trying to extract it separately first was a good idea :) [22:42] <JEEB> doublehp, I see only one subtitle track? [22:42] <JEEB> or wait [22:42] <doublehp> one in input is fine; but mplayer founds two in output [22:42] <JEEB> uhh [22:42] <JEEB> Stream #0:3(fre): Subtitle: text (default) Stream #0:4(fre): Subtitle: text [22:43] <JEEB> you have two subtitle tracks in the damn input! [22:44] <doublehp> won't be the worst issue even in a 1GB file; forget it. [22:44] <JEEB> in other words, it didn't /add/ any subtitle tracks [22:44] <JEEB> there were two to begin with [22:44] <JEEB> and yes [22:44] <JEEB> subtitle tracks don't really matter [22:44] <JEEB> they are really small in comparison to pretty much anything else [22:45] <JEEB> unless you are putting base64-encoded data there [22:45] <doublehp> i need to crop top and bottom, and then, resize to, let say, twice smaller [22:45] <Scorcerer> JEEB: yup, that's the way to do it, but i still have questions, fo egzample: do i have to provide .ac3 for ac3 sound and mp3 if it will be encoded im mpeg2layer3, or there is some container i can use to store it in? [22:45] <Scorcerer> universal container* [22:45] <JEEB> mkv I guess? [22:46] <Scorcerer> ah, ok [22:46] <Scorcerer> i would like to create one for loop to transcode everyfile in directory over, say a week :) [22:46] <JEEB> doublehp, http://ffmpeg.org/ffmpeg-filters.html#crop and http://ffmpeg.org/ffmpeg-filters.html#scale [22:46] <JEEB> both have examples, too :) [22:47] <doublehp> found -vf crop= [22:47] <JEEB> yes, that's the documentation for it that I just linked [22:47] <JEEB> the other filter is the scale filter [22:47] <JEEB> you first crop, then scale [23:03] <DJJeff> im really having a hard time with flv to swf quality here is my output [23:03] <DJJeff> http://pastie.org/pastes/5627484/text [23:06] <doublehp> my movie is 1920x800, that's 2.4; which smaller sizes respect this ratio ? [23:07] <doublehp> 1152 480 [23:19] <klaxa> 940x400? [23:19] <klaxa> uh... [23:19] <klaxa> 960x400 [23:19] <doublehp> yup, could do [23:20] <doublehp> but i think that 1152 is a standard for laptops; never saw 960x400 as native size for any screen [23:22] <klaxa> yeah generally using 480 for horizontal pixels is a good value (i guess?) [23:23] <doublehp> hmmm [23:23] <doublehp> you are right , with 960x400, interpolation will be nicer [23:24] <klaxa> well it's just half then horizontal size [23:25] <doublehp> now; how do i copy audio 1 ; but recode audio 2 ? [23:25] <klaxa> look at -map [23:26] <hub2> Hi,Is there a C function that checks how many frames left in queue? [23:29] <doublehp> this soft is wonderfull :) <3 [23:48] <doublehp> JEEB sub 0 is a fake empty one; i am removing it. thanks for help. I have written every thing in a text file for future use. I have spent long time on it, not just for this file, but wrote my tuto for next time :) [23:49] <doublehp> i think that, to see the difference between a 16GB and 1GB file, i would need a 42" screen; i only got 19" :P [23:50] <sacarasc> I think you need an eye test. :p [00:00] --- Sun Jan 6 2013

1 0

ffmpeg-devel.log.20130105
by burek 16 Jan '13

16 Jan '13

[02:44] <llogan> lol. someone seems to want a "porn" filter in #ffmpeg. [02:52] <Compn> to filter it out, or to add porn to every fram e? [02:52] <Compn> youtube has a porn filter i bet [02:53] <Compn> not sure if its manual slave labor, could be [02:53] <llogan> i don't know. he's gone. [03:31] <iive> Compn: you would need at least as many people to screen the videos, as the people who upload them... [03:31] <iive> it must be automated. to some degree. [04:50] <Compn> yes, i guess some kind of skin color detection [04:50] <Compn> more skin = more flags [05:34] <cone-241> ffmpeg.git 03Matthieu Bouron 07master:bbab9cceb953: lavfi/drawutils: fix typo [09:22] <cone-899> ffmpeg.git 03Carl Eugen Hoyos 07master:c52e07bb6e91: Fix AVCI50 SPS to specify a SAR of 4:3 instead of 3:4. [10:31] <saste> michaelni: libavutil/common.h:100:34: warning: "ASSERT_LEVEL" is not defined [-Wundef] [10:34] <cone-899> ffmpeg.git 03Stefano Sabatini 07master:92f1bed14cef: tests/lavfi-regression: always require three parameters in do_lavfi_pixfmts() [10:34] <cone-899> ffmpeg.git 03Stefano Sabatini 07master:172505b8bc36: lavfi: add kerndeint filter [10:34] <cone-899> ffmpeg.git 03Stefano Sabatini 07master:014056635944: lavfi: add histeq filter [10:38] <cone-899> ffmpeg.git 03Paul B Mahol 07master:d4211c47220c: alsdec: change channel sorting so it match reference implementation [10:54] <durandal_1707> does libswresample supports dithering from 24/32 to 16bps? [11:08] <ubitux> does anyone know how the "return meaningful error codes." will affect the move to lavu AVFrame and ref counter? [11:08] <ubitux> because if it's important, we'd better start doing it our own stuff too [11:09] <ubitux> saste: going to remove mp=kerndeint? [11:09] <nevcairiel> ubitux: ask the master of the evil plan? :) [11:10] <ubitux> i'm going to get stabbed [11:10] <nevcairiel> thats a risk, true [11:11] <wm4> more like the evil api breakage [11:14] <ubitux> mateo`: your drawutils blending patch doesn't affect fate? [11:16] <michaelni> saste, http://fate.ffmpeg.org/report.cgi?time=20130105100534&slot=x86_64-darwin-gc… [11:16] <michaelni> kerndeint failure on ppc [11:17] <ubitux> [scale @ 0x1015136c0] w:iw h:ih flags:'' interl:0 [11:17] <ubitux> i guess that's sws flags again :) [11:24] <mateo`> ubitux: it seems it does not, i ran fate and everything goes ok [11:24] <ubitux> so how do you reproduce the issue? [11:25] <ubitux> it seems that part of the code is completely uncovered [11:25] <ubitux> http://lucy.pkh.me/ffmpeg-coverage/ffmpeg/libavfilter/drawutils.c.gcov.html [11:27] <ubitux> oh, it's in drawtext? [11:27] <mateo`> use drawtext with box=1:boxcolor=0x000000@0.8:x=101, wrong colors should appear at the left/right of the box. [11:27] <ubitux> oh it noticed that i remember, indeed. [11:27] <mateo`> the problem is even more visible with low subsampling like yuv411p [11:28] <ubitux> i wonder if we couldn't find font settings where it renders the same everywhere [11:28] <mateo`> might be a good idea to add a fate test ? [11:28] <ubitux> some kind of standard monospace font without aliasing [11:28] <ubitux> mateo`: the problem is making the test portable [11:29] <ubitux> font stuff can certainly behave totally differently between platforms [11:29] <ubitux> maybe we can just print some spaces, but i'm not even sure the surface will have the same dimensions everywhere [11:29] <ubitux> (and the test won't be really useful) [11:31] <mateo`> ubitux: but the problem will always show up at the left side [11:31] <mateo`> ubitux: whatever font you use [11:31] <ubitux> yes, for that particular thing [11:31] <ubitux> but if we're going to cover drawtext [11:32] <ubitux> better make the test a little useful [11:32] <ubitux> +more [11:32] <mateo`> sure [11:33] <saste> michaelni, why is sws bitexact flag not set? [11:35] Action: durandal_1707 :( i get spam : ./libavutil/common.h:225:34: warning: 'ASSERT_LEVEL' is not defined, evaluates to 0 [-Wundef] [11:36] <saste> how are sws_flags propagated to the filtergraph? [11:36] <saste> i see that sws_flags is set in the ffmpeg commandline, but apparently it is not propagated to the filterchain [11:37] <saste> also i'd expect to see many failures in that case, not only in kerndeint [11:37] <saste> apart from that, i can't see much platform dependent code which may affect the result of the test [11:38] <durandal_1707> floats [11:40] <wm4> does it even make sense to have different swsflags for different things? [11:40] <saste> durandal_1707, no floats are used with those parameters [11:45] <saste> ubitux, i'll remove kerndeint if michaelni is fine with that [11:45] <saste> as for me, i see no reason to keep it [11:45] <ubitux> perf! :) [11:45] <durandal_1707> if it is faster than native, i object [11:46] <saste> durandal_1707, that's going to be pretty ridiculous [11:46] <saste> same for tinterlace, it is broken and the absolute gain in speed is unnoticeable [11:47] <ubitux> the question is to understand why it happens, and if it's fixable [11:47] <durandal_1707> if it is > 5% faster than native keep it [11:47] <saste> then since it basically only performs stride arithmetic, it is faster since the native tinterlace is more generic/less redundant [11:48] <saste> relative efficiency is meaningless in that case [11:48] <durandal_1707> saste: i do not talk about tinterlace [11:48] <ubitux> do you have the same problem with kerndeint? [11:48] <durandal_1707> if tinterlace is different filter, shure remove mp one [11:48] <ubitux> (note: i'm also fine with removing mp=tinterlace) [11:51] <saste> ubitux, i'm not very motivated into benchmarking mp=kerndeint/kerndeint [11:51] <saste> i wasted already a lot of time with tinterlace [11:52] <ubitux> :) [11:53] <durandal_1707> really, it only takes to add 2 lines of code [11:53] <saste> durandal_1707, it's at least three lines of code, in two different files [11:53] <saste> so it is six lines of code in total [11:54] <ubitux> :D [11:55] <ubitux> doesn't sound like worth the effort! [11:55] <ubitux> except when you know that you'll have to remove them aftwerward [12:03] <durandal_1707> michaelni: 697b476c075a and ticket mentioned in log have nothing in common [12:06] <cone-899> ffmpeg.git 03Paul B Mahol 07master:14d50c19dc56: w64dec: support metadata (summarylist guid) [12:22] <cone-899> ffmpeg.git 03Michael Niedermayer 07master:f27eb1b702d8: lavu: check that assert level is defined [12:22] <durandal_1707> michaelni: so there is case when assert level is defined? [12:23] <durandal_1707> there is no way to fix circular shit? [12:43] <michaelni> durandal_1707, IIRC fixing the circular stuff wasnt trivial without introducing lots of changes to libav that would make merging harder, but its a while ago that i wrote the patch [13:05] <J_Darnley> fflogger: If you're just going to state the number of machines with failures, why not just say "10 machines have failures"? [13:18] <michaelni> J_Darnley, ask burek [13:21] <cone-899> ffmpeg.git 03Matthieu Bouron 07master:be0a67bd6508: lavfi/drawutils: fix blending computation in blend_line function [13:33] <cone-899> ffmpeg.git 03Anton Khirnov 07master:dda20a6e2c0c: avprobe: also output dar/par if only defined in stream [13:33] <cone-899> ffmpeg.git 03Anton Khirnov 07master:a5d8c9243a1d: Update release notes for the 9 release. [13:33] <cone-899> ffmpeg.git 03Anton Khirnov 07master:b14e89b3c5e6: Prepare for 9 Release. [13:33] <cone-899> ffmpeg.git 03Michael Niedermayer 07master:2163c8828d21: Merge commit 'b14e89b3c5e6d7f6401a2ff1e3d198fa902e988a' [13:34] <wm4> michaelni: when they release 9, will you do a release as well? it would possibly make things less painful for those who have to support both ffmpeg and libav [13:34] <durandal_1707> lol, why RELEASE_NOTES is stuck with 0.10 ? [13:35] <durandal_1707> wm4: yes, release 96 [13:35] <wm4> and making sure that the pkgconf versions match [13:41] <saste> otoh, what're release_notes useful for? [13:41] <saste> *what's [13:41] <durandal_1707> their relase notes files is their propaganda for all bad stuff they do [13:57] <cone-899> ffmpeg.git 03Reinhard Tartler 07master:3f89b49b0738: finalize changelog for version 9 [13:57] <cone-899> ffmpeg.git 03Xi Wang 07master:3b81bba3bc5a: mxfdec: fix NULL checking in mxf_get_sorted_table_segments() [13:57] <cone-899> ffmpeg.git 03Xi Wang 07master:f73f76fd202b: swscale: fix NULL checking in sws_alloc_context() [13:57] <cone-899> ffmpeg.git 03Michael Niedermayer 07master:bb4fb7715cf7: Merge remote-tracking branch 'qatar/master' [13:59] <saste> oh i messed the kerndeint argb test... [14:06] <cone-899> ffmpeg.git 03Stefano Sabatini 07master:860b5c0a631a: lavfi/kerndeint: fix mismatch between declared pixel format and test [14:24] <ubitux> is the lavfi filter_frame() merge completely done? [14:41] <cone-899> ffmpeg.git 03Carl Eugen Hoyos 07master:6a9af925654c: Allow remaining 32bit RGB packed pix_fmts in kerndeint filter. [14:51] <durandal_1707> can our filters do what sox effects do? like allpass [15:42] <cone-899> ffmpeg.git 03Nicolas George 07master:305180f5259e: lavu/base64: return meaningful error code. [17:08] <cone-899> ffmpeg.git 03Michael Niedermayer 07master:9a697cfe716e: lavu: test for broken binutils on ARM [17:26] <cone-899> ffmpeg.git 03James Almer 07master:b7d77f8e64d5: astenc: Enable the loop flag only when needed [17:26] <cone-899> ffmpeg.git 03James Almer 07master:6717d1a96ff2: MAINTAINERS: add myself as maintainer of lavf/astenc.c [18:09] <mateo`> is there a simple way to duplicate an AVPacket for later use ? [18:10] <wm4> av_dup_packet(), apparently [18:23] <mateo`> wm4: the documentation sounds scary :) [18:23] <nevcairiel> its one of those functions thats really rather crazy [18:24] <wm4> how could a function to copy something be crazy? [18:26] <mateo`> wm4: why av_dup_packet do not return a new allocated packet ? [18:26] <nevcairiel> I think it only copies the packet if the packet uses a static buffer [18:26] <nevcairiel> so it doesnt return a new packet [18:26] <nevcairiel> it just copies the internal data from a static to an allocated b uffer [18:27] <wm4> wat [18:28] <nevcairiel> i told you its a crazy function [18:28] <nevcairiel> some demuxers have static buffers into which they read data, and then put these static buffers into an AVPacket [18:28] <nevcairiel> av_dup_packets only purpose is to copy that data into a freshly allocated buffer [18:28] <wm4> hm, looks like it copies the packet contents [18:29] <wm4> so there isn't actually a static buffer [18:29] <wm4> but still, crazy enough [18:30] <wm4> how ffmpeg would design strdup(): void av_dup_string(char **string); [18:33] <wm4> ubitux: why did you use sscanf for new code? [19:07] <cone-899> ffmpeg.git 03Michael Niedermayer 07master:f3c9d66bafde: libspeexdec: fix terminator check [22:14] <ubitux> wm4: i guess i did, anything wrong with it? [23:13] <cone-370> ffmpeg.git 030sE 07master:7e5d4fa97d28: mmf.c: Do not write metadata into the SMAF Contents Info chunk. [23:36] <cone-370> ffmpeg.git 03Michael Niedermayer 07master:2468827c06ad: internal.h: remove start/end_frame from AVFilterPad [23:36] <cone-370> ffmpeg.git 03Michael Niedermayer 07master:39d187545596: libavfilter/video.h: remove unused things related to the start/slice/end API [23:41] <cone-370> ffmpeg.git 03Carl Eugen Hoyos 07master:1a34103f0f22: mmf.c: Use LIBAVFORMAT_IDENT when writing Yamaha SMAF version information. [23:42] <cone-370> ffmpeg.git 03Carl Eugen Hoyos 07master:8bf70159dcf4: 10l: Update fate seeking reference after last commit. [00:00] --- Sun Jan 6 2013

1 0

ffmpeg-devel.log.20130115
by burek 16 Jan '13

16 Jan '13

[00:07] <pettter> Compn: no, the transcoding is supposed to happen serverside [00:07] <bcoudurier> Skyler_, if any work is needed, definitely :) [00:07] <pettter> XBMC on the Pi is simply a client (in theory) [00:07] <Skyler_> I'll poke you if they need more stuff done [00:07] <bcoudurier> grazie mille [00:07] <Skyler_> in the meantime I said "ffmpeg already does that" XD [00:15] <cone-902> ffmpeg.git 03Carl Eugen Hoyos 07master:5ce023b790e5: Fix color filter example. [00:29] <gnafu> I know I have been loving OpenELEC on my Pi, though I have mostly been using it for online content that is already H.264 and viewing live TV from a MythTV backend on another machine (and viewing the raw MPEG-2 streams; I did pay for the decoder license). [00:29] <gnafu> Still the cheapest solution to accomplish what I wanted, so... [00:31] <Compn> pettter : then why dont you want to install xbmc on the server side? lol [00:38] <gnafu> Compn: Might want something headless. Can XBMC perform those functions without its GUI? (I don't really know, but it seems like it wouldn't.) [00:38] <Compn> is html gui , headless ? [00:39] <Compn> if you're watching films, you should have a head, somewhere :P [00:39] <gnafu> You have a point. I don't know if you can install, configure, run, and maintain XBMC without running its main interface, and he might want to avoid that. I know I didn't like having to run GUI setup applications to get MythTV up and running :-P. [00:40] <Compn> bah got me there [00:40] <Compn> installs are hard to do without a head :P [00:40] <Compn> course, what server comes autoconfigured? [00:40] <Compn> not ssh, http, ftp, irc, nfs, samba etc... [00:40] <gnafu> I had to wait until I got home from work to set it up, because running mythtv-setup with ssh -CX was painfully slow and not as inconspicuous as just a terminal with vi. [00:41] <gnafu> I like installing packages with sane default configurations, and then editing config files to suit my needs :-D. [00:41] <Compn> an html installer would be awesome [00:41] <Compn> for many programs [00:41] <gnafu> That's what I did with things like httpd, munin, and almost everything else running at home. [00:41] <gnafu> That is very true. [00:41] <Compn> maybe that would be useful, create some kind of html wrapper to x11/gnome [00:42] <Compn> so they render with html instead of window effects [00:42] <Compn> or maybe just a remote install option [00:42] <gnafu> Ooh, and name it something flashy, like /Broadway/. [00:43] <gnafu> ;D [00:43] <Compn> broadway 2.0 [00:43] <gnafu> I agree with you, though. And hopefully things like GTK+'s Broadway backend will enable it. [00:46] <saste> [video4linux2,v4l2 @ 0x7fb5200042d0] Cannot find a proper format for codec_id 0, pix_fmt -1. [00:47] <saste> ^^ why it is not auto-selecting the format (like before)? [00:47] <wm4> did elenril break it? or the merge? [00:48] <saste> wm4, dunno, hard to say without bisecting [00:48] <saste> it was since some time i wasn't using it, may be a local problem [00:48] <saste> but the command ffplay -f video4linux2 /dev/video0 usually was working fine [00:48] <cehoyos> saste: ffmpeg -f v4l2 -i /dev/video1 works fine here [00:49] <cehoyos> Same with ffplay [00:49] <saste> cehoyos, uh carl? [00:49] <cehoyos> Yes [00:50] <saste> you're not usually on irc [00:50] <saste> maybe it's the first time i see you here [00:50] <cehoyos> So should I go? [00:50] <cbsrobot> wut just hapend [00:50] <cbsrobot> am I dreaming ? [00:50] <saste> cehoyos, you're welcome ;-) [00:50] <saste> and we lost ubitux since a few days [00:51] <cbsrobot> no ubitux was here a few hours ago [00:51] <saste> good, so he's still alive :) [00:51] <cbsrobot> hi carl [00:52] <cehoyos> Hi [01:06] <michaelni> Daemon404, Is the latest "[PATCH] Fix compilation with libutvideo version 12.0.0" patch ok ? [01:10] <Daemon404> the latest one seemed fine [01:15] <cone-902> ffmpeg.git 03Stephen Hutchinson 07master:df4203ac6f00: Fix compilation with libutvideo version 12.0.0 [02:52] <Compn> oooooooo carl was online [02:52] <Compn> missed him! [05:37] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:8ac8f04993e5: mpegvideo: Fix long standing race condition with frame threads [07:53] <highgod> Hi, I want to ask a question. How can I know the VC1 video format is 420 or 444 in VC1Context [07:56] <funman> highgod: hello [07:57] <highgod> hi funman [07:59] <nevcairiel> VC1 is always 420 [07:59] <funman> highgod: you're working on dxva2 standalone decoder, right? (using hwaccel internally) [07:59] <j-b> funman: t'es debout ma caille? [08:00] <funman> on dirait que oui [08:00] <highgod> funman:yes,I'm fixing the patch now [08:00] <highgod> nevcairiel:Thanks [08:00] <funman> highgod: btw about vlc attribution you can just copy the Copyright lines from vlc files [08:04] <highgod> Thanks,funman,I have copy the copyright lins form vlc.Should I copy all the lines on the top of the files? [08:06] <funman> highgod: no I think that is enough [08:08] <highgod> OK,thanks. [09:01] <highgod> Hi, I want to ask a question again, can I use the chroma_format in MpegEncContext to know whether the mpeg2 video is 420 or 422 or 444? Thanks [13:46] <Darkarnium> Hey there, a quick question regarding acceptable standards for passing arbitrary information to modules. I've seen a few mail list posts regarding HLS cookies, and I'm working on a project which requires the abillity to provide a cookie in fetching the AES keys - where applicable - or when grabbing the TS URLs. However, I've seen a few applications where the cookie / authentication data is actually provided before the M3U8 request is f [13:47] <Darkarnium> Although the application is broken, the abillity to provide customer header data - such as User-Agents, etc - might be a welcome addition to HLS? Or is this something that shouldn't be implimented for a given reason? If not, I'd be happy to give it a shot :) [13:48] <saste> Darkarnium, hi think Micah Galizia is working on something similar [13:48] <saste> right now I'm not sure how HTTP options are propagated to the HLS muxer [13:49] <saste> but sure patches are welcome [13:50] <Darkarnium> Ah, very cool. I did some testing with some static cookies - temporary, very temporary, to confirm behaviour. I can confirm that the AVDictionary passed to ffurl_open in hls.c supports the addition of a headers option. This propagates through into the HTTP call - for both HTTP and HTTPS protocols - confirmed with a couple of packet captures. [13:50] <Darkarnium> So, that might be a useful hook. I'll have a bit more of a read, and see how I go. Once I have something workable, that's to spec, I'll ask for a review. What's the best method for the latter, a pull request, or otherwise? [13:51] <Darkarnium> If there's a document regarding code standards or development standards process, I'd be more than happy to shut up and read it :P [13:56] <saste> Darkarnium, http://ffmpeg.org/developer.html [13:56] <saste> patches created with git-format are usually fine [13:56] <Darkarnium> Fantastic. Thanks for your time saste, much appreciated :) [13:58] <pross-au> Darkarnium: just on arbitary information. do you mean supplied by the user? like a decrypt key? [14:01] <Darkarnium> pross-au: at this stage, just cookie data. I say arbitrary as the HLS streaming providers I've had a look at so far all differ in their A/V names for cookie data - as is to be expected. decryption keys are still fetched as per the HLS draft RFC, I wouldn't be tampering with that, as I'd end up breaking the proposed standard [14:01] <saste> Darkarnium, anyway, I'm going to push the cookie stuff tonight [14:02] <Darkarnium> Oh, cool, is it available to the world currently? Or private channels only before it's incorporated [14:03] <pross-au> ok. for the pcoip demuxer (working in progress) i use an av_url type option to pass cookie-like info to the decoder. [14:04] <Darkarnium> Interesting, is PCoIP it's own protocol, or wrapped around another? As the reason I was thinking of exposing '-headers' was that it'd allow for easy addition of extra HTTP headers in the case that they're required - in line with HLS and HTTP standards of course. [14:05] <pross-au> it is 'two' protocols, one to initialise the session, the other encapsulated in ssl. [14:07] <pross-au> correction: the session initialisation is encapsulated in ssl, the second protocol (carry a/v/kvm) is salsa20/aes128/aes256 encrypted udp (tcp optional)/ [14:08] <Darkarnium> Ah cool [14:08] <Darkarnium> I assume the authentication data for the latter is provided as part of the session setup, via SSL? [14:08] <Darkarnium> Thus the need to pass between modules? [14:09] <pross-au> no [14:09] <Darkarnium> Hah, you know what they say about assumptions :p [14:09] <pross-au> there is some additional salt-like data, that needs to be supplied seperately [14:09] <pross-au> hence the av_url argument [14:09] <Darkarnium> Ahh [14:10] <pross-au> IIRC, the url was looking like pcoip://destination:port?ssl_options=foo&salt=bar [14:12] <pross-au> the salt data is communicated through another (third) channel, called the broker. [14:12] <pross-au> its more of a vdi protocol, than a multimedia streaming protocol. [14:12] <Darkarnium> Fair enough. [14:14] <Darkarnium> This HLS implementation I'm working to write a client for is rather awful. Client authenticates via web service, is provided with a loginToken, and a set of allowed 'channels'. From there additional HTTP calls are performed, which yield XML back from the provider, containing a custom XML schema with cookie data and an M3U8 URL. I'm not sure why the cookie data wasn't returned using set cookie, but what can you do. [14:15] <Darkarnium> So the juggling act is to ensure it can be implimented, as per HLS standard, but flexible enough to be standards compliant and able to be distributed. [14:15] <Darkarnium> So the abillity to pass the relevant data via CLI may work out to be the best method of doing so [14:17] <Darkarnium> I should rephrase, the client itself is Python, but ffmpeg is used to handle the hls streams - in line with the HLS standard. Not cowboying things up :P [14:19] <Darkarnium> Anyway, it's late here, so about time for me to get some sleep [14:19] <Darkarnium> Cheers for all the assistance and suggestions :) [14:19] <Compn> Darkarnium : to answer your question, its a patch on the mailing list [14:19] <Compn> saste is either going to ping it or commit it :) [14:33] <cone-169> ffmpeg.git 03Ronald S. Bultje 07master:f6badba1859f: h264: don't clobber mmco opcode tables for non-first slice headers. [14:40] <cone-169> ffmpeg.git 03Martin Storsjö 07master:62761934b024: rtpdec: Remove a woefully misplaced comment [14:40] <cone-169> ffmpeg.git 03Martin Storsjö 07master:d0fe217e3990: rtpdec: Simplify insertion into the linked list queue [14:40] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:eaf1f0116935: Merge commit 'd0fe217e3990b003b3b3f2c2daaadfb2af590def' [14:46] <cone-169> ffmpeg.git 03Martin Storsjö 07master:30b50f79aea3: rtpdec: Handle more received packets than expected when sending RR [14:46] <cone-169> ffmpeg.git 03Diego Biurrun 07master:ba0c72a9ae1e: build: Remove stray Makefile entry for non-existent VCR1 encoder [14:46] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:8686b6c68bac: Merge commit 'ba0c72a9ae1e2954e5dcf920f7b4e9a8f8a22f3e' [15:13] <cone-169> ffmpeg.git 03Martin Storsjö 07master:d596f2b32264: rtpdec: Make variables that should wrap unsigned [15:13] <cone-169> ffmpeg.git 03Tom Finegan 07master:66aabd76a9c0: mkv: support vp9 tag [15:13] <cone-169> ffmpeg.git 03Luca Barbato 07master:23a610b9d66a: nut: support vp9 tag [15:13] <cone-169> ffmpeg.git 03Luca Barbato 07master:dab1f543fcac: libvpx: support vp9 [15:13] <cone-169> ffmpeg.git 03Luca Barbato 07master:3f111804eb5c: libvpx: make vp8 and vp9 selectable [15:13] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:353dbaa29714: Merge commit '3f111804eb5c603a344706b84b7164cbf7b4e0df' [15:21] <cone-169> ffmpeg.git 03Ronald S. Bultje 07master:bad446e25140: h264: don't clobber mmco opcode tables for non-first slice headers. [15:21] <cone-169> ffmpeg.git 03Giorgio Vazzana 07master:39403c6c1baf: oggparsetheora: fix comment header parsing [15:21] <cone-169> ffmpeg.git 03Maximilian Seesslen 07master:055b37308080: libtheoraenc: fix granularity of video quality [15:21] <cone-169> ffmpeg.git 03Sean McGovern 07master:5e753ed502d3: suncc: Replace more GCC flags by their equivalents in suncc_flags() [15:21] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:24d06cb2089f: Merge commit '5e753ed502d3597077d8675ca1438e1bcade1459' [15:30] <cone-169> ffmpeg.git 03Anton Khirnov 07master:ea382767ad21: h264: fix ff_generate_sliding_window_mmcos() prototype. [15:30] <cone-169> ffmpeg.git 03Rémi Denis-Courmont 07master:171f1446f054: vdpau: Remove av_unused attribute from function declaration [15:30] <cone-169> ffmpeg.git 03Diego Biurrun 07master:51969a652c2e: x86: ABS2: port to cpuflags [15:30] <cone-169> ffmpeg.git 03Diego Biurrun 07master:99853cb8d423: configure: Make warnings from -Wreturn-type fatal errors [15:30] <cone-169> ffmpeg.git 03Diego Biurrun 07master:d8c772de53d2: nutdec: Always return a value from nut_read_timestamp() [15:30] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:cfc40a6aff8d: Merge commit 'd8c772de53d29afb1bada88afa859fce8489c668' [15:55] <cone-169> ffmpeg.git 03Luca Barbato 07master:bff3607547fd: lavc: set the default rc_initial_buffer_occupancy [15:55] <cone-169> ffmpeg.git 03Luca Barbato 07master:47812070a267: libx264: use the library specific default rc_initial_buffer_occupancy [15:55] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:9aeffb3c2a0d: Merge commit 'bff3607547fdbb6e32b3830a351e6a33280c1e0d' [15:55] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:e074fe2962df: Merge commit '47812070a267cbdf74164e154d03d99bf8ced100' [16:03] <cone-169> ffmpeg.git 03Martin Storsjö 07master:8ee288d2586f: lavu: Add an API for calculating HMAC (RFC 2104) [16:04] <cone-169> ffmpeg.git 03Martin Storsjö 07master:ab2ad8bd5688: lavf: Add functions for SRTP decryption/encryption [16:04] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:e7e0186eeb0a: Merge commit 'ab2ad8bd56882c0ea160b154e8b836eb71abc49d' [16:12] <cone-169> ffmpeg.git 03Martin Storsjö 07master:424da308302b: rtsp: Support decryption of SRTP signalled via RFC 4568 (SDES) [16:12] <cone-169> ffmpeg.git 03Martin Storsjö 07master:2f3bada63e57: lavf: Add a protocol for SRTP encryption/decryption [16:12] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:b52925d2cde9: Merge commit '2f3bada63e57345329c4f9b48e9b81b5cfc03d05' [16:16] <cone-169> ffmpeg.git 03Martin Storsjö 07master:611bf39bde60: sdp: Include SRTP crypto params if using the srtp protocol [16:16] <cone-169> ffmpeg.git 03Diego Biurrun 07master:094a7405e5d8: x86: ABSB: port to cpuflags [16:16] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:77041e2474a1: Merge commit '094a7405e5d8463d7d167d893e04934ec1a84ecd' [16:20] <cone-169> ffmpeg.git 03Diego Biurrun 07master:320e1d0df3df: x86: ABSB2: port to cpuflags [16:20] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:b7ede94bbd29: Merge remote-tracking branch 'qatar/master' [16:28] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:918b41163671: rtpdec: support CSRC [20:18] <michaelni> bcoudurier, following patch should fix the chunked duration issue: "[FFmpeg-devel] [PATCH 2/2] mux: fix chunked interleaver" [20:18] <llogan> siretart: do you expect the ffmpeg package to be dropped by raring? i couldn't tell in the "Intending to..." thread [20:27] <michaelni> bcoudurier, review and comment if it works for you are welcome! [20:28] <siretart> llogan: well, let's say I still have hope. but I don't have the time to fix all packages in the archive, which needs to be done by feature freeze for that. my hope is that other packagers can jump in [20:28] <bcoudurier> let me see [20:28] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:778069c83255: vorbisdec: support freeing partially allocated contexts. [20:28] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:c5cf58d4b9b0: oggdec: resync from the last page. [20:28] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:e9ffee23f372: vorbisdec: handle midstream parameter changes [20:28] <cone-169> ffmpeg.git 03Michael Niedermayer 07master:c994bb2fb772: oggdec: Leave treatment of serial changes to the decoder. [20:29] <llogan> siretart: i see. thanks. [20:29] <siretart> as you can see, the fallout is much larger than expected. [20:30] <bcoudurier> ah you fixed it in a way I was investigating :) [20:31] <cehoyos> llogan : You see, this is not about providing users with a working A/V converter that is not exploitable out-of-the-box but about finding more packagers who support hostage-taking... [20:32] <llogan> cehoyos: hell hath frozen over when cehoyos appears on IRC! [20:32] <durandal11707> @_@ [20:34] <llogan> siretart: obviously i'd like to see an ffmpeg package with FFmpeg as upstream..even if it is just a binary and libraries with no depends. assuming we have the manpower to provide this, do you see this as a possibility? [20:35] <llogan> and obviously i don't know the details and work involved in package sponsorship, mantaintaing, etc [20:35] <llogan> yet [20:52] <llogan> ...i was just about to do that [21:08] <Daemon404> should really add threading to utvideoenc.c and just drop libutvideo*.cpp [21:14] <cone-169> ffmpeg.git 03Michael Niedermayer 07release/1.1:66a311210008: oggdec: resync from the last page. [21:14] <cone-169> ffmpeg.git 03Michael Niedermayer 07release/1.1:dc3349024a06: vorbisdec: support freeing partially allocated contexts. [21:14] <cone-169> ffmpeg.git 03Michael Niedermayer 07release/1.1:9636266cbdab: vorbisdec: handle midstream parameter changes [21:14] <cone-169> ffmpeg.git 03Michael Niedermayer 07release/1.1:1c373456f638: oggdec: Leave treatment of serial changes to the decoder. [21:18] <durandal11707> why ffmpeg -h full does not show default option? [21:45] <wm4> are the symbols in libavcodec/vda.h public? is it a public header? are the ff_vda_ functions in it public? [21:46] <durandal11707> generally * is private except av* [21:47] <durandal11707> so it is big mess that vda stuff [21:47] <wm4> VLC uses ff_vda_ symbols [21:48] <durandal11707> it may use own hacked version or statically linked [21:49] <durandal11707> i did not do that mess, so this is guessing on my side... [21:56] <cehoyos> Afaict, vda.h is an installed header, so although the functions are not av* but ff* they are public [21:57] <durandal11707> but ff_ in that header is not in libavcodec.v [21:58] <durandal11707> it is deprecated function btw [21:59] <cone-169> ffmpeg.git 03Stefano Sabatini 07master:1b325ce91ab7: lavd/v4l2: extend error/debug feedback in case of invalid codec/pix_fmt [21:59] <cone-169> ffmpeg.git 03Stefano Sabatini 07master:fce165027fd8: lavd/v4l2: fix misc messages [21:59] <cone-169> ffmpeg.git 03Stefano Sabatini 07master:aa359d3808e3: lavd/v4l2: return meaningful error code from device_init() [21:59] <cone-169> ffmpeg.git 03Stefano Sabatini 07master:d012059e7ba6: lavd/v4l2: apply grammar/consistency fixes to options help fields [22:29] <cone-169> ffmpeg.git 03Micah Galizia 07master:0b80a12184ef: lavf/http: add HTTP protocol cookie support [22:29] <cone-169> ffmpeg.git 03Micah Galizia 07master:10315b1cb817: doc/protocols: document HTTP protocol cookie support [22:48] <Darkarnium> hehe, thank god for time zones. [22:50] <Darkarnium> Just got up, ready for work and saw that Micah has pushed the cookie support, which supports passing of cookies via -cookies as well as via set cookie. [22:50] <Darkarnium> That makes my life easy! :) [22:51] <cone-169> ffmpeg.git 03Stefano Sabatini 07master:c92b6f347dca: tools: add plotframes script [22:58] <Compn> Darkarnium : you said something about adding features to hls ? only if its in the spec [23:05] <Darkarnium> Compn: it was, then again, the spec for HLS is a bit light currently. The feature was to do with authentication for grabbing AES keys. The spec specifies that HTTP session cookies and SSL should be used; the only questionable element is how the cookies are set. The expected way would be via a Set-Cookie header, however, an override may be appropriate to cover the use cases where the provider is specifying the session cookie in another [23:06] <Darkarnium> I should probably say proposed spec / RFC [23:12] <kierank> <troll>as if HLS and friends follow specs</troll> [23:12] <Daemon404> lolapple [23:12] <Daemon404> lolakamai [23:13] <Darkarnium> kierank: I know -__ [23:13] <Darkarnium> The RFC is so damned open to interpretation. [23:13] <kierank> that's a significant reason i don't work on that web stuff [23:13] <kierank> people will just assume iDevices are correct [23:14] <Darkarnium> HURR, let's allow AES for key delivery! That way X provider can decide to use the most OBSECURE HTTP header they can find, munge it into only *JUST* being compliant and then deploying it. [23:14] <Darkarnium> Err, HTTP that should be [23:15] <JEEB> gentlemen, welcome... to the internets [23:15] <Daemon404> kierank, we've successfully reported bugs to apple before [23:15] <Daemon404> it /is/ possible. [23:15] <Darkarnium> Just painful? [23:16] <Daemon404> i wasnt involved in it, so i cant say [23:16] <Darkarnium> Ahh [23:18] <Darkarnium> The only platform I've found HLS to work on, well, out of the box is an iDevice. Probably because apple submitted the RFC, and now it's being picked up and used. However, Android doesn't like HLS with certain attributes - such as cookies as part of key retreival - so you're pretty much backed into a corner until there is significant demand enough for it to be implimented, or you write a patch and submit it yourself. [23:18] <Darkarnium> Anyway, I should probably stop ranting... [23:18] <Daemon404> well [23:19] <Daemon404> also, services all are built around iDevice [23:19] <Daemon404> bent to apples possibly broken impl [23:20] <Darkarnium> Exactly. The reason I wrote a tentative patch for cookies to HLS - which I'm canning and now using Micah's as it's much better written - is for a cable provider in Australia who have an iDevice application for live streaming. Until their broken HLS implimentation is supported on Android, I doubt we'll see it released across platforms. Hopefully, the implimentation of it inside ffmpeg, and more adoption of this method of 'securing' key [23:21] <Daemon404> (you got cut off) [23:21] <Darkarnium> Oh :( [23:24] <Darkarnium> Anyway, it's off to work I go [23:38] <cone-169> ffmpeg.git 03Angelo Haller 07master:e7a39e163ddb: examples/demuxing: free AVPacket after usage [00:00] --- Wed Jan 16 2013

1 0

ffmpeg.log.20130115
by burek 16 Jan '13

16 Jan '13

[00:09] <Sashmo_> some told me that there is some C library out there that can do very good PAL-NTSC conversion, but I dont have a name, anyone know anything about it? [01:05] <Fedcomp> hi guys [01:05] <Fedcomp> can i ask? [01:05] <klaxa> always [01:05] <Fedcomp> i have PC computer with Intel Core i3, and linux machine with core 2 duo [01:06] <Fedcomp> i want to stream. But xsplit gives me ugly quality on my pc, so i want to use linux machine as encoding [01:06] <Fedcomp> does pc:ffmpeg to linux:crtmpserver+ffmpeg to stream to twitch any better than ffsplit? [01:07] <Fedcomp> worth to try? [01:07] <klaxa> worth to try? probably, it's an interesting setup and not trivial [01:07] <Fedcomp> it's not that hard to setup, i have some knowledge and almost done with crtmpserver [01:08] <klaxa> linux machine with core 2 duo might be a bit weak for live-encoding, depending on the content [01:08] <Fedcomp> can ffmpeg grab opengl/directx video directly? [01:08] <klaxa> you could let ffmpeg stream over udp throughout the network and let it grab that stream on the linux machine [01:08] <Fedcomp> well, i agree, but it's better than capture from screen, encode, send [01:08] <Fedcomp> hmm interesting advice. [01:09] <klaxa> since using rtmp involves using udp anyways, it might reduce overhead [01:09] <Fedcomp> got that. can ffmpeg grab opengl/directx video directly? [01:09] <Fedcomp> i haven't found answer for this anywhere. [01:09] <klaxa> http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20grab%20the%20desktop%20%28scr… [01:09] <klaxa> i doubt it can [01:09] <Fedcomp> it's not desktop, it's a game :3 [01:09] <saste> Fedcomp, it can't, support X11 grabbing though [01:10] <Fedcomp> okay ty. [01:10] <Fedcomp> using windows for now due to bad games support on linux. [01:10] <saste> and watches pelcome [01:10] <Fedcomp> is it planned? or you don't consider it? [01:10] <Fedcomp> i mean opengl/directx grabbing [01:10] <saste> Fedcomp, create a ticket and we'll see [01:11] <saste> no opengl expertise here [01:11] <Fedcomp> okay [01:11] <Fedcomp> sorry no time to make ticket, but thanks for answers. [09:35] <dott_infernus> Hi, I was here, yesterday...I need some help [10:43] <EricAhn> are there any "VideoCodec vcopy" in ffserver.conf? I reviewed the ffserver.c but, I can't found. But, It have the feature in ffmpeg. [10:44] <EricAhn> are there any reason that does not support copying video/audio data into ffserver (ffm)? [10:45] <EricAhn> It doesn't think that far. [10:45] <EricAhn> ? [11:24] <dott_infernus> Hi. guys...I'm trying to use ffmpeg libraries on Qt... It gives me undefined reference when I try to call some function inside libraries...what I have to write into .pro file? [12:01] <viric> umh [12:01] <viric> I've used "-t 11400" encoding from video4linux2 [12:01] <viric> And after 3h40m of video, it was still encoding [12:35] <dott_infernus> Hi, is there anyone? [13:02] <dott_infernus> Guys, could someone help me with this-> http://pastebin.com/X9u52xDs [13:02] <dott_infernus> This problem with ffmpeg is trying to kill my life [13:02] <dott_infernus> afk [13:04] <cbsrobot> dott_infernus: shouldnt the include path just be /usr/local/include ? [13:11] <saste> dott_infernus, please use pkg-config [13:11] <saste> check doc/example/Makefile to see how to integrate ffmpeg libs into your projects [13:28] <Podium> Hello all! Does anyone know something about color-shift in the contrast/saturation ?? This happends here when I try to transcode a ProRes (yuv422p10le) file to a libx264 (yuv420p) file... [13:31] <Mavrik> define color shift [13:31] <Mavrik> but yes, you'll lose some color information when doing that [13:32] <Podium> Hi Mavrik! White becomes gray.... Grass greens becomes gray/green... [13:32] <Podium> It's just a little... But it's really a little to much [13:33] <Mavrik> hmmm [13:33] <Mavrik> that sounds excessive (usually only reds become duller) [13:35] <Podium> So it's logical color changes a bit... (I can image) but is there maybe a "default" color correction preset? [13:36] <Mavrik> Podium, there's the colormatrix parameter for libx264 which specifies the color conversion matrix: http://mewiki.project357.com/wiki/X264_Settings#colormatrix [13:36] <Mavrik> I've found that it sometimes fixes the problems [13:36] <Mavrik> try setting different color matrices with -x264opts [13:48] <Podium> Thx... I don't know yet if its a solution. But I see to colormatrix's (smpte170m & smpte240m) who look familair! [13:48] <Podium> And about my audio problem yesterday... [13:49] <Podium> I now first export the audio with ffmpeg, than delete the 14 channels with sox and than again import it with ffmpeg and the other video filters.... [13:55] <pacior> hello i am compiling ffmpeg from source on linux mint using toolchain and during script i have problem with configure script. it writes: arm-linux-androideabi-gcc is unable to create an executable file. [13:55] <pacior> C compiler test failed. [13:55] <pacior> i've got this messages in log: Assembler messages: [13:55] <pacior> Fatal error: invalid -march= option: `armv7-a' [13:55] <pacior> /tmp/ffconf.cSR9QYsG.c:1:0: warning: target CPU does not support THUMB instructions [enabled by default] [13:55] <pacior> C compiler test failed. [13:55] <pacior> please help [13:58] <viric> add -marm [13:58] <viric> (if you don't want to generate thumb) [13:58] <viric> and, are you sure you want armv7-a? [14:03] <pacior> no i want really my app compatible with most of processors [14:03] <pacior> so armv6 is ok? [14:03] <pacior> and big thanks:) [14:13] <pacior> thank you for previous answer, but i've got now missing argument to -mcpu error [14:19] <viric> so it's clear, no? [14:23] <pacior> not yet, can i find this in toolchain docs? [14:24] <pacior> i have one question - if most android devices now are arm-based, mean nexus 7, galaxy ace etc [14:25] <pacior> my problem starts when capture frames library starts with api level 10 since i need api level 8 for my app [14:37] <viric> no idea about mobile phones. [14:37] <viric> or non-GNU/linux [15:01] <pacior> ok, thx [15:09] <DonGnom> is there any way to convert a side-by-side 3D video to non 3D (i.e. dropping one side)? [15:14] <pacior> viric: it's idea to compile it on android since it's kernel is linux-based [15:15] <viric> that's why I said GNU [15:48] <Abhijit> hi [15:48] <Abhijit> anyone familiar with libavg coding? [15:49] <Abhijit> i need to program libavt to use vaapi. and libavg uses ffmpeg to do the actual encoding/decoding. so how can i make ffmpeg enable to use vaapi? [15:49] <Abhijit> help please [21:14] <dott_infernus> Hello there... [21:15] <dott_infernus> I need some help [21:20] <stqn> who doesnt [21:20] <stqn> ;) [21:21] <dott_infernus> stqn [21:22] <dott_infernus> may I ask you? [21:22] <durandal11707> just ask [21:25] <dott_infernus> I've installed ffmpeg libraries on ubuntu, following the guide [21:26] <dott_infernus> almost here, everything is ok. I want to use this library on my own project with Qt Creator..but I didn'y find any wiki to link ffmpeg libraries into qt creator. [21:27] <dott_infernus> I've tried to include some libraries into project file (.pro) but when I try to compile, the compiler gives me undefined reference.. [21:28] <stqn> I know you can manually edit the .pro file, but theres probably a gui for that& [21:29] <dott_infernus> let me be more clear, I've created a new project, into to main file I call #include <libavutil/...> and go on...if I don't call any function, it compiles correctly...when I call just one function inside the libraries it gives me undefined reference [21:30] <dott_infernus> stqn, it is what I've done [21:30] <stqn> for example you can add : LIBS += -lssl in the .pro to link with libssl [21:30] <stqn> (at least for the project Im working on&) [21:31] <dott_infernus> ok...may I paste here some rows about my .pro? [21:31] <dott_infernus> LIBS += /usr/local/lib/libavcodec.a \ /usr/local/lib/libavdevice.a \ /usr/local/lib/libavfilter.a \ [21:31] <dott_infernus> is that wrong? [21:33] <stqn> cant say for sure, but seems ok to me [21:34] <dott_infernus> do I need to add -l when I link libs? [21:34] <stqn> not when you give the complete path, I think [21:35] <dott_infernus> ah... [21:36] <dott_infernus> another question.. [21:38] <dott_infernus> I want to compile the examples given to me after the installation of ffmpeg libraries...the examples inside doc/example...but when I launch gcc from terminal...it gives me the same error (undefined reference) ... someone told me that I've missed some libraries....how can I determine which libraries I have to call during that call? Using the makefile will avoid this problem? [22:25] <tyebud> Hello all. Does ffmpeg support qt chapter tracks? [22:32] <durandal11707> tyebud: you have link/sample/file ? [22:32] <durandal11707> chpl is supported [22:33] <tyebud> durandal11707: yeah let me upload my example real quick. [22:33] <tyebud> durandal11707: Here you go - https://dl.dropbox.com/u/132898/keynote_animation_test.mov [22:35] <durandal11707> i get error: Referenced QT chapter track not found [22:35] <tyebud> durandal11707: Yeah, same here. But, it loads and plays the chapters or whatever they're called fine in quicktime [22:36] <durandal11707> this is complete file? [22:36] <tyebud> Yep [22:37] <durandal11707> than it is bug, open bug report with link to this file [22:37] <durandal11707> you could also put what qt list for chapters [22:38] <durandal11707> btw, what created such file? [22:38] <tyebud> This was an export from keynote [22:38] <cbsrobot> and there is a sprite track aswell [22:38] <cbsrobot> what is that for ? [22:39] <tyebud> sprite track? [22:39] <tyebud> Perhaps suptitles? [22:41] <durandal11707> perhaps it is chapter for subtitle stream? [22:41] <teratorn> are there any well-known video resolutions where the width or height is *not* a multiple of 2? [22:43] <Mavrik> teratorn, multiple of 2 or divisible by 2? [22:43] <Mavrik> hm [22:43] <Mavrik> that's the same thing [22:43] <teratorn> yeah :) [22:44] <Mavrik> teratorn, pretty much any that don't use chroma subsampling [22:44] <Mavrik> so, 4:4:4 formats [22:44] <Mavrik> can be arbitrary [22:44] <teratorn> yeah, I was just looking for a well-known resolution to hit my co-workers over the head with [22:45] <Mavrik> well, since most decoders don't support anything else than 4:2:0 [22:45] <Mavrik> it's gonna be really hard to find :) [22:45] <JEEB> more like the thing is that most commonly used end-user formats use 4:2:0 :P [22:45] <teratorn> well, so what if there is chroma subsampling. you still have full luma resolution. [22:45] <teratorn> it just means that certain pixels have the same color values as certain other pixels [22:46] <JEEB> you generally can't encode subsampled stuff with a non-mod(subsampling) resolution [22:46] <teratorn> ok that is fair. the actual complain is about doing picture-in-picture compositing (overlays). [22:46] <Mavrik> teratorn, yeah, but encoders pretty much consistently demand that your video resolution is divisible by subsampled number [22:46] <teratorn> they think that it has to be done on 2x2 pixel boundaries [22:47] <teratorn> i say no [22:47] <teratorn> for YUV420 [22:47] <Mavrik> no reason for it really [22:47] <Mavrik> :) [22:47] <durandal11707> people still use that? [22:48] <teratorn> durandal11707: what? :) [23:26] <damaru> hi - I am on debian wheezy and want to revert to ffmpeg and not avcon [23:27] <damaru> got the .deb from the debian multimedia but then can't dpkg -i [23:27] <damaru> hum... [23:38] <damaru> ha ok apt-get remove acvon and the libav then download the ffmpeg deb, get the multimedia apt source, then apt-get install -f [23:41] <AL13N> i have an odd thing: i use -s 640x480 -r 12 and i convert to mp4/ogg/webm; however, the image preview for the webm is stretched into 640x480, while the others have 640x422, subsequently taking an image with -vf frames=1 also stretches the image [23:41] <AL13N> why would the image preview of mp4 and ogg not be stretched and webm would be stretched? [23:42] <AL13N> it would only make sense if the -vf frames=1 loses the aspect ratio and the webm having included preview and mp4/ogg not [23:44] <llogan> damaru: you got it working? [23:45] <llogan> alternatively you could compile ffmpeg or use a static build [23:45] <damaru> yep it finally worked, it was just annoying how ffmpeg was pointing to avconv [23:45] <llogan> yes, i agree. [23:46] <damaru> now I am trying to convert a .swf to mov or webm but get a incomplete frame [23:48] <damaru> http://pastebin.com/pAAgSLRi [23:49] <damaru> ousp it cuts the end [23:50] <damaru> ho no - it's all right [23:50] <damaru> with the error at the end of the pastbin [23:50] <llogan> does the output look ok? [23:50] <damaru> the output works for the few first second [23:50] <llogan> the inpt keeps changing frame size and pixel format [23:50] <damaru> then blanks up [23:51] <llogan> with what player? [23:51] <damaru> the flash video was made by a beginner so their might be error [23:51] <damaru> with vlc [23:51] <damaru> can I skip 'incomplete frame' ? [23:52] <llogan> i'd blame the input, but i don't really know what to suggest [23:52] <llogan> can vlc play the input swf file? [23:52] <damaru> yeah I might screen record the video - vlc cannot but gnash was able to play it [23:53] <damaru> any good recommendation for screen video recording on debian? [23:56] <damaru> ha good old kdenlive [23:57] <llogan> ffmpeg -f x11grab -r 30 -s 1024x768 -i :0.0 -vcodec libx264 -crf 18 -preset ultrafast -pix_fmt yuv420p output.mp4 [23:57] <stqn> damaru: the Arch wiki says ffmpeg& https://wiki.archlinux.org/index.php/FFmpeg#Screen_cast_to_.webm [23:57] <damaru> ha! [23:57] <damaru> that's sweet [23:58] <llogan> that wiki page needs a good cleanup [23:59] <stqn> I thought someone would say that ;) [23:59] <stqn> Im not an expert myself& [00:00] --- Wed Jan 16 2013

1 0

ffmpeg-devel.log.20130114
by burek 15 Jan '13

15 Jan '13

[00:25] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:2d8d63a9a07c: ffplay: fix sws_flags [00:25] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:c13e4e288c84: h264: fix () placement [03:05] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:55d052866964: swr/dither: fix division by 0 [03:05] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:aaa7d2fafcc3: h264: don t leave stale pointers in delayed_pic in flush_changes. [05:28] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:48a7981e6f93: doc/examples: fix assignments in if() [05:28] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:98fed59427ce: aacenc: Fix assignments in if() [05:28] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:68a25c64cda1: libfaac: Fix assignments in if() [05:28] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:9302ad1ac89d: libfdk-aacenc: Fix assignments in if() [05:28] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:871b6ec01d27: libmp3lame: Fix assignments in if() [05:28] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:d6180aa29741: libopencore-amr: Fix assignments in if() [05:28] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:3b8d66d5317d: libspeexenc: Fix assignments in if() [05:28] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:0ccb31dcad5a: libvo-aacenc: Fix assignments in if() [05:28] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:bdd71abe5f34: libvorbisenc: Fix assignments in if() [05:28] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:795d2dc23b16: nellymoserenc: Fix assignments in if() [05:28] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:e2704381e5f1: ra144enc: Fix assignments in if() [05:28] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:37be1d802f2e: svq1enc: Fix assignments in if() [05:28] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:659546b42d65: tiff: Fix assignments in if() [05:28] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:6f88d2d786b8: vsrc_testsrc: Fix assignments in if() [05:28] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:1ac5a8d7e334: lavf/mux: Fix assignments in if() [05:28] <cone-117> ffmpeg.git 03Michael Niedermayer 07master:a601eb9543ec: rtmpproto: Fix assignments in if() [11:09] <ludde> anyone knows any .ts analysis tools to see if they're valid with respect to timing / muxing? [12:01] <kierank> ludde: nothing that doesn't cost less than $10k [12:10] <ludde> ok [12:11] <cbsrobot> kierank: maybe dnxhd-tr uses it's own quant table too [12:11] <cbsrobot> there is some discussion on ffmbc, but bcoudurier is just saying it is not implemented. [12:51] <durandal_1707> there is bunch of spam for av_restrict on same fate machines [13:34] <cone-774> ffmpeg.git 03Rémi Denis-Courmont 07master:44e065d56c87: vdpau: Add context and common helpers for hwaccel support [13:34] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:8ee7b3881bfe: Merge commit '44e065d56c87d6a9d0effccec5f31517f72924ec' [13:54] <cone-774> ffmpeg.git 03Rémi Denis-Courmont 07master:aaf9d0e8f396: vdpau: Add MPEG-1/2 decoding via hwaccel infrastructure [13:54] <cone-774> ffmpeg.git 03Rémi Denis-Courmont 07master:200e8ac92007: vdpau: Add MPEG-4 decoding via hwaccel infrastructure [13:54] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:41bdef1d3958: Merge commit '200e8ac92007bc2fe30da05d3bd00ab620842a6b' [14:05] <cone-774> ffmpeg.git 03Rémi Denis-Courmont 07master:51b56a071602: vdpau: Add H.264 decoding via hwaccel infrastructure [14:05] <cone-774> ffmpeg.git 03Rémi Denis-Courmont 07master:775253278914: vdpau: Add VC-1 decoding via hwaccel infrastructure [14:05] <cone-774> ffmpeg.git 03Anton Khirnov 07master:246da0b13551: v4l2: avoid pointless indirection. [14:05] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:53c2f401f726: Merge commit '246da0b13551b1f80f067e4f258e5bd691f5ab33' [14:20] <Compn> funman : so are you reviewing that h264 dxva2 deocder patch ? [14:21] <funman> Compn: i only had a quick look, i saw nothing strange [14:23] <cone-774> ffmpeg.git 03Anton Khirnov 07master:838b849e70f1: v4l2: set the average framerate instead of codec timebase. [14:23] <cone-774> ffmpeg.git 03Anton Khirnov 07master:49dc82eef776: v4l2: do not assert on a value received from outside of Libav [14:23] <cone-774> ffmpeg.git 03Anton Khirnov 07master:90cfc084e3e6: avpacket: free side data in av_free_packet(). [14:24] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:a646ac8ef5c5: Merge commit '90cfc084e3e6d37ab88fc96a95f0401d8e8b4cd1' [14:25] <Compn> funman : ok thanks [14:25] <Compn> michaelni : any chance you could review/commit the h264 dxva2 decoder for ffmpeg patch ? [14:26] <Compn> or say whats up on the list, rejected, need more info, etc [14:27] <funman> Compn: except missing copyright of vlc authors though [14:30] <Compn> Subject: [FFmpeg-devel] [PATCH] libavcodec:add h264_dxva2 decoder correct the copyright line [14:30] <michaelni> Compn, iam happy to leave review to some other volunteer if there is one [14:31] <Compn> michaelni : i've been asking few people ... :) [14:31] <Compn> maybe i need to ask some vlc devs [14:34] <nevcairiel> review would be much easier if the inline patch wouldn't be mangled to death :p [14:35] <Compn> theres an attached patch as well ... [14:36] <Compn> nevcairiel : are you going to review it? [14:38] <Compn> i forgot , i think i asked you [14:38] Action: Compn losing brain cells [14:38] <michaelni> Compn, if all else fails you have to review it :) [14:38] <Compn> :) [14:39] <Compn> well there was some small things that could be fixed [14:40] <Compn> but i didnt want to bug patch author with those until i knew if the patch was correct / right design [14:40] <nevcairiel> i wonder if that person finds my comments in the middle of his patch, or if i should write a big text blob at the end :p [14:40] <nevcairiel> i think the whole design is terrible, but apparently noone wants to hear that [14:41] <nevcairiel> features above anything, i guess [14:41] <Compn> the hwaccel design [14:41] <Compn> ? [14:41] <nevcairiel> its not a hwaccel, thats the problem :P [14:41] <Compn> wasnt his first patch for the hwaccel ? [14:42] <Compn> before someone said to use the other api... [14:42] <divVerent> what does the new API do better/different? [14:43] <nevcairiel> there is no new api [14:43] <nevcairiel> its just a wrapper around the hwaccel so stupid applications can use it [14:43] <divVerent> the "hwaccel infrastructure" [14:43] <divVerent> so the old way that e.g. mplayer uses stays supported? [14:44] <Compn> >code copied from vlc, blame mplayer [14:44] <Compn> :P [14:44] <divVerent> that means? [14:44] <Compn> no idea, i'm not the person who decides what api is good or bad [14:44] <Compn> i dont have that power [14:44] <Compn> nor expertise [14:45] Action: michaelni wants fate tests for hwaccel [14:45] <Compn> michaelni : what about nevcairiel 's and divVerent 's comments about some hwaccel mess and wrappers that mplayer uses ? [14:46] <divVerent> mplayer used "some hwaccel" for H.264, VC-1 and the other two for ages now [14:46] <divVerent> for VDPAU support [14:46] <divVerent> I am mainly wondering, what [14:46] <divVerent> s the direction we're going, as apparently a new "hwaccel" API is being made [14:46] <nevcairiel> vdpau is a even worse hack in avcodec =p [14:47] <divVerent> well, both are in avcodec [14:47] <Compn> did you see that new vdpau that was committed ? i thought it was there already ... [14:47] <Compn> 'new' ... 2008 copyright from nvidia [14:47] <divVerent> yes, that's why I am asking [14:47] <nevcairiel> there is a AVHWAccel infrastructure, which should be the way to go forward, and vdpau was finally converted to that after years and years [14:47] <divVerent> thing is, we're already USING "some api" to decode H.264 through VDPAU with ffmpeg's help [14:47] <divVerent> so the old API is going to be removed? [14:48] <nevcairiel> not without a major bump, but personally, i hope so [14:48] <divVerent> what's better with the new one? [14:48] <Compn> whats the difference between them ? [14:48] <divVerent> also: [14:48] <divVerent> 14:40:54 nevcairiel | i think the whole design is terrible, but apparently noone wants to hear that [14:48] <nevcairiel> the integration with the decoders is much cleaner [14:48] <divVerent> so the old API#s design is MORE terrible? [14:48] <nevcairiel> divVerent: that was about the dxva patch currently on review [14:49] <cone-774> ffmpeg.git 03Luca Barbato 07master:d9cf5f516974: lavc: check for overflow in init_get_bits [14:49] <cone-774> ffmpeg.git 03Luca Barbato 07master:e28ac6e5e27e: lavc: introduce the convenience function init_get_bits8 [14:49] <cone-774> ffmpeg.git 03Daniel Kang 07master:67360ccd51f3: yadif: x86: fix build for compilers without aligned stack [14:49] <cone-774> ffmpeg.git 03Anton Khirnov 07master:8fd4d1f9b925: cscd: return meaningful error codes. [14:49] <cone-774> ffmpeg.git 03Anton Khirnov 07master:9e6764900a07: msrle: return meaningful error codes. [14:49] <cone-774> ffmpeg.git 03Anton Khirnov 07master:6bf70127d1d4: tmv: return meaningful error codes. [14:49] <cone-774> ffmpeg.git 03Anton Khirnov 07master:a1c525f7eb07: pcx: return meaningful error codes. [14:49] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:329675cfd71f: Merge commit 'a1c525f7eb0783d31ba7a653865b6cbd3dc880de' [14:49] <divVerent> how would code using the new API even look? [14:50] <wm4> what new api [14:50] <divVerent> I know that what mplayer does is "somehow related to CODEC_CAP_HWACCEL_VDPAU" [14:50] <wm4> the old way is selecting specific decoders [14:50] <wm4> for vdpau [14:50] <wm4> the new way, you just use the correct decoder, and request a hardware pixel format [14:50] <nevcairiel> oh my the decoder uses a static data buffer [14:50] <nevcairiel> that cant go well <.< [14:51] <wm4> wat [14:51] <Compn> wm4 : oh, no more mplayer -vc ffh264vdpau,ffvc1vdpau,ffmpeg2vdpau ? [14:51] <Compn> ehe [14:51] <Compn> hacks upon hacks [14:51] <nevcairiel> it should really behave more like any other decoder if it wants to be one [14:52] <wm4> Compn: no, that already works in mpv (mpv -hwdec vdpau -vo vdpau file.mkv) [14:52] <divVerent> and is there documentation on how to change the code to the new API? [14:52] <wm4> divVerent: there never is [14:52] <divVerent> as I suppoose choosing the pixel format isn't everything [14:52] <divVerent> well, USUALLY there is at least documentation in the headers [14:52] <divVerent> but that's not existing for this either [14:53] <Compn> i've seen various reports about 10bit not working with vdpau [14:53] <Compn> 10bit h264* [14:53] <wm4> Compn: mpv will fallback to software decoding in this case [14:53] <nevcairiel> obviously, the hardware only does 8-bit [14:53] <nevcairiel> and only 4:2:0 [14:53] <nevcairiel> and only main/high profile [14:53] <nevcairiel> and i guess baseline too =P [14:53] <Compn> nevcairiel : how would a calling application figure out 10bit before sending it to a decoder ? [14:53] <divVerent> or is the only difference between the APIs how the decoding is set up? [14:53] <divVerent> I see the new one also uses AV_PIX_FMT_VDPAU [14:54] <wm4> divVerent: the new one introduced that pixel formats; before that each codec had its own pixel format because mplayer needs that [14:54] <divVerent> ah, I see [14:54] <wm4> divVerent: as the decoder is created in the VO instead of the decoder wrapper [15:04] <durandal_1707> what happened to ubitux? [15:06] <mateo`> durandal_1707: he's on vacation [15:06] <durandal_1707> where? [15:08] <pigoz> south pole [15:09] <cone-774> ffmpeg.git 03Anton Khirnov 07master:3344f5cb747b: nuv: return meaningful error codes. [15:09] <cone-774> ffmpeg.git 03Anton Khirnov 07master:62d9655217f9: qpeg: return a meaningful error code. [15:09] <cone-774> ffmpeg.git 03Anton Khirnov 07master:01cbc6f6adf5: targa: return meaningful error codes. [15:09] <cone-774> ffmpeg.git 03Anton Khirnov 07master:14cf33e957fb: lcldec: return meaningful error codes. [15:09] <cone-774> ffmpeg.git 03Anton Khirnov 07master:089b3d6815a8: interplayvideo: return meaningful error codes. [15:09] <cone-774> ffmpeg.git 03Anton Khirnov 07master:3d973e461be6: interplayvideo: remove a static variable. [15:09] <cone-774> ffmpeg.git 03Anton Khirnov 07master:048ffb9bb26f: gifdec: return meaningful error codes. [15:09] <cone-774> ffmpeg.git 03Anton Khirnov 07master:8f1782945530: qtrle: return a meaningful error code. [15:09] <cone-774> ffmpeg.git 03Anton Khirnov 07master:688b132b881d: qdrw: return meaningful error codes. [15:09] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:8943925d1f68: Merge commit '688b132b881d423877e38dc82f17e23a604be676' [15:09] <mateo`> durandal_1707: japan [15:11] <michaelni> j-b, i just got "remote: mail: /tmp/mail.RsXXXXRzM96x: No space left on device" [15:13] <durandal_1707> OOO [15:13] <michaelni> seems the push worked fine though otherwise [15:27] <ubitux> hey, japan is connected to internet! [15:27] <ubitux> @_@ [15:28] <michaelni> ubitux, :)))))) [15:28] <JEEB> ubitux, yes -- if you are talking about public wifis they are indeed scarce as hell [15:28] <JEEB> it's easier to get into an internet cafe than find a publically usable wifi spot [15:28] <JEEB> even in Starbucks! [15:29] <ubitux> :) [15:29] Action: JEEB still rememembers him wardriving through Tokyo station's Yaesu exit [15:30] <JEEB> Starbucks -> "Sorry, we don't have wifi. But I think the McDonalds next to us has." -> McDonalds -> "Sorry, but the WiFi is only for Yahoo! BB users." -> ... [15:31] <JEEB> I think the widest free wifi in Tokyo is the Tokyo Metro one [15:31] <mateo`> ubitux: \o/ [15:31] <JEEB> there were some WEP networks in there, I just wish my eeepc had more than 2h of battery power :V [15:32] <JEEB> (@Yaesu Exit) [15:32] <cone-774> ffmpeg.git 03Anton Khirnov 07master:80cf2ebc59ba: qdrw: cosmetics, reformat [15:32] <cone-774> ffmpeg.git 03Anton Khirnov 07master:e4ca055b3ce3: rpza: return a meaningful error code. [15:32] <cone-774> ffmpeg.git 03Anton Khirnov 07master:74a9a624c5c4: vcr1: return a meaningful error code. [15:32] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:4adc8a29a056: Merge commit '74a9a624c5c4d50760d8d57458eba57366f6cb26' [15:33] <mateo`> is dnxhd-tr standardized ? [15:40] <cone-774> ffmpeg.git 03Anton Khirnov 07master:6b3a1229bb30: vcr1: remove disabled encoder stub [15:40] <cone-774> ffmpeg.git 03Anton Khirnov 07master:a9d970a019df: flicvideo: return meaningful error codes. [15:40] <cone-774> ffmpeg.git 03Anton Khirnov 07master:1a31dff9370b: loco: return meaningful error codes. [15:40] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:f9d5bdd0f652: Merge commit '1a31dff9370b4732c91df5cb1ca4b39f2cb3050e' [15:51] <cone-774> ffmpeg.git 03Anton Khirnov 07master:b627c3010b9b: loco: cosmetics, reformat [15:51] <cone-774> ffmpeg.git 03Anton Khirnov 07master:ac1e93f555aa: tscc: return meaningful error codes. [15:51] <cone-774> ffmpeg.git 03Anton Khirnov 07master:a4a26f518856: tscc: remove some pointless comments and empty lines. [15:51] <cone-774> ffmpeg.git 03Anton Khirnov 07master:df9036830b15: truemotion2: return meaningful error codes. [15:51] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:92b238715929: Merge commit 'df9036830b15997a3e9c3f2c632ed98d64f9deef' [16:21] <cone-774> ffmpeg.git 03Anton Khirnov 07master:60a42ef44cc0: truemotion2: cosmetics, reformat [16:21] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:59d568031008: h264: Fix assignments in if() [16:21] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:1d7ffd06e41e: lavc: Fix assignments in if() when calling ff_af_queue_add [16:21] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:30aeada3d8e9: svq1enc: Fix assignments in if() [16:21] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:d641ee94b57a: lavf: Fix assignments in if() [16:22] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:6dc8505417c9: rtmpproto: Fix assignments in if() [16:22] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:57d59e82cbdd: Merge remote-tracking branch 'qatar/master' [16:29] <cone-774> ffmpeg.git 03Carl Eugen Hoyos 07master:c29c7c1470f9: The c99-to-c89 binaries are now hosted on videolan.org. [16:36] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:ac73d3a12a33: init_get_bits8: check byte_size against being positive [16:36] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:153fad14e5a2: init_get_bits8: zero pointers & struct on error [16:36] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:7980cca05c7c: init_get_bits: fix off by 1 error [17:10] <cone-774> ffmpeg.git 03Michael Niedermayer 07master:922332e60df3: nuv: remove unused variable [17:14] Action: Compn remembers paying 100Y for 10 minutes internet [17:14] <Compn> stupid hotel rates! [17:15] <Compn> that was before internet cafes made it big [17:26] <Jumpyshoes> michaelni: yadif should be fixed [17:59] <mateo`> sorry to ask again, is dnxhd-tr standardized ? [18:13] <durandal_1707> there is no dnxhd spec? [18:14] <merbanan> smpte vc-3 [18:15] <merbanan> but lots of stuff is not specified [18:26] <kierank> mateo`: no, cid 1244 is not specified in vc-3 [18:31] <mateo`> kierank: i was looking for an update of the SMPTE document :( [18:31] <kierank> i don't think there is a revision [19:14] <michaelni> Jumpyshoes, thanks! [22:11] <Skyler_> if someone wanted to pay for an LXF demuxer/decoder(?), who should I point them to? [22:12] <Skyler_> (is that format related to MXF?) [22:12] <Skyler_> bcoudurier: ^ [22:12] <kierank> it has been done [22:12] <Skyler_> oh, already supported? [22:12] <kierank> yes [22:12] <kierank> ifb wrote a patch for newer variants [22:13] <kierank> but it didn't get accepted into libav [22:13] <kierank> because at the time they were rewriting the whole audio side [22:13] <Skyler_> okay, will poke them with that info [22:14] <kierank> http://lists.libav.org/pipermail/libav-devel/2011-December/017642.html [22:14] <kierank> 21:12:33 <"Skyler_> (is that format related to MXF?) --> only in that it is a broadcast codec [22:14] <kierank> container* [22:15] <Skyler_> I can't find it on the changelog on the website [22:15] <Skyler_> is it not in a release yet? [22:16] <beastd> should be in an ffmpeg release if i am not mistaken [22:17] <kierank> Skyler_: do they need vbi support? [22:17] <Skyler_> I don't know, they'll poke me if they do I suppose [22:17] <kierank> because there's no chance in hell that'll work any time soon in lavc [22:45] <Compn> Demuxer for Leitch/Harris' VR native stream format (LXF) [22:46] <Compn> is that the same LXF format Skyler_ ? [22:46] <Compn> it sounds like an old format ... [22:57] <Skyler_> Yes it's that [22:58] <pettter> is there any UPnP/DLNA server currently using FFmpeg for input/output/transcoding? [23:03] <cone-902> ffmpeg.git 03Michael Niedermayer 07master:30981a966f17: lavc: split snow and dirac DWTs [23:17] <Compn> pettter : oh, i remember a project doing that using ffmpeg [23:17] <Compn> but i dont remember now [23:18] <Compn> either xbmc or ps3mediaserver [23:18] <Compn> probably xbmc... [23:18] <Compn> XBMC also have a built-in UPnP-server (since the 24th of January 2007) [23:18] <Compn> PS3 Media Server is a DLNA compliant Upnp Media Server for the PS3, written in Java [23:18] <Compn> ah both of them :) [23:19] <pettter> Compn: right [23:19] <Compn> so whats the question ? [23:20] <pettter> Compn: basically, I'm running raspbmc (XBMC on a raspberry pi) and am considering various ways to get transcoding and streaming to work [23:20] <pettter> running xbmc on the media server _as well_ feel excessive... [23:20] <pettter> currently setting up minidlna, we'll see how it goes [23:24] <Compn> well , you expect the raspberry pi to transcode ? [23:25] <Compn> i dont think it has much cpu [23:27] <kierank> it has a hardwaren encoder [23:33] <cone-902> ffmpeg.git 03Michael Niedermayer 07master:9cb3c1a4d9db: x86/dirac: fix asm on win64 [23:45] <cone-902> ffmpeg.git 03Michael Niedermayer 07master:274e48d8ac1e: x86/Makefile: move dirac_dwt to right type [23:53] <Compn> kierank : still have to decode it , if its not h264 already [23:53] <kierank> it has mpeg-2/vc-1/h264 decoder [23:53] <Compn> not those then [23:54] <gnafu> Well, you have to pay for MPEG-2 and VC-1, so I could see situations where someone might want to reencode even those on another machine ;D. [23:55] <gnafu> You could reencode all non-H.264 content to H.264 that the Pi can play out of the box. [23:56] <Compn> they can have my money when they pry it from my cold dead hands [23:59] <gnafu> Hehe. [00:00] --- Tue Jan 15 2013

1 0

ffmpeg.log.20130114
by burek 15 Jan '13

15 Jan '13

[01:10] <p1r473> hey guys. having a problem where ffmpeg is guessing channel layout to be stereo when its really 5.1. heres a screenshot http://imgur.com/JWMvG [01:10] <p1r473> ive tried replacing the ffmpeg binary that came with avstovideo with one from a rpeository, but I think the console commands/piping is different because it fails immediately [01:10] <p1r473> sorry, avstodvd [07:14] <p1r473> whats the difference between 5.1 and 5.1(side)? [09:10] <cbsrobot> p1r473: I geuss simple 5.1 means 5.1(rear) [10:17] <dott_infernus> hi. [10:17] <Mavrik> mornin [10:18] <dott_infernus> I'm angelo and I'm looking for some help [10:18] <dott_infernus> can anyone help me? [10:18] <Mavrik> well [10:18] <viric> day of the tentacle [10:19] <Mavrik> dott_infernus, without your explanation of what's wrong... not really ;) [10:19] <dott_infernus> :-) ok...I will explain very well soon [10:21] <dott_infernus> I'm trying to handle video file with a program made by me in c++ (and qt), I've read that ffmpeg gives the opportunity to manage them....so what I need is a good tutorial to install the ffmpeg library on my computer [10:21] <dott_infernus> my os is ubuntu [10:21] <Mavrik> you're in luck [10:21] <Mavrik> dott_infernus, http://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [10:22] <Mavrik> dott_infernus, this is the guide to compile and install newest ffmpeg [10:22] <Mavrik> you can also get it from the repository but that's usually obsolete [10:22] <dott_infernus> repository? [10:23] <Mavrik> with apt-get [10:23] <cbsrobot> dott_infernus: and when you have cloned the ffmpeg repo check the "examples" directory [10:24] <dott_infernus> I tried apt-get install ffmpeg and it gives me the program to convert or play videos [10:26] <Mavrik> dott_infernus, the ffmpeg libraries (which you need to use to to manual transcode) are named libavcodec (en/decoders), libavformat (de/muxers), libswscale (image scalers), libavutil, libavfilter and libswresample [10:26] <Mavrik> those are the packages you need to look for [10:28] <dott_infernus> Ok...I will take note...actually, I tried yesterday to follow that guide, but some error occured...Now I don't remember which error came ut [10:29] <dott_infernus> but If I download this archive: http://ffmpeg.org/releases/ffmpeg-snapshot.tar.bz2, is it possible to integrate that library into my code? [10:30] <Mavrik> dott_infernus, you'll have to compile that and install it [10:30] <Mavrik> and then link against those libraries in your code yea [10:30] <Mavrik> it's not that easy though, are you sure you know what you're getting into? :) [10:31] <dott_infernus> well, if I don't know, I will know very soon..It's work.. [10:45] <p1r473> cbsrobo: ok whats the diff between 5.1 reat and 5.1 side? [10:45] <p1r473> rear* [11:14] <cbsrobot> p1r473: you need to spell my nick right or I wont see your ping ... [11:14] <cbsrobot> see http://files.tested.com/uploads/0/5/16018-speakerplacement_7_1.gif [11:14] <cbsrobot> suddound back is rear, surround right/left is side [11:15] <p1r473> ooh. isnt rear thr norm? shudnt people just encode for rear? [11:16] <cbsrobot> i think side is normal [11:16] <cbsrobot> you add the backs for a 7.1 system [11:17] <Mavrik> hmm, I'm quite sure you add sides for 7.1 [11:17] <p1r473> yea i thought rear was norm [11:18] <p1r473> only seen 5.1(sides) once in ffmpeg [11:18] <cbsrobot> well maybe [11:18] <cbsrobot> I don't remember exactly [11:19] <cbsrobot> our little cinema has sides, not rears [11:19] <p1r473> 2nd question, sometimes ffmpeh guesses channel layout as stereo but its 5.1. why does it mess it up sometimes? [11:19] <p1r473> and how do u fix it? [11:19] <cbsrobot> but it has only 24 places & so [11:20] <Mavrik> it's possible cinema setups are different, yea :) [11:26] <TeknoJuce> hi trying to capture with a dvc100 it seems to start capturing then fails http://pastie.org/5682152#44-45 [11:27] <TeknoJuce> /ioctl(VIDIOC_DQBUF): No such device [11:27] <TeknoJuce> /dev/video0: No such device [11:31] <TeknoJuce> http://pastie.org/5682170 [11:43] <viric> TeknoJuce: does the kernel log say anything? [11:45] <viric> Does x264 encoding preset (slow, fast, ...) affect the complexity of the *decoding*? [11:45] <viric> so, are h264 players constraint by any specific complexity in encoding? [11:46] <Mavrik> viric, not by preset, but by profile and level [11:46] <Mavrik> viric, or more specifically, by profile (baseline, main, high) and resolution+bitrate combo [11:46] <viric> aha [11:46] <Mavrik> on pretty much all decoders you'll see specification of what the highest level and profile they support is [11:47] <viric> umh I think I always mixed profile and preset. [11:47] <Mavrik> preset just sets how much time x264 spends trying to get the best result possible :) [11:47] <viric> maybe not on cheap $60 TV decoders [11:47] <Mavrik> even on those :) [11:47] <viric> (about see the specification) [11:48] <Mavrik> sometimes they just say "Blu-ray compatible" or something like that [11:48] <viric> ok, I'll try to find out [11:48] <viric> what's the default profile? [11:48] <Mavrik> by default the profile depends on output [11:48] <Mavrik> main for most, high for HD content [11:48] <viric> [libx264 @ 0x956a260] profile High 4:2:2, level 3.0, 4:2:2 8-bit [11:48] <Mavrik> um yea [11:48] <viric> it's PAL. [11:48] <viric> Stream #0:0: Video: h264, yuv422p, 720x576 [SAR 1:1 DAR 5:4], q=-1--1, 1k tbn, 25 tbc (default) [11:48] <Mavrik> also... most decoders won't be able to display YUV422 [11:49] <Mavrik> so watch out for that :) [11:49] <viric> no? [11:49] <viric> damn it [11:49] <viric> what has it to be? [11:49] <Mavrik> only 4:2:0 support is officialy standardized :\ [11:50] <viric> ok, ffmpeg can convert that I guess [11:50] <viric> but my input was 422 [11:50] <Mavrik> yeah, sadly that's the case [11:50] <viric> So, do you think it's normal to have the 'high' profile? [11:50] <Mavrik> um, not really [11:50] <Mavrik> there aren't many PAL decoders out there supporting "high" profile [11:50] <Mavrik> old mobiles don't even support main profile [11:51] <viric> I use: -vcodec libx264 -preset normal -acodec libvorbis [11:51] <Mavrik> I suggest you make a YUV420 video with main profile [11:51] <Mavrik> and use AAC please if you have a hardware decoder :D [11:51] <Mavrik> vorbis really isn't a widely supported format [12:03] <TeknoJuce> viric http://pastie.org/5682271 [12:04] <viric> Mavrik: yes I was just testing. [12:07] <TeknoJuce> viric does that look like the mobo usb hub is dropping the capture device? [12:16] <viric> TeknoJuce: looks like so [13:03] <amanforindia> where can I find good documentation for the libavformat library?? [13:04] <JEEB> http://ffmpeg.org/doxygen/trunk/group__libavf.html [13:04] <JEEB> the doxygen would be the first thing to check [13:05] <amanforindia> ok... thanks :D [13:05] <Mavrik> amanforindia, also, doc/examples dir in ffmpeg [13:05] <JEEB> also google with 'site:ffmpeg.org/doxygen/trunk/ keyword' is usually useful [14:21] <IamTrying> When i am converting this 24MB file to flv, why its not fitting in 4 or 5MB, i wanted to send it over 3G mobile (whatsApp to my Gf) but the converted file is very large 21MB, how to lower it? e.g: ffmpeg -i HeyGirl.mpeg -f flv newfile.flv [14:23] <IamTrying> -s qsif? [14:23] <Mavrik> IamTrying, you're not setting the quality of flv you want [14:24] <Mavrik> so ffmpeg just defaults to... something [14:24] <Mavrik> IamTrying, you are looking for bitrate parameter (-b:v) [14:24] <IamTrying> Still 18.2MB Mavrik while using -s sqcif [14:24] <Mavrik> -s is just size of the actual video picture [14:24] <IamTrying> Mavrik, -b for video and audio? [14:26] <jeroen-> How to check if an avconv process uses multiple cores? [14:26] <Mavrik> yes, -b tells ffmpeg how much bits you want each second of your video to take [14:27] <Mavrik> jeroen-, on whch os? [14:27] <IamTrying> Mavrik, -b 10 -s qcif still its 14.5MB [14:27] <jeroen-> Mavrik: kubuntu 2.10 [14:27] <jeroen-> 12.10 [14:27] <IamTrying> Mavrik, i need to have it in 5MB from 24MB i mean. [14:27] <Mavrik> IamTrying, you cant make video with 10 bits per second :P [14:27] <Mavrik> IamTrying, how long is your video (in seconds)? [14:28] <IamTrying> well in small resolution -b 10 is ok [14:28] <IamTrying> Mavrik, 4.54 seconds (Richard markx - hold on to the night music + her pictures). [14:28] <sacarasc> That is 10 bits per second. Or 1.25 bytes per second. [14:28] <Mavrik> IamTrying, well, try something like "-b:v 300k" :) [14:30] <IamTrying> Mavrik, ffmpeg -i My\ Cutest\ Baby.mpeg -b:v 300k -f flv newfile.flv # its still 21MB [14:31] <Mavrik> that I find very hard to believe. [14:31] <Mavrik> if the video is really 4 sec. [14:33] <IamTrying> Mavrik, 4 minute 54 seconds (audio/video) [14:36] <IamTrying> -r 5 -b 10 -s sqcif gives me still 13.3 MB . I need to have it in 5MB. [14:36] <jeroen-> anyone? [15:08] <viric> can anybody suggest some nice pocket camera, specifically for video? [15:08] <viric> maybe pocket is a bad idea though. Not very big atl east. [15:24] <Podium> Hello All (first time here), I have a question about the use of "ffmpeg". In the broadcast world we recored video by dedicated hardware in ProRes HQ 422 format with pcm_s24le (lpcm / 0x6D63706C), 48000 Hz, 16 channels, s32, 18432 kb/s [15:24] <Podium> I now transcode the video succesfully to h264... [15:25] <Podium> But the audio is the problem, I want only the first 2 audio channels in stream 1 to transcode... An this seems impossible. [15:26] <Podium> The error I get is:Input channel layout isnt supported 0kB time=00:00:00.00 bitrate= 0.0kbits/s [15:26] <Podium> swr_init() failed [15:26] <Podium> Can not resample 16 channels @ 48000 Hz to 2 channels @ 44100 Hz [15:26] <JEEB> uh-oh [15:26] <JEEB> that sounds like fun :D [15:27] <Mavrik> ^^ [15:27] <Podium> :-) Sounds :-) [15:27] <Mavrik> 16-channels? fun :D [15:27] <Mavrik> Podium, anyway, can you give us full output (+ command line) to a pastebin so we have more details :) [15:28] <JEEB> if that still fails with current ffmpeg (there are often quite up-to-date windows builds available for windows f.ex. from zeranoe), then I recommend you go to the issue tracker and report the issue [15:28] <Podium> http://pastebin.com/AW3nCJAw [15:29] <Podium> But I don't want to resample... I just want to delete 14 of the 16 channels... [15:29] <Mavrik> you're doing resampling as well (your input is 48000, output is 44100) [15:30] <Mavrik> I suggest you upgrade to ffmpeg 1.1 first [15:30] <Mavrik> since it seems it chokes on your 16-channel input [15:30] <Mavrik> and it might have been fixed since 0.10.6 [15:30] <JEEB> ah, then you shouldn't use -ac 2 [15:30] <JEEB> there was some specific way to remove channels completely from input [15:31] <Mavrik> using the "pan" audio filter [15:31] <Mavrik> you can define which channels map to where and what to keep [15:31] <Mavrik> http://ffmpeg.org/ffmpeg-filters.html#pan [15:32] <Mavrik> but I'm worried about the "INPUT channel layout not supported" [15:32] <Podium> @Mavrik: I couldn't find a newer repos for Ubuntu.... And channel_map doesnt seems to work [15:32] <JEEB> Mavrik, he /should/ be able to specify a stereo map after just removing all other channels [15:32] <JEEB> in theory [15:32] <Mavrik> yep [15:32] <Mavrik> Podium, try using the pan filter if it exists in your version [15:32] <Mavrik> to define mapping [15:33] <Mavrik> Podium, and grab a new build from here: http://dl.dropbox.com/u/24633983/ffmpeg/index.html [15:33] <Podium> Okeej, I will go ahead read the pan mapping documentation! [15:35] <Podium> @Mavrik, can I download that build an just replace the old one!? In Ubuntu I normally only use reposotories [15:35] <Mavrik> Podium, that's a totally static binary [15:35] <Mavrik> you just extract and run [15:35] <Mavrik> no dependencies [15:35] <Podium> okeej [15:39] <Podium> With the downloaded build I now get something totally diffrent: Option 'enable' not found [15:39] <Podium> [AVFilterGraph @ 0x29f5ca0] Error initializing filter 'yadif' with args '0:-1:0' [15:39] <Podium> Error opening filters! [15:40] <JEEB> I think the way you give arguments to yadif got changed [15:40] <JEEB> and/or it is/was broken for a while [15:50] <Hfuy> Hello. I have a large uncompressed AVI that I'd like to convert to ProRes LT. Is there a Windows build of ffmpeg that will do this for me? [15:57] <zap0> yes [16:05] <Hfuy> Any idea where I might get one? [16:06] <JEEB> zeranoe's builds for windows are usually relatively up-to-date [16:07] <Hfuy> Are the 64-bit versions worth having, or should I play it safe? [16:11] <JEEB> should be just fine [16:12] <Hfuy> It's snowing. [16:12] Action: Hfuy is cold [16:18] <hub2> Hi,Can you tell me what should I change in my code to use libvo-aacenc?Here's my code http://pastebin.com/qBWT0FhE [16:25] <luneff> hi, guys. i'm using ffmpeg with video from pipe (ppm stream) and a mp3 file. how do i say ffmpeg to stop encoding on pipe EOF so that it doesn't wait for audio to finish? [16:27] <luneff> aresample? [16:29] <TeknoJuce> is there a way to mount a channel from video4linux2 to like /dev/video1? [16:29] <Mavrik> luneff, I think you're looking for "-shortest" parameter [16:29] <Mavrik> which stops encoding when shortest stream ends [16:30] <luneff> Mavrik, thanks a lot :-) [16:30] <TeknoJuce> like if you have a capture card with video0 channel 1 2 3 4 could you make them video0 1 2 3 4 some how? [16:42] <Podium> @Mavrik: I come now to the conclussion the build of ffmpeg does not work at all here. Not even with the video only transcode. But no error messeges. He thinks he is done after one frame... [16:43] <dott_infernus> hi. [16:43] <Mavrik> Podium, ugh, strange [16:44] <dott_infernus> I need help.. I just installed ffmpeg libraries on my ubuntu but when I try to compile some example in /doc/example some errors have occured [16:44] <Podium> Yeah... Do the audio filters (-af) also work with mine repos version off ffmpeg? [16:44] <dott_infernus> undefined reference to `av_get_channel_layout_nb_channels' [16:45] <Mavrik> Podium, maybe... they're pretty new and I don't remember when were they introduced [16:45] <Mavrik> 0.10.6 is pretty new though [16:45] <Mavrik> dott_infernus, are you actually linking with ffmpeg libraries? [16:46] <Podium> @Mavrik.... Sorrry... I see I have deleted/overwrited my original :-) [16:46] <Podium> It's only 0kb [16:46] <Mavrik> ah :D [16:46] <dott_infernus> Mavrik, yes I am. I receive this error: undefined reference to `av_get_channel_layout_nb_channels' [16:47] <Podium> It's a long day... I'll go 2morrow try again! Thanks for the help ! [16:48] <dott_infernus> Mavrik, sorry, I've posted the wrong error...this is the actual: decoding_encoding.c:(.text+0x1831): undefined reference to `avcodec_register_all' [16:48] <Mavrik> yeah [16:48] <Mavrik> you're getting this because linker can't find functions in libraries [16:48] <Mavrik> now either you're not passing parameters to link against them [16:48] <Mavrik> or you're trying to link C++ code with C code without externs [16:49] <Mavrik> did you pass "-lavformat -lavcodec .." to linker?= [16:50] <dott_infernus> no, with terminal I call: gcc decoding_encoding.c -o out1 [16:50] <dott_infernus> what actually I have to call? [17:09] <dott_infernus> Mavrik, are you online? [17:27] <divVerent> http://www.youtube.com/watch?v=bh703B_xFAo [17:27] <divVerent> fun with a ffmpeg x11grab window [17:29] <Hfuy> When we point a camera at a monitor, we call it the "hall of mirrors" effect [17:30] <divVerent> yes [17:30] <divVerent> it's fun that it's delayed so much though [17:31] <Hfuy> Computers never have quite the performance we'd prefer. [17:41] <viric> umhh [17:41] <viric> I use '-acodec aac' [17:41] <viric> and ffmpeg says: [17:41] <viric> Stream #0:1: Audio: none, 44100 Hz, mono, flt, 128 kb/s (default) [17:41] <viric> Stream #0:1 -> #0:1 (flac -> aac) [17:42] <Mavrik> viric, er... use fdk-aac, faac or libvo-aacenc instead :) [17:42] <viric> ah [17:42] <viric> clear [17:42] <Mavrik> check ffmpeg -codecs to find which you have compiled in :) [17:43] <viric> nah I don't have it [17:43] <viric> I'll add it. [17:43] <viric> what are x264 profiles for? does 'main' mean less bitrate than 'high'? [17:44] <viric> - I fail to understand some very basic points :) [17:45] <madwill> Hi, we've been using ffmpeg to convert video in order to made educationnal content to read on the web for a while, but now on our latest install FFMPEG says we should use AVCONV. Anybody can point to where i can see the command line to convert any input into a mp4 using x264 and AAC audio ? [17:45] <madwill> IS the avconv warming because they installed the wrong FFMPEG ? [17:46] <madwill> is FFMPEG still alive or deprecated ? [17:46] <Mavrik> er, it's complicated [17:46] <Mavrik> basically, ffmpeg is still very much alive [17:46] <Mavrik> you're seeing that warning because some distros have switched to "avconv" from ffmpeg [17:46] <madwill> Oh [17:46] <Mavrik> and now warn that soon they won't provide binaries anymore [17:46] <JEEBsv> madwill: the project is just fine, the ffmpeg binary in the libav fork is deprecated and then later removed [17:46] <Mavrik> er [17:46] <Mavrik> won't provide binary with the name "ffmpeg" anymore [17:46] <Mavrik> so. :) [17:47] <JEEBsv> yeah [17:47] <madwill> Thank you, adding a bit of clarity [17:47] <JEEBsv> yeah, the message is hard to parse for a newbie [17:47] <madwill> So up to this its a personnal choice [17:47] <Mavrik> madwill, <shameless self-promotion> I've written a short guide on how to get web-ready mp4 with ffmpeg: https://www.virag.si/2012/01/web-video-encoding-tutorial-with-ffmpeg-0-9/ :) [17:48] <Mavrik> hm, I should fix that with fdk-aac [17:48] <madwill> Lovely Mavrik, No shame into plugging this "exactly what i'm looking for" post :) [17:48] <JEEBsv> madwill: it just means that if you are using the libav project's binaries you use 'avconv' as that's the up-to-date binary, while if you are using the ffmpeg project's binaries you use 'ffmpeg' [17:48] <JEEBsv> and libav's support is over at #libav [17:49] <JEEBsv> basically when elenril from libav updated a lot of stuff in ffmpeg.c, he renamed it to avconv.c and left ffmpeg.c to be [17:49] <dott_infernus> Hi, It's me, again. I' [17:49] <JEEBsv> the ffmpeg project then merged that to its ffmpeg.c [17:49] <madwill> It sounds weird, internal depabe on how to do things ? [17:50] <madwill> i guess i can google it if i'm curious about politics or forks [17:51] <Mavrik> yes, politics and forks ^^ [17:51] <JEEBsv> the good thing is that the development speed on both sides went way up [17:51] <JEEBsv> after the forking [17:53] <dott_infernus> I've installed the ffmpeg libraries on my ubuntu os, but when I try to compile the examples inside doc folder, the compiler gives me undefined reference....when I compile I only write on the terminal: gcc decoding_encoding.c -o out1 ... It's said to me that I don't pass something to the linker.. [17:54] <dott_infernus> which parameter I have to specify? [18:14] <quebre> hello [18:14] <quebre> i have newbie question [18:14] <quebre> how do i reencode 720p movie to DVDRip format ? so it would have from 4.4GB to 700MB ? [18:16] <13WAAJGCL> um [18:22] <quebre> i know it's hard.. [18:22] <quebre> ;P [18:28] <cbsrobot> quebre: what file, codec, size, type you want at the end ? [18:29] <cbsrobot> and why 700mb ? [18:30] <quebre> cbsrobot: because most of dvdrip movies are 700mb and they are watchable while it's not pain to send it over internet to my girl who have weak internet [18:31] <cbsrobot> and the first question ? [18:32] <quebre> hold on im pasting it on pastebin.ca so you can check it out [18:32] <quebre> http://pastebin.ca/2302056 <- thats the movie details, i want to convert it to dvdrip [18:34] <quebre> i hope it's doable... [18:37] <cbsrobot> try: ffmpeg -i input.mkv -vf scale=640:272 -c:v libx264 -crf 25 -preset:v slow -profile:v high -level 4.1 -c:a libvo_aac -b:a 128k -ac 2 -t 600 output.mkv [18:37] <cbsrobot> untested [18:37] <cbsrobot> that will encode 10 min - make the math to see if it fits 700 mb [18:37] <cbsrobot> if not, set crf to a higher value [18:38] <cbsrobot> if yes, encode or set the crf to a lower value and try again [18:39] <cbsrobot> and the size will be 640x272 which is pretty small [18:39] <cbsrobot> and if you compiled yourself and have libfdk_aac hten use this instead of libvo_aac [18:42] <quebre> cbsrobot: how can i check whether i have libfdk_aac ? [18:43] <cbsrobot> quebre: did you compile it yourself ? [18:43] <quebre> umm i think so [18:43] <cbsrobot> hehe [18:43] <quebre> was long time ago and i don't remember heh [18:43] <cbsrobot> what now [18:43] <JEEBsv> quebre: ffmpeg -codecs |grep "aac" [18:43] <cbsrobot> so first - upgrade ! [18:43] <JEEBsv> and see what is listed :P [18:44] <quebre> DEA.L. aac AAC (Advanced Audio Coding) (encoders: aac libfaac libvo_aacenc libaacplus ) [18:44] <quebre> D.A.L. aac_latm AAC LATM (Advanced Audio Coding LATM syntax) [18:44] <quebre> this ;P [18:44] <cbsrobot> so you dont have it [18:44] <cbsrobot> but you have libaacplus [18:44] <cbsrobot> so eighter use -c:a libaacplus -b:a 64k [18:45] <cbsrobot> or -c:a libvo_aacenc -b:a 128k [18:45] <quebre> pure magic but alright ;) [18:45] <quebre> going to try and convert it [18:45] <cbsrobot> and upgrade first & :) [18:45] <cbsrobot> so you can add libfdk_aac to the soup [18:46] <quebre> ok [18:47] <quebre> git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg <- this will be latest ? [18:48] <JEEBsv> yes [19:23] <quebre> cbsrobot: one question [19:23] <quebre> in the -output output.mkv can it be .avi instead of .mkv ? [19:24] <cbsrobot> quebre: eeeh - I didn't use avi for > 10 years, so no idea [19:25] <cbsrobot> why would you use avi ? [19:25] <quebre> dvdrip movies are in .avi format ;) [19:28] <cbsrobot> quebre: where did you get that from ? [19:29] <quebre> Unrecognized option 'crf'. [19:29] <quebre> i guess that's releated to -crf 25 option.. [19:31] <cbsrobot> quebre: update ... [19:32] <cbsrobot> and http://scenenotice.org/details.php?id=2081 [19:32] <cbsrobot> nothing mentiones avi there ... [19:32] <cbsrobot> I guess it is even on a banlist ! [19:33] <quebre> i see [19:33] <quebre> hm but i have the git ffmpeg [19:33] <quebre> $ /usr/local/ffmpeg-latest/bin/ffmpeg -version [19:33] <quebre> ffmpeg version N-48882-g922332e [19:33] <quebre> built on Jan 14 2013 19:28:21 with gcc 4.7 (Debian 4.7.2-4) [19:33] <quebre> configuration: --prefix=/usr/local/ffmpeg-latest [19:33] <quebre> and it complains about 'crf' [19:33] <quebre> /usr/local/ffmpeg-latest/bin/ffmpeg -i Cloud.Atlas.2012.R.C.720p.BRRip.x264.AC3-TRiNiTY.sample.mkv -vf scale=640:272 -c:v libx264 -crf 25 -preset:v slow -profile:v high -level 4.1 -c:a libvo_aac -b:a 128k -ac 2 -t 600 output.mkv [19:34] <quebre> this i did [19:36] <quebre> what -crf does ? [19:36] <viric> x264 setting, about the quality required [19:36] <viric> http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide#faq [19:36] <viric> forget the anchor [19:37] <quebre> anchor ? [19:37] <viric> #faq [19:37] <quebre> no 'anchor' word in #faq :P [19:38] <viric> ehem [19:38] <quebre> so should i skip the -crf option ? [19:38] <viric> html anchor [19:39] <quebre> i think i don't understand you ;P [19:39] <quebre> ah [19:39] <quebre> you mean http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide without #faq [19:40] <quebre> -preset:v slow [19:41] <quebre> is this correct ? [19:46] <viric> I'm not any good at this. just learning [19:54] <quebre> this is not working [19:54] <quebre> http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide#twopass [19:54] <quebre> here is example given: [19:54] <quebre> ffmpeg -i input -c:v libx264 -preset fast -b:v 555k -pass 1 -an -f mp4 -y /dev/null [19:54] <quebre> and when i do it like that: [19:54] <quebre> $ /usr/local/ffmpeg-latest/bin/ffmpeg -i Cloud.Atlas.2012.R.C.720p.BRRip.x264.AC3-TRiNiTY.sample.mkv -c:v libx264 -preset veryslow -b:v 555k -pass 1 -an -f mp4 -y /dev/null [19:54] <quebre> i get: [19:54] <quebre> Unrecognized option 'preset'. [19:54] <quebre> Error splitting the argument list: Option not found [19:54] <quebre> ..... [20:05] <cbsrobot> and please show me the first command you did [20:05] <Sashmo_> some told me that there is some C library out there that can do very good PAL-NTSC conversion, but I dont have a name, anyone know anything about it? [20:05] <cbsrobot> with the -crf option [20:05] <quebre> for now i'm following https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [20:05] <quebre> want to have everything installed correctly before complaining ;) [20:06] <cbsrobot> quebre: good idea [20:21] <viric> is amd64 a big win for x264 encoding, compared to i686? [20:39] <sesam> hi [20:39] <sesam> how do u guys stitch together h264 files without reencoding em? [20:40] <Fjorgynn> ffmpeg -i file1.mp4 -i file2.mp4 -c:a copy -c:v copy output.mp4 [20:40] <Fjorgynn> sesam: right? [20:40] <sesam> ill try that [20:40] <sesam> thx [20:42] <quebre> okay okay [20:42] <quebre> it works now ! [20:42] <quebre> but :P [20:42] <quebre> cbsrobot: wake up ;) [20:43] <quebre> i'm doing Two-Pass conversion [20:43] <quebre> like this: [20:43] <quebre> ffmpeg -i Cloud.Atlas.2012.R.C.720p.BRRip.x264.AC3-TRiNiTY.sample.mkv -c:v libx264 -preset veryslow -b:v 555k -pass 1 -an -f mp4 -y /dev/null && ffmpeg -i Cloud.Atlas.2012.R.C.720p.BRRip.x264.AC3-TRiNiTY.sample.mkv -c:v libx264 -preset veryslow -b:v 555k -pass 2 -c:a libfaac -b:a 128k output.mp4 [20:43] <quebre> from this, the most important option is bitrate - 555k, i taken 555k from the example on the site [20:43] <quebre> but i don't know how to calculate fair value, [20:44] <quebre> also, how to convert the movie so the main condition is "get as good quality as possible BUT don't exceed 800MB" [20:44] <viric> why 800MB? [20:45] <quebre> because it's less than 1gb and it's about transfering the file over internet and not to wait too long [20:45] <viric> ah ok [20:45] <viric> I thought it was about some media size [20:45] <quebre> so, how to achieve my goal ? [20:45] <quebre> ;) [20:46] <viric> 555 kilobit/second + 128 kilobit/second [20:46] <viric> get the number of seconds, multiply, and you have the number of bits for the film (muxing apart) [20:47] <viric> you know that 8 bits make one byte, ... and that's all [20:47] <Fjorgynn> quebre: why not just use -crf 21 ? [20:47] <viric> Fjorgynn: why 21? [20:47] <Fjorgynn> instead of 2 pass encoding och bitrate [20:47] <Fjorgynn> viric: because take 23, 28 or 16 [20:47] <Fjorgynn> whatever you like [20:48] <quebre> Fjorgynn: what does -crf 21 ? [20:48] <Fjorgynn> 21 is common for documentaries etc. [20:48] <Fjorgynn> quebre: -crf is awesome, because then you don't need to use 2 pass encoding [20:48] <viric> I don't see the source for the magic value yet. [20:48] <Fjorgynn> Constant Rate Factor [20:48] <sesam> what distro u guys use for ffmpeg? [20:48] <Fjorgynn> sesam: Windows 7 [20:49] <Fjorgynn> =D [20:49] <sesam> =) [20:49] <sesam> and the fastest one? arch? [20:49] <quebre> Fjorgynn: my main goal is to reduce size of the file with movie, movie lenght is 02:52:07, my main goal is to convert it so it doesn't exceed 800MB of size, currently it's 4.1GB, will i achieve that if i use -crf ? [20:49] <sesam> please not gentoo though... [20:49] <viric> 'fastest'? [20:49] <Fjorgynn> quebre: yes, if you do -crf 28 or something. Probably [20:49] <viric> choose one you know how to use. That will be the fastest for you [20:50] <quebre> Fjorgynn: so, highest crf value = smaller output video size ? [20:50] <Fjorgynn> high is low quality, yes [20:50] <sesam> viric: yes the best distro for ffmpeg that doesnt take much of the resources for other things. want as much power as i can get [20:50] <quebre> ok, what's the minimal and maximal value for crf ? [20:50] <Fjorgynn> quebre: why 800 mb? Get a bigger usb stick [20:50] <quebre> Fjorgynn: it can't be higher than 800mb, peroid ;) [20:50] <Fjorgynn> quebre: standard is 23 [20:51] <quebre> larger* [20:51] <viric> sesam: for what codec? [20:51] <Fjorgynn> if it's an ipod or whatever. You can also use mono [20:51] <quebre> The range of the quantizer scale is 0-51; where 0 is lossless, 23 is default, and 51 is worst possible. [20:51] <sesam> virvic: h264 [20:51] <quebre> so here we are [20:51] <quebre> hmm [20:51] <quebre> ok im going to try the crf [20:51] <viric> sesam: x264 detects the cpu at runtime; it has code ready for any cpu. Any distro will do. [20:51] <sesam> vircic: what distro u use? [20:52] <viric> NixOS [20:52] <sesam> virvic: ffmpeg installed by default? or did u have to build it? [20:53] <viric> it's installed if you ask it to be installed [20:53] <quebre> ok another question [20:53] <sesam> viric: thx [20:53] <quebre> if high crf = low quality (and small output video file) then if i select -profile veryslow will i get even smaller output video file size ? or ? [20:54] <Fjorgynn> you'll get better bitrate accuracy [20:54] <viric> quebre: you mean preset? [20:54] <quebre> yes, preset [20:54] <quebre> typo [20:54] <quebre> Fjorgynn: and that means? :p [20:54] <Fjorgynn> I think slow works [20:55] <Fjorgynn> instead of veryslow [20:55] <quebre> veryslow is also as an option [20:56] <Fjorgynn> I know [20:56] <quebre> there is even 'placebo' [20:56] <quebre> ;D [20:56] <Fjorgynn> use placebo then [20:56] <quebre> but that's ffmpeg's devs joke :p [20:56] <quebre> and is waste of time [20:56] <quebre> ;d [20:56] <Fjorgynn> and veryslow isn't? [20:56] <quebre> no [20:57] <quebre> Why is placebo a waste of time? [20:57] <quebre> It helps at most ~1% compared to the veryslow preset at the cost of a much higher encoding time. It's diminishing returns: veryslow helps about 3% compared to the slower preset, slower helps about 5% compared to the slow preset, and slow helps about 5-10% compared to the medium preset. [20:57] <viric> can ffmpeg prepare DVDs? [20:57] <viric> dvd-video I mean [20:58] <Fjorgynn> viric: what do you mean? [20:59] <viric> something to be burnt by a dvd burn program [20:59] <viric> and played in dvd video players [21:00] <viric> with sections, maybe menus, ... [21:00] <viric> chapters [21:00] <quebre> hey [21:00] <quebre> if i have 8gb of ram [21:00] <quebre> does ffmpeg uses it during encoding ? [21:02] <Fjorgynn> I think cpu is more important [21:04] <quebre> ok last thing [21:04] <quebre> that i cannot figure out !!! [21:05] <quebre> ffmpeg -i Cloud.Atlas.2012.R.C.720p.BRRip.x264.AC3-TRiNiTY.sample.mkv -c:v libx264 -preset veryslow -crf 29 -c:a libfdk_aac -b:a 128k -ac 2 Cloud.mkv [21:05] <quebre> 37M Cloud.Atlas.2012.R.C.720p.BRRip.x264.AC3-TRiNiTY.sample.mkv [21:05] <quebre> that's before [21:05] <quebre> and [21:05] <quebre> 8.3M Cloud.mkv [21:05] <quebre> after :p [21:05] <Fjorgynn> yeah? [21:05] <quebre> how can i know how small will 4.1GB file be after i use -crf 29 ? [21:05] <Fjorgynn> 20% less [21:06] <quebre> and.. how do you know that ? :P [21:06] <Fjorgynn> quebre: I don't know, you can't really [21:06] <Fjorgynn> quebre: math. 8,3 M / 37 M. [21:06] <quebre> so it will be also 20% less for 4.1gb file ? :P [21:06] <Fjorgynn> you can also use VBR instead of constant aac bitrate. [21:06] <quebre> what VBR changes ? [21:07] <Fjorgynn> the audio bitrate changes [21:07] <Fjorgynn> -vbr [0-5] [21:08] <quebre> erm [21:08] <quebre> i don't get it, you mean the "-b:a 128k" that i used? [21:08] <Fjorgynn> yes [21:09] <Fjorgynn> but it doesn't matter [21:09] <quebre> because ? [21:11] <Fjorgynn> orka bry [21:14] <quebre> alright, 4.1gb conversion started, with 'veryfast' and -cr 30 ;) [21:14] <Fjorgynn> veryfast? [21:15] <Fjorgynn> are you fun? [21:15] <quebre> btw 'veryfast' produces smaller filesize than 'veryslow' would you think it works like that? [21:15] <quebre> ;-) [21:15] <divVerent> that's neither bug nor feature :P [21:15] <divVerent> CRF is not comparable across presets [21:16] <divVerent> veryfast crf 30 may have lower quality than veryslow crf 30 [21:16] <quebre> hmm [21:16] <divVerent> the rule of thumb is to first find the preset you want for the performance you need [21:16] <divVerent> and then tune the crf [21:17] <divVerent> the reason is, the settings the presets change can affect both quality and size [21:17] <quebre> ah [21:17] <quebre> i see [21:17] <divVerent> so maybe veryfast contains settings that make the file bigger at same quality, but ALSO settings that reduce quality a lot and make the size a bit smaller [21:18] <quebre> but 'veryslow' from what i see is really slow, 4.1gb will take 2 or 3 days to finish.. [21:18] <divVerent> the quality-reducing options basically reduce the number of choices the encoder can use, which primarily makes it faster [21:18] <divVerent> and also quite clearly reduces quality [21:18] <divVerent> but it also can lead to less data stored [21:18] <divVerent> it all depends on what you need [21:18] <divVerent> if it's worth the 2 or 3 days, then good :P [21:19] <quebre> ok i will have to experiment a bit on my 2minutes sample before i go for 4.1gb conversion ;p [21:19] <quebre> thx ;> [21:19] <divVerent> that's how to do it [21:19] <divVerent> how important is output file size to you? [21:19] <Fjorgynn> he must have 800 mb [21:19] <divVerent> a fast preset with a lower crf may be a safe and good choice [21:19] <divVerent> oh, ok [21:20] <divVerent> I wonder if you should rather use 2pass encoding then :P [21:20] <divVerent> as that allows you to define the target file size quite well [21:22] <Fjorgynn> whatever [21:22] <Fjorgynn> AFK [21:22] <quebre> divVerent: hmm [21:23] <quebre> divVerent: well that was sensible for me as i tried 2pass before went for crf, but Fjorgynn suggested me to use crf and now he's going AFK (hahaha) [21:24] <divVerent> crf is good if you don't have an exact target [21:24] <divVerent> then it does just the right thing [21:24] <divVerent> if you want to hit 700 to 800 MB, CRF is still good [21:24] <divVerent> and probabyl faster, at the expense of a little quality from not fillng the 800 MB exactly [21:25] <Fjorgynn> the problem is that his movie is +2hrs long [21:26] <quebre> don't have to be exactly 800, can be +-200mb [21:26] <quebre> at most [21:50] <cbsrobot> quebre: finished the compilation ? [21:51] <quebre> cbsrobot: yep [21:51] <quebre> ;) [21:51] <cbsrobot> and ? [21:51] <cbsrobot> crf available ? [21:51] <quebre> and no errors, able to use crf and all stuff [21:51] <quebre> ;) [21:51] <cbsrobot> libfdk_aac compiled ? [21:51] <cbsrobot> nice [21:51] <cbsrobot> so make a small sample and give it a run [21:52] <cbsrobot> then you make the math for the whole movie [21:52] <quebre> ffmpeg -i Cloud.Atlas.2012.R.C.720p.BRRip.x264.AC3-TRiNiTY.sample.mkv -c:v libx264 -preset veryslow -crf 29 -c:a libfdk_aac -b:a 128k -ac 2 Cloud.mkv [21:52] <quebre> that's my line [21:52] <cbsrobot> sounds reasonable [21:53] <quebre> but i changed 'veryslow' to 'fast' and crf set on 30, tried on samle and from 37MB input, got 7.3MB output so it's nice score as for me [21:53] <cbsrobot> use veryslow if you have enough time [21:53] <quebre> don't have, not this time ;p [21:53] <cbsrobot> veryslow tries harder to encode the file, while fast doesn't give a shit. [21:53] <quebre> i promised to deliver the movie before midnight ;p [21:54] <viric> it seems as if noone used the preset 'normal' [21:54] <viric> I use 'normal'. is that strange? [21:54] <quebre> but i will convert the same movie with 'veryslow' when i go to bed coz i'm deadly curious to compare both outputs 'fast' and 'veryslow' [21:54] <cbsrobot> normal is default [21:54] <cbsrobot> i guess [21:54] <viric> yes [21:54] <viric> I mean.. everyone talks about either fast or slow [21:54] <viric> :) [21:54] <quebre> or placebo ;> [21:54] <viric> I feel alone running normal. [21:55] <Fjorgynn> Good night! [21:56] <cbsrobot> gn [21:59] <viric> is it important to run up to date x264 code? [21:59] <viric> I'm using x264 from one year ago [22:00] <cbsrobot> viric: there's not much developpement atm, but on the other hand - one year is a lot of time in videospeak [22:01] <viric> so hard to tell [22:01] <viric> I've checked the git log; not a lot of checkins [00:00] --- Tue Jan 15 2013

1 0

ffmpeg-devel.log.20130113
by burek 14 Jan '13

14 Jan '13

[06:38] <cone-658> ffmpeg.git 03Michael Niedermayer 07master:a6e4796fbf0a: pthread: Do not use a half updated context as master for deallocation. [12:25] <durandal_1707> it is really disapointer that nobody answers in ml [12:28] <saste> durandal_1707, which patch? [12:29] <durandal_1707> no patch, but user questions on user* ml [12:49] <durandal_1707> ATTENTION!!! [12:49] <durandal_1707> http://forum.doom9.org/showthread.php?p=1610364&highlight=ffmpeg#post1610364 [12:56] <wm4> ffmpeg helping applications having to add horrible workarounds [12:56] <wm4> because people will use it anyway, and such mismuxed files might make it into the wild [12:58] <durandal_1707> michaelni: the commit that allows muxing opus before matroska even defined how it shoul be done shoule be reverted asap [13:24] <funman> durandal_1707: how about adding EXPERIMENTAL tag like mkvtoolnix does? [13:40] <nevcairiel> but its not compatible with mkvtoolnix experimental version =p [13:42] <durandal_1707> so remove it, feel free to properly implement experimental version [13:47] <cone-153> ffmpeg.git 03Diego Biurrun 07master:4d3b144c5ea8: fate: cosmetics: Order some test entries [13:47] <cone-153> ffmpeg.git 03Michael Niedermayer 07master:841a288cb327: Merge commit '4d3b144c5ea824193019019d33740a1ae9e0bb69' [13:49] <michaelni> durandal_1707, ive mailed nicolas about the opus issue, i suspect he will look into it soon [13:50] <michaelni> but dont hesitate to disable it before nicolas has time [13:59] <cone-153> ffmpeg.git 03Diego Biurrun 07master:0f94c0b1b68a: fate: vp3: Fix fate-vp3-coeff-level64 test dependencies [13:59] <cone-153> ffmpeg.git 03Martin Storsjö 07master:abae27ed3acd: rtpdec: Fix the calculation of expected number of packets [13:59] <cone-153> ffmpeg.git 03Michael Niedermayer 07master:6d6eb7c12c83: Merge commit 'abae27ed3acd0a7c54f11760c5be2d2653c4edf8' [14:01] <durandal_1707> so dolby-e is not supported? [14:02] <kierank> no [14:04] <durandal_1707> but multimedia.cx claims it is [14:05] <cone-153> ffmpeg.git 03Martin Storsjö 07master:e568db40258d: rtpdec: Calculate and report packet reception jitter [14:05] <cone-153> ffmpeg.git 03Michael Niedermayer 07master:6cd1dbe6dffe: Merge commit 'e568db40258d549777ac1c16971678e18a18f5f5' [14:06] <kierank> it is writing about my unreleased decoder [14:08] <durandal_1707> the sample in mxf is reported as pcm [14:08] <kierank> that's the point [14:09] <kierank> the point of dolby e is to masquarade as pcm to pass through legacy equipment [14:10] <nevcairiel> does it have the same silly rules as DTS to be like, only 14-bit or so, so its not too loud when played accidentally like this? .d [14:11] <kierank> they seem to run some sort of XOR on some variants of the bitstream which might be for that reason [14:12] <cone-153> ffmpeg.git 03Martin Storsjö 07master:22c436c85e0d: rtpdec: Send a valid "delay since SR" value in the RTCP RR packets [14:12] <cone-153> ffmpeg.git 03Michael Niedermayer 07master:3bcf443f91d8: Merge remote-tracking branch 'qatar/master' [14:13] <durandal_1707> kierank: so 3 years passed and nobody interested in donating some money? [14:13] <kierank> not about money [14:13] <durandal_1707> but original issue on roundup mentioned it [14:13] <kierank> things change [14:14] <kierank> the issue is dolby will sue anyone who uses it professionally (including myself) [14:14] <JEEB> ya [14:14] <JEEB> dolby and dts -- the folk who'll never license anything (F)OSS [14:15] <durandal_1707> but we have dts encoder [14:15] <kierank> durandal_1707: doesn't mean you can use it professionally [14:15] <kierank> same with ac3 [14:15] <durandal_1707> so even decoder is no-go? [14:16] <kierank> i will release it when "the process" (which i can't elaborate on) that will allow me and others to use it professionally ends [14:17] <kierank> also fwiw there are problems in lavf/lavc with regards to demux that make dolby e integration hard without hacks [14:17] <kierank> i guess less of a problem these days because you can just put one track as a substream in an avframe [14:18] <kierank> not 100% correct mind you [14:19] <kierank> getting it to demux in mxf will be a total pita [14:25] <cone-153> ffmpeg.git 03Paul B Mahol 07master:5b9126ba845d: lavfi/gradfun: support YUV440P [14:30] <pross-au> kierank: just put your encoder in international waters. license prblm solved. [15:14] <cone-153> ffmpeg.git 03Stefano Sabatini 07master:c499d45c6bea: doc/texipod: add rule to correctly interpret @ref{ANCHOR,XREF,SECTION_NAME,...} [15:14] <cone-153> ffmpeg.git 03Stefano Sabatini 07master:89ef5868679f: doc/ffmpeg: extend documentation for the -filter options and -af/vf aliases [15:14] <cone-153> ffmpeg.git 03Stefano Sabatini 07master:78e27c44c3c2: lavf/utils: clarify/extend messages in avformat_find_stream_info() [15:14] <cone-153> ffmpeg.git 03Stefano Sabatini 07master:98247e3368fb: lavf: clarify/extend documentation for AVFormatContext::analyzeduration [15:32] <cone-153> ffmpeg.git 03Michael Niedermayer 07master:dc6588421e09: swr: move buffer used to discard sample to context [15:32] <cone-153> ffmpeg.git 03Michael Niedermayer 07master:db4e0ecaf0e9: swr: move silence buffer to context to avoid per use malloc/free [16:04] <cone-153> ffmpeg.git 03Stefano Sabatini 07master:e8bab4c6fca0: ffplay: extend documentation for the -vf option [16:04] <cone-153> ffmpeg.git 03Michael Niedermayer 07master:b481d09bd9c7: swr: limit buffer size for discarding. [16:04] <cone-153> ffmpeg.git 03Michael Niedermayer 07master:a34aee464628: swr: limit buffer size for silence injection [17:57] <rik__> I have a question [17:58] <jmbclegend> it was a dream [17:58] <rik__> in ffio_limit I see a reference to the maxsize field of an AVIOContext. In some of the demuxers, I see that value set when the header is read and sometimes it's not mentioned anywhere in the demuxer at all. [17:59] <rik__> So I'm assuming that there's some default value that it's set to. Where does that happen if it does? [18:05] <kierank> cbsrobot: ping [18:11] <michaelni> rik__, default is 0 set during allocation [18:11] <rik__> michaelni, thanks a lot [19:09] <cone-484> ffmpeg.git 03Michael Niedermayer 07master:d676598f879b: swr: fix handling of timestamps that cause multiple drops or silence injections [20:04] <cbsrobot> kierank: pong [20:05] <kierank> cbsrobot: what can decode that sample of yours [20:05] <kierank> with cid 1244 [20:05] <kierank> dnhxd-tr [20:05] <cbsrobot> quicktime can [20:06] <cbsrobot> with dnxhd codec of course [20:07] <cbsrobot> I guess it's AvidAVdnCodec.component version 2.1.0 [20:09] <kierank> i reckon it's just dnhxd with a certain bitrate/resolution [00:00] --- Mon Jan 14 2013

1 0

ffmpeg.log.20130113
by burek 14 Jan '13

14 Jan '13

[05:14] <Tronic> I haven't been able to keep track of the recent changes to PTS calculation in ffmpeg and libda. What is the best way to do that now when decoding with libavformat/codec of the FFMPEG project? [05:14] <Tronic> I want actual codec-aware timestamps. [05:15] <Tronic> Apparently the libda project is doing something different and there is hardly any documentation on this. [05:15] <Tronic> *libav [06:05] <Demon_Fox> Has anyone else noticed that sound from libvorbis sounds good until you get below 51.5 Kb/s? [06:05] <Demon_Fox> as in per channel [10:33] <jacobs1> hi, anyone here that can help with ffserver ? [10:51] <Demon_Fox> jacobs1, I might. [10:54] <jacobs1> I am trying to stream a video file via rtsp with mpeg4 video codec but I have frame rate issues. [10:54] <jacobs1> I am getting this error from ffmpeg: [10:54] <jacobs1> timebase 125000/2997003 not supported by MPEG 4 standard, the maximum admitted value for the timebase denominator is 65535. [10:54] <jacobs1> I tried adding -r 65535/2733 before the input in the ffmpeg command and it worked but I get a high frame rate and I want only 10 fps. if I set -r 8 in ffmpeg it causes AV to not be synced and the video to play in slow motion. [10:54] <jacobs1> What is the proper way to configure ffserver and ffmpeg to work with mpeg4 and stream a video file in 8 fps ? [10:55] <Demon_Fox> Which mpeg4? [10:55] <jacobs1> here are the detailes : http://ffmpeg-users.933282.n4.nabble.com/ffserver-ffmpeg-feed-mpeg4-video-f… [10:56] <Demon_Fox> As in mpeg4 part 2 or 10 [10:56] <jacobs1> I dont know I just configured ffserver with 'VideoCodec mpeg4' [10:57] <jacobs1> how can I check ? [10:57] <Demon_Fox> What container? [10:57] <jacobs1> rtp [10:58] <Demon_Fox> Let me see if I can give this a try [10:59] <Demon_Fox> I need the command you used [10:59] <Demon_Fox> or equivalent [11:00] <jacobs1> I postged here the ffserver config and the ffmpeg command: http://ffmpeg-users.933282.n4.nabble.com/ffserver-ffmpeg-feed-mpeg4-video-f… [11:01] <Demon_Fox> Don't add it before the input [11:01] <Demon_Fox> add it after [11:02] <Demon_Fox> jacobs1, http://pastebin.com/Bs3CSYTU [11:02] <Demon_Fox> That is where the command is supposed to be and it is not there. [11:04] <jacobs1> sorry try this: http://pastebin.com/S4XhDLKX [11:05] <Demon_Fox> Could you input a video for the log? [11:06] <Demon_Fox> Try something like: [11:06] <Demon_Fox> ffmpeg23 -i input website -r 8 [11:07] <Demon_Fox> though I can give that a try [11:07] <jacobs1> I will give it a try, the video file is very large but I can run ffprobe on it if it helps [11:08] <Demon_Fox> or shorten the file [11:08] <Demon_Fox> like [11:08] <Demon_Fox> dd if=filename of=output bs=1024 count=1024 [11:09] <Demon_Fox> That will give you a 1MB file [11:13] <Demon_Fox> any questions? [11:13] <jacobs1> this is the file: https://www.dropbox.com/s/qx2tp291d5d8fuj/video.avi [11:15] <jacobs1> I tried ffmpeg23 -i input website -r 8 and same timebase error [11:16] <Demon_Fox> Hold on I am going to give this a try [11:16] <Demon_Fox> Also you might want to see if it works without the server [11:19] <luc4> Hi! Is it possible to extract frames of the stream from a container placing it in a buffer I specify? [11:21] <buhman> while using -target ntsc-dvd; I get crazy obnoxious amounts of error messages like: http://sprunge.us/RGda [11:21] <Keyboard_Warrior> buhman, thats better [11:22] <Demon_Fox> I give up [11:23] <jacobs1> doesnt work for you ? [11:49] <Demon_Fox> jacobs1, Looks like some error on the side of the programmers, and I don't want to get involved [11:49] <Demon_Fox> I mean [11:49] <Demon_Fox> I don't know how to fix it [11:52] <luc4> Hi! Do I have to allocate the buffer in a AVPacket myself or is it allocated automatically? [12:17] <jacobs1> Demon_Fox, OK thanks. I also tried another video file that works but no matter what I tried I wasnt able to change the frame rate from the files original 25 fps. when I try either video files from ffmpeg without ffserver I am able to set frame rate, maybe its somthing to do with my ffserver config ? [12:32] <Demon_Fox> jacobs1, Some containers only support certain frame rates you know [12:32] <luc4> Anyone who knows if there is a way to provide ffmpeg a buffer to fill with compressed frames? [12:32] <Demon_Fox> Not sure what you mean [12:32] <Demon_Fox> as in some sort of piping deal? [12:33] <luc4> Demon_Fox: sorry, are you referring to me? [12:33] <jacobs1> Deamon_fox, rtp doesnt support frame rates lower than 25 fps ? [12:33] <Demon_Fox> luc4, Are you trying to pipe something? [12:33] <Demon_Fox> jacobs1, I will check [12:34] <Demon_Fox> jacobs1, the container [12:34] <Demon_Fox> which I assume is mpg [12:35] <luc4> Demon_Fox: yes, I would like to get compressed frames from the stream in the container and decode using hw. But I see that AVPacket is filled with a buffer in the data member. Can I provide a buffer myself and have that buffer filled by ffmpeg? [12:35] <luc4> Demon_Fox: the buffer itself is one used by my hw decoder. [12:36] <Demon_Fox> If you would like you could encode with ffmpeg and send it to some program to hold so many frames then send them off every time it gets full [12:37] <luc4> Demon_Fox: sorry, encode? I want to decode. [12:38] <Demon_Fox> Not sure what you mean, but someone else might [12:40] <Demon_Fox> jacobs1, try 15 or 18 instead of 8 [12:40] <luc4> Demon_Fox: there is something that I might have misunderstood, when I call the av_read_frame function, passing an AVPacket. This AVPacket I see has the data member set to NULL. So, I guess that function allocates a buffer and returns me the address of that buffer in the data member. Is this correct? [13:22] <jacobs1> cd .. [13:48] <jmbclegend> it's no good i still have ffmpeg-0.10.6 [15:38] <luc4> Hi! Given a AVPacket containing h264 from a mov file, can I somehow apply a filter to convert to annexb? [15:39] <saste> luc4, check bitstream-filters in libavcodec [15:39] <beastd> hi [15:41] <luc4> saste: thanks. Can you suggest a page or a sample code so that I can read how it works? [15:42] <saste> luc4, read ffmpeg-bitstream-filters, then you need to read the implementation in libavcodec and how it is used in ffmpeg.c [15:43] <luc4> saste: thanks [16:30] <luc4> Does av_read_frame reads an entire NAL packet in AVC? [17:42] <matthavener> whats a generally well supported audio codec thats typically paired with h264? mp3? aac? [17:45] <Mavrik> aac [17:46] <Mavrik> H.264/AAC in mp4 is such a standard widely supported combo [17:54] <matthavener> thanks Mavrik [20:30] <bdrung> hi, i like to port audacity to support newer ffmpeg version. URLContext is gone. what's the replacement for it? is there documentation explaining how to migrate code from deprecated functions to the new ones? [20:42] <viric> Hello ffmpeg! [20:42] <viric> I came for wisdom [20:43] <viric> I've to get some video from a PAL capture card [20:43] <viric> I want to encode it in something reasonable, for high quality, I don't mind much the size, but that could be encoded realtime [20:43] <viric> any suggestion? [20:43] <viric> - for the codec and parameters [20:47] <JEEB> viric, ffvhuff or utvideo [20:48] <JEEB> both are lossless and rather quick because they are based on huffman coding [20:48] <viric> ok [20:51] <viric> maybe I can't afford so much data though [20:51] <viric> :) [20:52] <JEEB> the rest depends on how much CPU power you can throw at it [20:53] <JEEB> libx264 can handle lossless too, threads well and has a wide variety of presets on the speed vs compression line [20:54] <JEEB> -preset ultrafast -crf 0 / -qp 0 should make it encode lossless 4:2:0, and if you set the output csp you can make it encode 4:2:2 or whatever you get from the capture card [20:54] <JEEB> and then by making the preset higher you can try and compress it more if the speed is fast enough [20:54] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset [20:54] <JEEB> ^ list of internal libx264 presets [20:55] <viric> hm I can't handle that ultrafast. that gives 16mbps [20:55] <viric> utvideo gives ~30mbps [20:56] <JEEB> make sure you had crf 0 or qp 0 set too [20:56] <viric> sure I hda [20:56] <viric> had [20:56] <JEEB> k [20:56] <JEEB> and libx264 tells you it has set itself to high 4:4:4 predictive profile [20:56] <JEEB> in the log [20:56] <viric> exactly [20:56] <viric> [libx264 @ 0x1ad4520] profile High 4:4:4 Predictive, level 3.0, 4:2:0 8-bit [20:56] <klaxa> i though-- yeah i thought x264 only supported lossless with high444p [20:56] <JEEB> yup [20:56] <JEEB> then all's good [20:57] <JEEB> you can set the output csp to your input csp if it's not 4:2:0 [20:57] <JEEB> and that should give a more even comparison ground [20:57] <viric> and audio, flac? [20:57] <JEEB> I'd just dump raw pcm [20:57] <JEEB> because it won't cost much [20:59] <viric> ok I'll try [21:00] <JEEB> and as I said, make sure it's actually encoding the same csp as the input is if possible :) You might gain something from it when processing it down the line [21:01] <JEEB> because I think libx264 defaults to 4:2:0 for compatibility reasons [21:02] <viric> ok I got another computer, that can do 5x mine [21:02] <viric> :) [21:02] <viric> (tested the ultrafast) [21:03] <viric> what is csp? [21:03] <JEEB> colorspace [21:03] <viric> it's YUYV [21:03] <viric> 422 [21:03] <JEEB> ok [21:03] <JEEB> now see what libx264 is encoding [21:03] <JEEB> probably 4:2:0 [21:03] <viric> with ultrafast? [21:04] <JEEB> well, it should default to 4:2:0 in all cases :P [21:04] <JEEB> just making sure [21:04] <viric> I fail to understand any report about what does it encode [21:04] <JEEB> just pastebin the terminal log the :D [21:04] <JEEB> and link it there [21:04] <JEEB> *here [21:04] <viric> thank you [21:05] <viric> damn driver. Sometimes it doesn't want to allocate buffers [21:06] <viric> [video4linux2,v4l2 @ 0x1e01400] ioctl(VIDIOC_QBUF): Cannot allocate memory [21:06] <viric> /dev/video1: Cannot allocate memory [21:06] <viric> [226870.437339] unable to allocate 185088 bytes for transfer buffer 0 [21:07] <JEEB> anyways, if the input is 422 [21:08] <JEEB> -pix_fmt yuv422p [21:08] <JEEB> should do it [21:08] <JEEB> and then libx264 should encode 4:2:2 [21:09] <viric> http://sprunge.us/FULU [21:09] <JEEB> uh-oh [21:09] <viric> ? [21:09] <JEEB> you're not setting an audio codec, right? [21:09] <viric> is this lossless? [21:10] <viric> I'm setting, yes [21:10] <viric> sorry, just testing [21:10] <JEEB> so you want it to use libvorbis? [21:10] <JEEB> then yes, it's correct [21:10] <viric> ok [21:10] <JEEB> and yes, I can see by [libx264 @ 0x8b2d2e0] profile High 4:4:4 Predictive, level 3.0, 4:2:2 8-bit [21:10] <JEEB> that you should be encoding lossless 4:2:2 :) [21:10] <viric> perfect! [21:10] <JEEB> and by the output part of Stream #0:0: Video: h264, yuv422p, 720x576, q=-1--1, 1k tbn, 25 tbc [21:10] <JEEB> so yup [21:10] <viric> aha [21:10] <JEEB> this should do [21:11] <viric> Then I've to learn to encode this to some reasonable-quality xvid [21:11] <viric> (for a specific player) [21:11] <viric> but first let's capture. [21:12] <viric> hmmm 15mbps, 120min, how much is it in matroska? [21:12] <JEEB> uhh [21:12] <JEEB> matroska itself should have a very small overhead [21:12] <viric> ok [21:12] <JEEB> so just calculate by the bit rate and length [21:12] <viric> 15/8*60*120? [21:13] <viric> in MB [21:14] <JEEB> more or less, I guess [21:14] <viric> ok [21:14] <viric> can I tell ffmpeg to downmix to one audio channel? [21:14] <JEEB> -ac 1 [21:15] <viric> The track is mono, but I've only one cable (one channel) to capture, and the card can only capture stereo. [21:15] <viric> One channel will be muted, the only not. [21:15] <viric> the other [21:15] <JEEB> there was a way to control channels better [21:15] <viric> ah [21:15] <JEEB> but you'll have to check the documentation for that on ffmpeg.org :) [21:16] <viric> I'll go for -ac1 [21:24] <viric> JEEB: utvideo takes less kbps [21:24] <viric> than x264 lossless [21:24] <viric> well, somehow the 'bitrate' increases and increases... [21:24] <JEEB> make sure both use the same csp [21:24] <viric> Stream #0:0: Video: utvideo (ULY2 / 0x32594C55), yuv422p, 720x576, q=2-31, 200 kb/s, 1k tbn, 25 tbc [21:24] <JEEB> and yes, naturally it will increase and decrease depending on the content [21:25] <viric> then it's ~50Mbps now. [21:25] <JEEB> x264 is generally better, so you might want to set a slower preset :P [21:25] <viric> that's 43GB [21:25] <JEEB> and yes, you should only see how big it gets after it has encoded some already [21:59] <viric> will ffmpeg try to keep av syncronous, even if the 'drop' counter increases? [21:59] <viric> (it's capturing realtime from video + audio, two cards apart) [22:00] <JEEB> it will try, and matroska has timestamps in the container so in theory you should be able to [22:00] <JEEB> but [22:00] <JEEB> I'd just use something that could actually handle it all in real time :P [22:00] <viric> well, it's 42% of cpu [22:00] <viric> I don't know why ffmpeg drops [22:02] <viric> nice -20 [22:21] <viric> what does 'drop' mean, currently? [22:21] <viric> That it could not make one fit into the stream for whatever reason, and duplicated one frame? [22:21] <viric> one = one frame [22:23] <JEEB> no duplication, duplication is separate [22:24] <JEEB> drop is just that a frame was dropped because it couldn't be encoded quickly enough [22:25] <viric> umh weird [22:25] <Mavrik> or you ordered less frames per second than input has [22:25] <viric> how would that happen, at 40% of cpu usage? [22:26] <viric> umh maybe come from the capture device more frames than 25fps? no idea [22:36] <Mavrik> viric, just how many dropped frames are you seeing? [22:37] <Mavrik> a few? [22:37] <Mavrik> 10 every second? [22:37] <Mavrik> having alot of dropped frames is also a warning sign when trying to do something dumb, like encoding interlaced video to progressive without deinterlacing [22:45] <viric> 120 in one hour [22:45] <viric> I use "-filter:v yadif" before "-vcodec" [22:45] <viric> I hope that deinterlaces [22:46] <Mavrik> yes, it does [22:46] <Mavrik> 120 dropped frames in 1 hour isn't alot really [22:46] <viric> no? ok [22:46] <Mavrik> it shouldn't happen... [22:46] <viric> ah [22:46] <Mavrik> but it's not catastrophic [22:46] <Mavrik> :) [22:47] <viric> and what may be causing it? [22:47] <Mavrik> probably your cam creates a broken video [22:47] <viric> I don't know how dumb the v4l2 device is [22:47] <JEEB> try removing the yadif? [22:47] <JEEB> and deinterlacing after the fact [22:47] <Mavrik> or the v4l2 driver feeds in too many frames [22:47] <viric> JEEB: yes, going lossless, sounds right [22:47] <Mavrik> JEEB, btw, ffmpeg doesn't drop frames when it can't keep up, it just slows down :\ [22:48] <JEEB> that's not what I see in general >_> [22:48] <JEEB> in general it just drops them [22:48] <JEEB> at least that's what I've seen so far [22:48] <Mavrik> if it does they're dropped by the input source [22:48] <Mavrik> and not added to the drop count of the encoder [22:48] <Mavrik> unless the code changed after 1.1 :\ [22:49] <viric> there is 1.1? [22:49] <JEEB> I've seen ffmpeg drop frames it can't keep up with for quite a while [22:49] <viric> oh I'm on 1.0 [22:49] <JEEB> at least looking at the #ffmpeg experiences I've had so far [23:02] <viric> btw [23:02] <viric> mh nothing [23:53] <viric> don't 120min VHS tapes last 120min? grmbl [00:00] --- Mon Jan 14 2013

1 0

ffmpeg-devel.log.20130112
by burek 13 Jan '13

13 Jan '13

[01:00] <saste> ubitux: any news with the kerndeint benchmark? [01:21] <llogan> saste: what about "Create a simple filtergraph"? [01:25] <saste> llogan, what is a "simple filtergraph"? [01:26] <saste> "simple" is misleading, since what characterizes it is just 1 in and 1 out [01:27] <llogan> i tohught it would help users know that there is a different between simple and complex [01:27] <llogan> many users seem to think -vf is used with any number of inputs [01:28] <llogan> which is often possible, but not really recommended, IMO [01:28] <llogan> it was just something for you to consider, not something that i feel strongly about. [02:34] <cone-463> ffmpeg.git 03André Pankratz 07release/1.1:3dab6e542941: lavfi/yadif: fix shorthand/option mismatch [02:52] <Nano> is it ok to use the FFmpeg logo in my app, to visualize what its based on ? [02:56] <Compn> is your app open source ? [02:56] <Compn> nano [02:56] <kierank> llogan: afterburner is default [02:56] <Compn> nano : where would you want to use our logo ? in the about box ? does your app use ffmpeg ? [02:56] <Compn> or on your webpage ? [02:57] <Compn> i'm not sure who gets the answer your question Nano . maybe you will have to wait for michaelni to answer . [02:58] <beastd> Nano: i think 2 things should be clear when looking at the logo in your app: 1) your app is not ffmpeg 2) your app is not advertised by ffmpeg [02:58] <Nano> on the mainpage of my app, to give ffmpeg devs credit for their work. [02:59] <Nano> sure! [03:00] <llogan> in #ffmpeg i recommend contacting herve [03:01] <beastd> also we should consider providing some versions of the logo that state "powered by" or "based on" or similar versions [03:02] <llogan> we also need to get the logo under a proper license. it is currently ambiguous. [03:02] <beastd> +1 dislike current license [03:02] <llogan> kierank: ah, thanks. someone else added that section, but i'll update it. [03:03] <beastd> but i wanted to contact herve and others who did art for ffmpeg and ask if they provide their work under some share-freely license with some restrictions [03:03] <llogan> yeah, i meant to contact him about that once i become less trademark and license ignorant. [03:03] <llogan> give me 5 years. [03:04] <beastd> i hope i will beat you to it [03:04] <llogan> me too [03:05] Action: llogan has been summoned to have some beers [03:07] <llogan> Nano: might want to request adding your project to the ffmpeg projects page... [03:11] <Nano> yes, when it's done [03:19] <beastd> Maybe this would be a good license suggestion for ffmpeg artwork: http://creativecommons.org/licenses/by-nc-nd/3.0/ [03:39] <mfg> does Stefano hang aroundin here? [03:45] <beastd> mfg: sometimes. what are you working on? [03:46] <mfg> I fixed up the stuff he recommended I fix for HTTP cookies. I just had a question about one part I didn't think made sense [03:48] <mfg> also, he said send the http stuff as a separate patch, so I'm worried we'll loose some of the context of the email chain. [03:48] <mfg> but anyway, I'll just resubmit tomorrow with a note on the bit I didn't agree with. [03:48] <mfg> ttyl [03:49] <beastd> huh? that was quick... [05:09] <cone-463> ffmpeg.git 03Michael Niedermayer 07master:cc548ea7a603: vc1dec: ensure cbpcy_vlc has been set before decoding a frame. [05:09] <cone-463> ffmpeg.git 03Michael Niedermayer 07master:d9226b3717fd: mpegvideo: dont leave stale pointers in next/last picture [10:56] <cone-384> ffmpeg.git 03Stefano Sabatini 07master:255ec768da6f: lavf/http: fix/extend option descriptions [10:56] <cone-384> ffmpeg.git 03Stefano Sabatini 07master:0a7cd740438b: doc/protocols: document http protocol options [10:56] <cone-384> ffmpeg.git 03Stefano Sabatini 07master:cab85051c0f0: doc/ffserver: rework introducing paragraphs of the "description" chapter [10:56] <cone-384> ffmpeg.git 03Stefano Sabatini 07master:df018207f90b: doc/ffserver: remove paragraph in the introductory blurb [10:56] <cone-384> ffmpeg.git 03Stefano Sabatini 07master:0edb44028284: doc/ffserver: remove painfully outdated "What do I need?" section [11:03] <cone-384> ffmpeg.git 03Stefano Sabatini 07master:4a0d1b2159a8: doc/protocols: improve wording of a sentence in http docs [11:43] <durandal_1707> michaelni: so you do not plant to fix ff_find_pix_fmt? [12:04] Action: durandal_1707 just found how trac is crap [12:16] <durandal_1707> for some reason when i run fate it loops forever for g726 test [12:45] <durandal_1707> there should be check if two codecs use same class [12:46] <durandal_1707> this caues infinite loop [12:51] <durandal_1707> is g7231 entry in riff for 0x42 really correct ? [12:52] <durandal_1707> didnt it come much longer after all lower numbers got filled up [12:52] <nevcairiel> #define WAVE_FORMAT_MSG723 0x0042 /* Microsoft Corporation */ [12:52] <nevcairiel> microsoft says it is [12:52] <michaelni> durandal_1707, ff_find_pix_fmt ? you mean that its used in lavf ? [12:53] <durandal_1707> nevcairiel: g723.1 != g723 [12:53] <durandal_1707> michaelni: exactly [12:53] <durandal_1707> used in utils [12:53] <durandal_1707> and recently in frm demuxer [12:55] <durandal_1707> git blame is useless [12:55] <durandal_1707> because every 5 seconds some idiot reindent code [12:56] <nevcairiel> so use git blame -w? [12:57] <durandal_1707> how could that help? [12:57] <nevcairiel> it ignores whitespace [12:57] <nevcairiel> indent = whitespace [12:57] <durandal_1707> heh but it cant handle prefixes [12:57] <durandal_1707> like stupid renames [12:58] <nevcairiel> well cant have everthing [13:00] <wm4> gitk usually is rather helpful by jumping to the surrounded context of the change, and you can select what line next to blame [13:03] <durandal_1707> i prefer cli, so implement that in ncurses [13:13] <cone-384> ffmpeg.git 03Martin Storsjö 07master:58b59718813b: rtp: Cosmetic cleanup [13:13] <cone-384> ffmpeg.git 03Martin Storsjö 07master:c44784c9bb9d: rtp: Rename a static variable to normal naming conventions [13:13] <cone-384> ffmpeg.git 03Michael Niedermayer 07master:dda2d2974863: Merge commit 'c44784c9bb9d0ddf5d39d0dfa640816a57b8f457' [13:18] <cone-384> ffmpeg.git 03Martin Storsjö 07master:3900d53fb1b9: rtsp: Remove references to weirdly named variables in other files [13:18] <cone-384> ffmpeg.git 03Martin Storsjö 07master:54cb096ee455: rtsp: Remove an outdated comment [13:18] <cone-384> ffmpeg.git 03Michael Niedermayer 07master:e730c3a2cbc4: Merge commit '54cb096ee4558b3bfc28c2fcd6418ce82dc39fe1' [13:22] <durandal_1707> michaelni: how adpcm with various bps can be recognized in nut? [13:34] <michaelni> durandal_1707, codec_tag or extradata [13:34] <michaelni> or data of packets itself [13:35] <durandal_1707> afaik, it is not done currently that way [13:38] <cone-384> ffmpeg.git 03Martin Storsjö 07master:f6804c3e1ba5: rtpdec: Remove a useless todo comment [13:38] <cone-384> ffmpeg.git 03Luca Barbato 07master:f61272f0efd8: ratecontrol: K&R cosmetic formatting [13:38] <cone-384> ffmpeg.git 03Michael Niedermayer 07master:15daa8f9ddaf: Merge commit 'f61272f0efd80da437570aad2c40e00f9d3f4fe6' [14:01] <cone-384> ffmpeg.git 03Rémi Denis-Courmont 07master:169fb94f0f65: pixfmt: add picture format for VDPAU [14:01] <cone-384> ffmpeg.git 03Diego Biurrun 07master:f89466ad6fd6: Add version bump and APIchanges entry for Add AV_PIX_FMT_VDPAU. [14:01] <cone-384> ffmpeg.git 03Michael Niedermayer 07master:dae382b5b28d: Merge remote-tracking branch 'qatar/master' [14:05] <durandal_1707> hmm but there is already bunch of vdpau pix fmts [14:05] <wm4> durandal_1707: they're adding hwaccel support [14:06] <wm4> which doesn't need pixel formats to distinguish the decoder [14:06] <wm4> michaelni: is there any reason why some decoders decode flipped, instead of flipping the output image? [14:07] <michaelni> wm4, which decoders? [14:08] <durandal_1707> wm4: vdpau pix fmts already use hwaccel [14:08] <wm4> durandal_1707: no, not the framework [14:08] <wm4> michaelni: good question, I get it with Film_200_zygo_pro.mov, which is h263 [14:09] <durandal_1707> that is container related flag [14:09] <wm4> looks like it [14:09] <durandal_1707> so they are adding hwaccel support without hwaccel framework? [14:10] <wm4> durandal_1707: no... they're making it so that the normal decoder can use it, just like with dxva/vaapi/vla [14:10] <wm4> *vda [14:12] <durandal_1707> isn't someting similar aready done in ffmpeg? [14:12] <wm4> no [14:12] <wm4> you need a special vdpau specific decoder, which works nothing like the codecs that use the proper hwaccel architecture [14:17] <cone-384> ffmpeg.git 03Michael Niedermayer 07master:43d6ac53f262: lavc: ff_find_pix_fmt ->avpriv [14:17] <cone-384> ffmpeg.git 03Michael Niedermayer 07master:db1ba2213f70: lavf: use avpriv_find_pix_fmt instead of ff_ [15:42] <cone-384> ffmpeg.git 03Paul B Mahol 07master:868ac91c8dbe: frmdec: do not abuse ff_codec_get_id() [16:31] <kierank> "If it can not be reproduced with ffmpeg it is not ffmpeg bug." [16:31] <kierank> that's not true at all [16:31] <kierank> the bug i reported with yadif could not be reproduced with ffmpeg [16:31] <kierank> but it was still a bug that got fixed [16:32] <durandal_1707> kierank: yes, that statement was not meant to considered absolute truth [16:32] <durandal_1707> *to be ... [17:26] <saste> michaelni, I mean, what's the name of the binary? [17:27] <saste> "asan" or whatever, so i don't have to google for it [17:31] <cone-384> ffmpeg.git 03Matthieu Bouron 07master:f9eed5d7e620: lavfi/aevalsrc: try to honor specified duration [17:37] <michaelni> saste, its clang [17:37] <michaelni> here at least [17:39] <saste> michaelni, ok please fix it accordingly [17:41] <michaelni> this will only confuse readers [17:43] <michaelni> valgrind and asan differ [17:45] <michaelni> address sanitizer adds checks at compile time [17:46] <michaelni> valgrind does it at runtime [17:46] <michaelni> so anything can be run under valgrind but for address sanitizer you need to build it with a clang that supports that stuff [17:48] <cone-384> ffmpeg.git 03Michael Niedermayer 07master:b5d9e5d06cb6: swr: minor simplification for the noise shaping pos update [17:48] <cone-384> ffmpeg.git 03Michael Niedermayer 07master:68ff7d265f17: swr: use local variable for ns_errors [17:48] <cone-384> ffmpeg.git 03Michael Niedermayer 07master:ef7fdc8cff2c: swr: use a local variable for ns_coeffs [17:51] <saste> michaelni, whatever, just fix the typos and commit [17:52] <michaelni> ok [17:52] <wm4> saste: gradfun has a nice horizontal line through the middle of the video with nv12 [17:53] <saste> wm4, possible, do you want to fix it? ;-) [17:54] <wm4> saste: no, I concluded that gradfun simply doesn't work with nv12 [17:54] <wm4> but maybe I'm wrong [17:54] <nevcairiel> does it claim to? [17:54] <saste> wm4: odd height? [17:56] <wm4> saste: no [17:56] <wm4> nevcairiel: yes [17:57] <wm4> saste: unrelated, can I ask how format negotiation within lavfi works? [17:57] <saste> wm4, no [17:57] <saste> that's trade secret [17:57] <nevcairiel> the real answer is that noone knows [17:57] <wm4> how does lavfi know if a filter, if it accepts a certain output format on the input pads, what formats the output pads will have? [17:58] <wm4> query_formats seems to set format lists per pad, and doesn't involve any interaction between them [17:58] <saste> wm4, libavfilter/filtfmts-test filter args [17:59] <saste> the exact algo is quite complicated, and i can't give you a short explanation [18:00] <saste> especially considering that i have to run now [18:00] <pengvado> wm4: yeah, that's wrong. gradfun supports anything planar, not NV12. [18:00] <saste> but if you want to force a format, the usual trick is: format=X,filter,format=X [18:00] <wm4> saste: lavfi will insert scale filters, won't it [18:01] <saste> wm4: only if necessary [18:01] <wm4> does it print anywhere if it does? [18:01] <saste> wm4, in the log (but even that could be improved) [18:02] <saste> gotta go now, bye [18:08] <wm4> there are funny things [18:09] <wm4> like execution a given section of code for 3 times with no reason given, or large sections of commented code [18:09] <wm4> *executing [18:10] <wm4> hm maybe the filter list references have some magic meaning [18:10] <wm4> and setting two pads to the same filter list means they always use the same format? [18:11] <wm4> as opposed to merely supporting the same formats [18:12] <michaelni> wm4 yes [18:12] <michaelni> same lists means they must have the same format [18:13] <wm4> michaelni: ah... well that's certainly surprising [18:14] <wm4> I guess this also means that lavfi doesn't support arbitrary lists of pairs of input/output formats, or is there a trick? [18:15] <michaelni> the goal of the format negotiation design was that it can be solved (near) optimally for arbitrary filter graphs in polynomial time [18:15] <michaelni> graphs with arbitrary lists of pairs cant be solved in polynomial time so they are not supported currently [18:16] <wm4> k [18:16] <michaelni> they could be supported in small graphs optimally or non optimally with lots of scale/convert in a large graph [18:19] <durandal_1707> lol, people pick random project that use ffmpeg and compiles it with latest ffmpeg and complains it crash [18:20] <wm4> because ffmpeg changes its api all the time for no fucking reason [18:20] <wm4> (almost) [18:20] <Compn> yes, api changes are crazy [18:20] <Compn> mostly just to look like they are doing lots of work and to mess up the code history [18:20] <wm4> wat [18:21] <Compn> wat [18:21] <Compn> you dont consider renaming functions to be changing api ? [18:21] <durandal_1707> michaelni: how to normalize floats? RI just cast them to short [18:22] <wm4> anyway, basically it goes like this: ffmpeg devs make a mess, people see themselves forced to use the mess, ffmpeg devs try to clean up the mess and remove the old functions for no reason, people try to keep up, ffmpeg devs realize the new mess isn't so great either, etc. (you can replace ffmpeg with Libav, too) [18:22] <wm4> which probably happens in all projects, but if your API is public, you got to hold up [18:23] <wm4> where's the ffmpeg version of Linus Torvalds who insults people on christmas if they break ABI? [18:23] <wm4> /rant [18:28] <kierank> well it's more of a case that if libav api/abi changes are merged then ffmpeg needs to bump accordingly [18:30] <michaelni> durandal_1707, normalize ? like av_float2int() ? [18:32] <michaelni> kierank, libav did not bump for the wma audio sample format change [18:33] <kierank> wma alone wasn't an api change, though? [18:33] <michaelni> no [18:33] <kierank> because you could query the sample_fmt [18:34] <michaelni> yes, but i think apps dont [18:34] <michaelni> otherwise they shouldnzt crash [18:34] <michaelni> but either work or say "sample fmt not supported" [18:35] <nevcairiel> a decoder giving you another sample format is not a API change imho [18:35] <nevcairiel> adding a new sample format is [18:36] <Compn> hmm [18:36] <Compn> good question [18:36] <michaelni> the problem is neither adding nor changing its adding AND changing to a new one [18:36] <Compn> a decoder returns different data than what an app is used to [18:36] Action: Compn afk [18:36] <nevcairiel> well sure, if you add a new sample format and its not used at all, it wont cause trouble :) [18:37] <durandal_1707> michaelni: well point is to still outputs floats, but currently output is too loud/clipped if I do not divide some number(s) with say 16384 [18:37] <nevcairiel> but someone using avcodec and assuming that one variable field is always the same value is rather risky [18:37] <michaelni> if its just used in new decoders it also shouldnt be a problem, the new decoders just would not be supported by old apps [18:38] <michaelni> durandal_1707, is there some spec saying something about how to scale / clip ? [18:57] <durandal_1707> ubitux: you still did not commit patch that speedup kerdeint ? [19:03] <durandal_1707> michaelni: i will need to look... [19:26] <cone-384> ffmpeg.git 03Michael Niedermayer 07master:1c14c3412ea1: dec/developer: Add Valgrind / Address Sanitizer to the patch checklist [19:26] <cone-384> ffmpeg.git 03Michael Niedermayer 07master:93bc0f018039: swr/noise_shaping_data: pad coeffs to multiple of 4 when they are 1 below [19:26] <cone-384> ffmpeg.git 03Michael Niedermayer 07master:c8737d348be5: swr: work with 4 noise shaping coeffs at a time [19:52] <cha0s_> is anyone available to help diagnose and possibly correct an ffmpeg/winff issue? [20:11] <cha0s_> cananyone help me find out whats going onwithmy ffmpeg/winff? [20:27] <durandal11707> what next i should do? [20:28] <cbsrobot> durandal_1707: what about a codec ? [20:29] <durandal_1707> what one? [20:29] <cbsrobot> f.ex dxv codec: http://resolume.com/download/ [20:29] <cbsrobot> lower right corner [20:30] <cbsrobot> it hase some debug informations in it [20:31] <durandal_1707> does it work without gpu? [20:31] <cbsrobot> I guess not [20:33] <durandal_1707> i think it does from QT - where it may be slow because no gpu is used [20:34] <durandal_1707> how much is it popular? [20:35] <cbsrobot> not popular at all [20:35] <cbsrobot> just fun [20:35] <cbsrobot> it seems it work even without gpu [20:36] <cbsrobot> "The video will play in any other software but it will not be hardware accelerated." [20:42] <durandal_1707> cbsrobot: but that codec is their selling point, if i RE they will loose job [20:43] <cbsrobot> hmm I don't think so [20:43] <cbsrobot> they don't sell the codec - it's for free [20:43] <cbsrobot> but you can only use it in qt [20:44] <durandal_1707> yes but opening it will lead to open hw acceleration [20:44] <durandal_1707> that is main reason why it is still closed [20:45] <cbsrobot> well I think they use it in their software and it seems to be the only one that can use that codec for hwaccel [20:47] <cbsrobot> maybe they will also winn job [20:47] <cbsrobot> because you can use ffmpeg to generate such files [21:07] <saste> wm4: how can I reproduce the gradfun issue? [21:11] <wm4> saste: use an extreme strength value [21:45] <cone-384> ffmpeg.git 03Michael Niedermayer 07master:754dd7e88955: dvdsubdec: use unsigned shifts to avoid shifting into the sign bit [21:45] <cone-384> ffmpeg.git 03Michael Niedermayer 07master:2ea3f37d5f97: dvdsubenc: use unsigned shifts to avoid shifting into the sign bit [21:57] <barque> can you compile ffmpeg with libstagefright for Android 2.3.1 using the NDK? [21:57] <barque> or just even Android 2.3 (e.g. 2.3.3) [22:01] <barque> anyone? [22:17] <cbsrobot> barque: see http://fate.ffmpeg.org/ [22:20] <barque> are there compiled binaries for those tests as well? :P [22:22] <cbsrobot> no I don't think so [22:22] <cbsrobot> or search google for it [22:22] <cbsrobot> not sure what project uses libstagefright [22:22] <cbsrobot> maybe https://github.com/appunite/AndroidFFmpeg [22:26] <barque> There's also FFmpeg4Android [22:44] <barque> thanks for the info anyway [22:54] <Daemon404> nevcairiel, how do you work around the broken-ass eval stuff in fate on windows [22:54] <Daemon404> scenedetect always fails cause of it [22:55] <nevcairiel> relative path to fate samples [22:55] <Daemon404> i see [22:55] <Daemon404> -_- [23:38] <Compn> wonder if subitux is alive [23:46] <cbsrobot> hehe [00:00] --- Sun Jan 13 2013

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ffmpeg.log.20130112
by burek 13 Jan '13

13 Jan '13

[00:11] <Trashlord> now it's built again, this time with libfdk_aac [00:15] <Trashlord> now it works [00:17] <Trashlord> thanks for your help, llogan [00:20] <Trashlord> now it says this for several files: [mp4 @ 0x25c1e40] track 0: could not find tag, codec not currently supported in container [00:21] <Trashlord> it won't work for libvorbis either [00:25] <Trashlord> right [00:25] <Trashlord> turns out, because I want the output to be in mp4, I must convert the video as well, using libx264 [00:26] <Trashlord> so I'll leave it as AVI [00:31] <Trashlord> well, thanks for all the help guys. got my shit sorted out [00:37] <Navion> Anybody here a Serviio wizard? [00:44] <llogan> fluid tripod heads suck when it is below freezing [02:51] <Nano> is it ok to use the FFmpeg logo in my app, to visualize what its based on ? [02:52] <llogan> Nano: you will have to ask herve.flores [at] free.fr [02:53] <Nano> ok, why ? [02:55] <beastd> llogan: depends probably on which logo he wants to use [02:56] <llogan> yes, i assumed the current one on the site [02:57] <beastd> llogan: the current one is the winter edition by muxson [02:57] <llogan> ah...i mean the standard one [02:57] <Nano> i'd like to use the zigzag logo [02:59] <llogan> Nano: contact herve. the usage/license/whatever was never really finalized properly [02:59] <Nano> ok [03:00] <Nano> just thought that theres already a license type defined for the logo, like the CC, GPL or something... [03:00] <Nano> but i will contact herve to be save before publishing my app [03:01] <llogan> herve never put it under an standard license [03:38] <savskii> hi [03:38] <savskii> http://pastie.org/5671976 [03:54] <beastd> savskii: [ac3 @ 0x93a66e0] invalid bit rate [03:56] <beastd> savskii: please note that the program you are using is *not* ffmpeg. it is just a program named ffmpeg from a fork. naturally we do not support it here. if you want to use ffmpeg look here http://ffmpeg.org/download.html#LinuxBuilds [03:58] <savskii> ok thanks mate [03:59] <savskii> is there any tutrial i can follow? [03:59] <beastd> savskii: you should find lots of stuff [04:00] <savskii> cheers [04:00] <beastd> savskii: you need to be clear on what you want to achieve. then you go on and choose proper multimedia codecs and container format [05:51] <savskii> hell can anyone help me send a live stream please [05:51] <savskii> ffmpeg -i Sample.mp4 -f rtsp -rtsp_transport tcp rtsp://85.17.169.209:3051/live.sdp [05:52] <savskii> tryed that but no luck [06:17] <barque> what is libstagefright? [06:24] <barque> hello? [06:33] <beastd> barque: IIRC it is hw decode acceleration from android platform. [06:33] <barque> Hmm, so it kinda interfaces with the hidden OpenMAX? [06:34] <barque> I mean, I've downloaded FFMpeg4Android [06:34] <barque> haven't compiled it yet [06:34] <barque> but it should provide HW accel on my android device? [06:34] <beastd> maybe [06:35] <beastd> at least hw accel on adroid device should be possible (given android can provide it for your device) [06:36] <barque> I need to decode videos in JNI [06:36] <barque> cause I need to support android 2.3.1 [06:36] <barque> and need the decoded stuff as textures [06:36] <barque> I can't use mediaplayer, it does not provide byte arrays and such, and SurfaceTexture/external opengl texture stuff was added in API level 16 [06:37] <barque> MediaMetadataretriever only does thumbnails (decodes only 1 frame for every second) [06:37] <barque> so I'm pretty much left with doing the decoding in JNI [06:38] <barque> which pretty much only leaves ffmpeg [06:38] <barque> and libavcodec [06:38] <sware> does anyone here have a generated config.h from a recent windows build [06:39] <beastd> sware: why can't you generate it yourself? [06:39] <barque> possibly cause he doesn't have cygwin [06:39] <sware> not sure the best way [06:40] <barque> automake etc. [06:40] <barque> autoconfig [06:40] <sware> I just need it to see if any changes were made in the latest snapshot [06:41] <beastd> barque: not sure if i can help you. but using ffmpeg libs on android should be possible. there were also some mails on ffmpeg-devel from some people making a dedicated project for using ffmpeg libs on android. maybe looking at their code may help you [06:41] <beastd> barque: there is no autotools involved in ffmpeg [06:41] <sware> beastd: I don't need to build, just need a config [06:42] <beastd> sware: i still don't understand. also there are at least 4 different ways to build for windows [06:42] <sware> beastd: it's for msvc version [06:42] <beastd> sware: the config.h is an internal file needed for the build and never exported directly. so i can't follow. what are you trying to accomplish? [06:43] <beastd> sware: do you have a problem with your current build? [06:43] <sware> beastd: about every 6 months i port ffmpeg to msvc, wanted to get a recent config.h to check for changes [06:44] <beastd> sware: port? it doesn't work out of the box for you? [06:44] <barque> beastd, yeah I'm currently using ffmpeg on android [06:44] <barque> actual ffmpeg.so [06:44] <barque> with swscaler.a [06:44] <barque> it's just that, it's software decoding [06:44] <barque> and it's rather slow-ish [06:44] <barque> it can decode theora the fastest [06:44] <barque> but it's very lackluster [06:44] <beastd> barque: do you have a build with libstagefright enabled? [06:44] <barque> 480p video at 21 FPS [06:44] <barque> beastd, sadly no [06:44] <barque> or I don't know actually [06:44] <barque> but what do I need to check/ensure? [06:45] <sware> beastd: I make it so ffmpeg can be built with visual studio [06:45] <barque> unfortunately my tablet is also Tegra2 [06:45] <barque> which lacks NEON simd [06:45] <barque> which makes it all the more... ARRRRGH-ish [06:46] <beastd> sware: should work out of the box since a few months. please try it [06:46] <sware> beastd: what do you mean ? [06:46] <barque> thinking back, you gotta love every iPad/iPod platform, all come with NEON SIMD on the A4-5 chips [06:46] <barque> so beastd, does libstagefright normally compile along with ffmpeg? [06:46] <barque> or do I need to do something special? [06:46] <beastd> sware: http://ffmpeg.org/platform.html#Microsoft-Visual-C_002b_002b [06:47] <beastd> barque: did you cross-compile ffmpeg yourself for android? [06:47] <barque> I downloaded one available on the net [06:47] <barque> I can't find the link [06:47] <barque> the guy had it all compiled already [06:47] <barque> I tried compiling myself... I could not (for the love of me) get the configure script right [06:47] <barque> nor could I use anyone else's [06:48] <sware> ya I don't care about that tool heh [06:48] <barque> something was up with my cygwin ndk compiler [06:48] <barque> g++ [06:48] <beastd> barque: huh. ok :( [06:48] <beastd> sware: you roll your own? with lots of code modifications? [06:49] <sware> yes [06:50] <barque> beastd, if libstagefright.cpp was included in libavcodec/ folder... chances are, I should have it compiled along with what was provided? [06:50] <beastd> sware: ok. so you also know how to adress config.h to suit your needs and just need current one so you can work from that? [06:50] <barque> if that is the case, this should be wicked sick [06:50] <sware> beastd: yes [06:52] <beastd> barque: you can try to find the decoder with a call to libavcodec [06:53] <sware> beastd: the last build I did was june 18th. I'm assuming something have changed in the config since [06:54] <beastd> barque: these are the other people that posted they work on ffmpeg for android: http://www.insignal.co.kr/wiki/index.php/FFmpeg_on_Android (I do not know anything about there project, and it will maybe not be helpful to your problem.) [06:54] <beastd> sware: for sure. lots of things [06:58] <sware> oh well, guess I'll have to figure it out [06:58] <sware> I have never been able to build on a windows system heh [06:58] <sware> with mingw or anything else [06:59] <beastd> sware: wait [06:59] <sware> other than my ports, the last time purplepeter provided me one [07:00] <barque> wait, when was OpenMAX and stagefright exposed as an API? [07:00] <barque> Android 4.0? [07:02] <beastd> sware: maybe this helps http://pastebin.com/Y5Vr0qKi [07:02] <beastd> sware: i generated it on mingw with --disable-yasm [07:03] <beastd> barque: i have no special knowledge about android [07:03] <sware> thanks beastd. I will cross reference that thanks [07:10] <beastd> barque: you can e.g. test sth like this "codec = avcodec_find_decoder(AV_CODEC_ID_H264);" then look into codec->name if it is not "libstagefright_h264" then you are not using libstagefright [11:25] <Trashlord> hey. just wanted to say I ended up ditching the AVI container. the sound was all choppy on my TV, unwatchable. for the past 9 and a half hours I've been converting all the videos to mp4 container format, with H264 video and AAC audio, that one works [11:25] <Trashlord> it's gonna take at least another 10 - 12 hours to convert the rest of the files [12:28] <elkng> is there difference of having option before or after option "-i file" ? [12:32] <JEEB> elkng, yes [12:32] <JEEB> before -i you set input/decoding settings [12:32] <JEEB> after -i you set output/encoding settings [12:34] <elkng> is that information in man page ? [12:35] <elkng> its confusing, is there any other console software where order of options is matter ? [12:35] <JEEB> I think there are, but don't ask me for an example right now :P generally you'd then have -input-options and -output-option separately [12:36] <JEEB> and yes, it's in the manpage/documentation as far as i know [12:36] <JEEB> because I think the possible manpage and friends are made of the same source as http://ffmpeg.org/ffmpeg.html#Synopsis [12:36] <elkng> and how to recognize what are options depend on order ? like option -ss and others [12:36] <JEEB> everything that you set before -i is gonna be done before decoding [12:37] <JEEB> if it is a compressed stream [12:37] <JEEB> for example -ss and -t set before -i will try to cut the stream so you can copy it over [12:37] <JEEB> at GOP/frame lengths [12:37] <elkng> and what that command could mean: "-ss 10:0 -f file.mp4 -ss 20:0 output.avi" ? [12:38] <Fjorgynn> no [12:38] <Fjorgynn> isn't it -t output? [12:38] <Fjorgynn> aha [12:39] <JEEB> elkng, that seems invalid to me [12:39] <JEEB> -f mp4 is valid, -f file.mp4 is not valid because -f is format [12:40] <JEEB> and that most probably would be /after/ -i [12:40] <JEEB> it would seek to whatever you mean with 10:0, set output format to mp4, set ss again (!?) and then write mp4 container stuff into a file output.avi [12:50] <elkng> JEEB: I mean "-i" not "-f" [12:51] <elkng> the combination "-ss 10:0 -i file.mp4 -ss 20:0" [12:51] <elkng> or "-ss 20:0 -i file.mp4 -ss 10:0" [12:52] <JEEB> so it would first seek as well as possible to whatever you mean with 20:0 by the audio frames and video GOPs, then it would start decoding and seeking to the point that you note as 20:0 [12:53] <elkng> its tricky [12:53] <JEEB> as long as you know that coded audio consists of audio frames and that coded video has GOPs [12:54] <JEEB> you will expect that if you have both video and audio, that before decoding starts seek won't be exact [12:54] <JEEB> with just audio, the audio frames are small enough that you usually get close enough [12:54] <elkng> more convinient was to have something like these options: "-i file.mp4 --start-position 0:20:0 --end-position 0:30:0" [12:54] <JEEB> of course, lossless formats like FLAC or wavpack might disagree [12:55] <JEEB> uhhh [12:55] <JEEB> you didn't do that [12:55] <JEEB> and if you want that you just put -ss and -t /after/ -i [12:55] <JEEB> -ss is seek, and -t is how long you want to keep decoding in timestamps [12:56] <elkng> JEEB: "/after/ -i" that way instead of seeking in file it will decode input video until that mark and it will be much longer [12:56] <JEEB> naturally [12:57] <JEEB> but do remember that unless you actually checked where the GOPs start in the file [12:57] <JEEB> you can't use before -i either too easily :P [13:01] <Fjorgynn> what does -absf aac_adtstoasc do? [13:02] <Abhijit> hey guys [13:02] <JEEB> audio bit stream filter that converts ADTS AAC to ASC [13:02] <Abhijit> is libavg.de down for for everyone or just me? [13:02] <Fjorgynn> JEEB: which means what+ [13:02] <JEEB> Fjorgynn, there are two types of handling AAC inside things? [13:02] <JEEB> just like there's that mp4 type H.264 and Annex B H.264 [13:02] <Fjorgynn> okay [13:03] <JEEB> I don't think those changes should always be visible to the user tho, at times you just have to add the bit stream filter manually when using ffmpeg :P [13:04] <Fjorgynn> JEEB: a service I use to "save videos" has that command [13:10] <Fjorgynn> [mp4 @ 04B95620] malformated aac bitstream, use -absf aac_adtstoasc [13:10] <Fjorgynn> av_interleaved_write_frame(): Operation not permitted [13:10] <Fjorgynn> that's why [13:15] <JEEB> yes, so mp4 container wants ASC [13:16] <JEEB> and your input is ADTS [13:16] <JEEB> don't ask me why ffmpeg doesn't do this automagically :D [13:44] <elkng> thats an interesting question why ffmpeg doesn't do this automagically [13:45] <Mavrik_> iirc there's no common logic to turn on required bitstream filters [13:46] <Mavrik> so muxer dies :) [13:48] <JEEB> and it isn't really an interesting task so it's mostly a case of "watches pelcome" [13:53] <Mavrik> indeed [14:06] <Abhijit> hey guys. the freecal game is just disappeared from the internet. i can not find any reference anywhere? am i missing something? [14:06] <Abhijit> sorry wrong channnel [14:55] <Fjorgynn> added vlc to path <3 [16:45] <espr3ss0> hi [16:45] Action: espr3ss0 waves [16:45] <espr3ss0> anyone know how to set / fix, "stream 0, timescale not set" ? [16:46] <espr3ss0> is there a "timescale" flag ? [16:46] <espr3ss0> although it might not be the problem :-/ [16:47] <espr3ss0> I have 2 mp4's and they both give the "timescale not set" problem using ffmpeg, but one can't skip (when played) further the about 10 mins [16:48] <espr3ss0> i'm counting on ffmpeg coz it should be able to do it ! ;-) [16:51] <JEEB> can you compile L-SMASH and see what boxdumper --timestamp that_file.mp4 > timestamps.txt gives? [16:51] <JEEB> http://code.google.com/p/l-smash/ [16:51] <JEEB> l-smash [16:51] <espr3ss0> i could do ! [16:51] <JEEB> l-smash should be simple to compile, ./configure and then make [16:52] <espr3ss0> after that using ffmpeg how would one fix it ? o_O [16:52] <JEEB> uhh [16:52] <JEEB> boxdumper is to get info on how the file is created [16:53] <JEEB> if you would do boxdumper file.mp4 > all_box_data.txt you would get a file full of the whole file's structure [16:53] <JEEB> in a more readable format [16:53] <JEEB> if boxdumper can find a timescale for the file, and can output timestamps on that timescale, then ffmpeg has a problem somewhere [16:53] <JEEB> if boxdumper can't find it either [16:53] <espr3ss0> hmm, might have fixed it via an mkv version [16:53] <JEEB> then we have a problem [16:54] <JEEB> well, the problem is still interesting :P [16:55] <espr3ss0> :-) [16:57] <JEEB> because if there really is no timescale then there's really no way of finding the timestamps for that track [16:57] <JEEB> because timestamps are X/Y [16:57] <JEEB> where Y is timescale [16:57] <JEEB> timescale = ticks for one second [16:58] <espr3ss0> thanks for the heads-up [16:58] <JEEB> how does the boxdumper --timestamp output look? [16:58] <JEEB> it should output timestamps for all tracks in the file [16:59] <espr3ss0> i will get round to installing it ! ;-p have stuff to do :-( [17:00] <JEEB> no need to install it really [17:00] <JEEB> but yeah, compile it when you have time :) [17:00] <JEEB> it's a nice mp4-style container implementation [19:49] <cha0s_> is anyone available for a little help in troubleshooting and possiblly correcting an ffmpeg/winff issue? [20:43] <skifreak> hey all, having some trouble with a bluray file, specifically decoding with ffmpeg while playing. When I ripped the file via passthrough I didn't receive any errors. I think my problem due to the block code look-up failure, just trying to figure out where to start http://dpaste.com/868741/ current ffmpeg version 1.0.1 [21:32] <CStanners> Does anyone know what "Marker bit missing in complexity estimation part 1" is about? [22:11] <elkng> I noticed many videos of films of good quality and in size of less than 1GB but when I try to achieve that size I end up with quiet bad quality, what options those videso could have been encoded with ? [22:14] <lectrick> Anyone know a strategy to avoid "rb buffer underflow" and general buffer underflow errors when transcoding a 720p MKV file to a dvd vob? [22:20] <lectrick> The other thing is, I am trying a multipass strategy in order to perhaps help mitigate the underflow issue. The first pass creates the pass log file but writes 0 bytes to it when it's done :O [22:20] <cbsrobot> lectrick: what version of ffmpeg are you using ? [22:20] <cbsrobot> there was a bug from sept. till nov. 2012 &. [22:21] <cbsrobot> env. [22:21] <cbsrobot> elkng: depends on what codec, size etc. you want to use [22:21] <cbsrobot> it's kinda hard to help you with so little informations [22:24] <elkng> cbsrobot: it should depend on codec and not on some tricky ffmpeg's options ? [22:26] <lectrick> cbsrobot: ffmpeg version 1.0 Copyright (c) 2000-2012 the FFmpeg developers [22:26] <cbsrobot> lectrick: try to upgrade [22:26] <cbsrobot> September, 28, 2012, FFmpeg 1.0 [22:27] <lectrick> cbsrobot: I am using a managed tool maintainer (homebrew on OS X in this case), and someone else is responsible for updating the repo with later ffmpeg versions. I could build it manually I suppose but this is probably a pain [22:27] <cbsrobot> lectrick: you have xtools installed ? [22:27] <lectrick> you know what? I haven't updated homebrew in a while, let me see if it has updated repos [22:28] <cbsrobot> sounds rasonable [22:28] <lectrick> oh shit, it's going to update to 1.1 [22:28] <cbsrobot> else: http://ffmpeg.org/trac/ffmpeg/wiki/MacOSXCompilationGuide [22:28] <lectrick> well that's a prerequisite I can do lol [22:28] <lectrick> stand by. [22:29] <cbsrobot> elkng: well it depends on a lot of things ... [22:29] <elkng> cbsrobot: wheather, pressure ? [22:30] <cbsrobot> elkng: sure use -weather and -pressure options to set them correctly [22:31] <elkng> it was pun [22:31] <cbsrobot> just make sure to play them in similar conditions or the files will have bad quality & :-) [23:08] <lectrick> cbsrobot: after an extensive set of updates and ultimately upgrading ffmpeg from 1.0 to 1.1, and retrying the same command, the buffer underflow errors seem to be gone :O [23:09] <lectrick> i guess my last question is, how do I create a dvd ISO given a .vob file? :) [23:09] <sacarasc> Use something like dvdauthor. [23:10] <lectrick> yup, a/v plays fine in my 7 minute test output sample :) [23:12] <lectrick> sacarasc: i have dvdauthor, i will look into how to use it, but if anyone knew the command which just took a .vob file and spit out a menuless .iso that would be wonderful haha [23:19] <cbsrobot> I think you'll need mkisofs or similar to create the iso [23:25] <lectrick> yeah i'm installing that now too... fun afternoon figuring out how to do all this lol [00:00] --- Sun Jan 13 2013

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