[Ffmpeg-devel-irc] ffmpeg.log.20120106

[burek]

[00:08] <ubitux> (diff: FFmpeg 0.9 ’ FFmpeg 0.9.1)
[00:11] <iOOO9> scaling/resizing with "-vf scale=-1:704" instead of "-s 704x396" seems to be much slower
[00:11] <iOOO9> 55fps vs 21fps
[00:11] <iOOO9> is that understandable?
[00:14] <praedo> btw, why do i get this: Please use -b:a or -b:v, -b is ambiguous
[00:15] <sacarasc> Because that's the new syntax.
[00:20] <relaxed> iOOO9: because scale=-1:704 is encoding a larger frame.
[00:24] <rowan-> Is it possible to copy the video and audio streams from an AVCHD *.mts file without any alteration to the original into an M4V container?  I have not been able to figure out the commands to do this.
[00:27] <iOOO9> relaxed: am i misinterpreting the syntax? wouldn't the longest side be 704px with both?
[00:34] <rowan-> Using: ffmpeg -i input.mts -vcodec copy -acodec copy test.m4v Results in no sound.
[00:44] <maujhsn> Can anybody tell me about the re-compile process in ffmpeg for using a plugin called "download helper"?
[00:48] <relaxed> iOOO9: "-vf scale=704:-1" == "-s 704x396"  (depending on the aspect ratio)
[00:49] <maujhsn> Last year you could download videos off youtube using "realplayer's plug-in for "firefox" or "internet explorer" it seems to be much harder these days anybody have answers?
[00:49] <relaxed> rowan-: .... -f mp4 test.m4v
[00:51] <relaxed> maujhsn: why would you need to recompile ffmpeg for "download helper"?
[00:51] <rowan-> relaxed: thanks.  Still no sound.
[00:51] <relaxed> pastebin the command and all output
[00:52] <rowan-> relaxed: MediaInfo was shoing that it was an MP$ with or without the -f flag.
[00:53] <maujhsn> relaxed I do not believe everything that I read take a look at this: http://www.downloadhelper.net/conversion-manual.php#install-other 
[00:53] <rowan-> C:\Users\Rowan\Desktop\FFMPeg\bin>ffmpeg -i input.mts -vcodec copy -acodec copy
[00:53] <rowan-> -f mp4 test.m4v
[00:53] <rowan-> ffmpeg version N-36193-gf514695, Copyright (c) 2000-2011 the FFmpeg developers
[00:53] <rowan->   built on Dec 26 2011 17:55:53 with gcc 4.6.2
[00:53] <rowan->   configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-ru
[00:54] <relaxed> pastebin.com
[00:54] <rowan-> relaxed: sorry :-) Let me figure that ourt...
[00:54] <rowan-> I really thought you meant past in and had just missed the spacve key!
[00:55] <rowan-> relaxed: http://pastebin.com/embed_js.php?i=mKM3yBVN
[00:56] <iOOO9> relaxed: ah, i had turned it the wrong way :)
[00:56] <iOOO9> thanks
[00:58] <relaxed> rowan-: do you hear audio with ffplay?
[00:59] <keeperofdakeys> iOOO9: in answer to your question about parts and profiles, they are different
[00:59] <iOOO9> keeperofdakeys: okay
[00:59] <maujhsn> relaxed "never mind" I have decided to uncomplicate things 
[00:59] <keeperofdakeys> the two video parts (mpeg4 and h264) are entirely separate codecs, can can be implemented separately
[01:00] <iOOO9> keeperofdakeys: i see. then what are the profiles?
[01:00] <rowan-> relaxed: yes, but it does not actually play.  It hangs at the end, just "clicking" repeating the last maybe 1 sec of sound over and over and the video is just stutter, stutter, stutter.  Windows Media Player plays the video perfectly, but without sound.
[01:00] <keeperofdakeys> they define what technologies are used
[01:00] <iOOO9> how does ffmpeg chose container format for my output file? guessing by the file extension i give it?
[01:00] <keeperofdakeys> put simply, the higher the profile, the more cpu is required for encoding and decoding, but the better quality vs bitrate
[01:01] <keeperofdakeys> for example, mpeg4 has SP and ASP (simple profile isn't very good, due to the lack of bframes)
[01:01] <keeperofdakeys> h264 has base, mainline and high
[01:01] <relaxed> iOOO9: yes or you can force it with -f $format
[01:01] <keeperofdakeys> where minaline is decent, but you really want hihg
[01:01] <keeperofdakeys> *high
[01:02] <keeperofdakeys> for example, iOS only supports base and mainline
[01:02] <keeperofdakeys> even though it would be possible to use hardware decoding for it
[01:03] <keeperofdakeys> h264 also has levels, but they are just a more specific version of profiles
[01:04] <relaxed> rowan-: Well, ac3 is valid in mp4 so it should work. Try, ffmpeg -i input.mts -vcodec copy -acodec libvo_aacenc -ab 192k -f mp4 test.m4v
[01:06] <rowan-> relaxed: that seems to have worked.  The video is not good enough quality to see if the audio is synced correctly.  I am guessing that the audio was recompressed with this command?
[01:07] <relaxed> yes, aac in mp4 is more windows/os x friendly
[01:09] <rowan-> So I can presume that their would of been some loss to the audio stream, although unlikely to be descernible given the audio quality is fairly low anyway, but the Video has been coppied bit for bit?
[01:09] <maujhsn> relaxed in your example "ffmpeg -i input.mts -vcodec copy -acodec libvo_aacenc -ab 192k -f mp4 test.m4v" does test.m4v in the statement or can you use output.m4v?
[01:10] <rowan-> maujhsn: It's just the output filename, I could of used anything "helloworld.m4v"
[01:10] <relaxed> test.m4v is just the name of the output, you can call the output whatever you want
[01:10] <maujhsn> relaxed  "Thanks"
[01:11] <maujhsn> rowan "Thanks"!
[01:13] <relaxed> rowan-: correct
[01:13] <rowan-> Thanks Relaxed, thanks for answering my questions.  Hopefully the forum will be back online soon :-)
[01:13] <rowan-> Bye.
[01:13] <maujhsn> "Thanks guys take care"!
[01:13] <iOOO9> ah, great!
[01:25] <SteveTheCat> How do you copy subtitles in the conversion?
[01:25] <SteveTheCat> where you hardcode subtitles?
[01:34] <SteveTheCat> Anyone know?
[01:46] <keeperofdakeys> SteveTheCat: if they are hardsubs, they will be in the video stream itself
[01:46] <keeperofdakeys> if they are softsubs, -scodec copy
[01:46] <SteveTheCat> i tried that but i got an error for using -scodec copy
[01:47] <keeperofdakeys> SteveTheCat: in the output, under metadata, does it only list video and audio?
[01:47] <keeperofdakeys> if so, the subtitles are actually in the video
[01:49] <SteveTheCat> Could not write header for output file #0 (incorrect codec parameters ?)
[01:49] <SteveTheCat> and that's the video stream and i used -vcodec copy
[01:53] <metvik> is there any way for ffmpeg to extract dvb subtitles?
[01:56] <SteveTheCat> Is there a way to list subtitle codecs?
[01:57] <SteveTheCat> do i need to compile with --enable-libass?
[02:11] <SteveTheCat> i seem to be getting an error using -scodec ass: Could not write header for output file #0 (incorrect codec parameters ?)
[02:11] <SteveTheCat> this is what i've put ffmpeg -i "file.mkv" -vcodec mpeg4 -acodec libfaac -scodec ass test.avi
[02:19] <SteveTheCat> keeperofdakeys, is the paramters correct for what i've put?
[02:19] <keeperofdakeys> SteveTheCat: can you post the full output on dpaste.org?
[02:21] <SteveTheCat> keeperofdakeys: http://dpaste.com/682475/
[02:21] <SteveTheCat> wait
[02:21] <SteveTheCat> wrong paste
[02:22] <SteveTheCat> keeperofdakeys: I meant this one http://dpaste.com/682477/
[02:23] <keeperofdakeys> SteveTheCat: avi does not support subtitles
[02:23] <SteveTheCat> i want to hardcode it though
[02:23] <SteveTheCat> hardsub
[02:24] <keeperofdakeys> (theoretically it doesn't support mpeg4 or h264 either, but it has been hacked to work)
[02:24] <keeperofdakeys> SteveTheCat: I don't know how to do that with fmpeg
[02:24] <keeperofdakeys> *ffmpeg
[02:25] <keeperofdakeys> from looking around on google, it seems impossible with ffmpeg
[02:26] <SteveTheCat> keeperofdakeys: Is it possible to convert/export one audio track? So I can choose if I want english or jap?
[02:28] <keeperofdakeys> SteveTheCat: look at the -map argument
[02:29] <keeperofdakeys> you'll need to write -map aruments for all the streams you want
[02:29] <keeperofdakeys> any specific reason you want to hardcode subtitiles?
[02:55] <sam1> hi ,i use -x264opts profile=main   ,but it is error with "[libx264 @ 0x2ada500] bad option 'profile': 'main'"
[02:59] <gfto> Guys -ss before output do not work as expected, it still is accurate to closest gop not the exact frame.
[02:59] <gfto> I tried latest ffmpeg git (and libav git) and the result is the same. ffmpeg -i file.mpg -vframes 100 -f image2 -ss 00:00:00.960 file-%03d.png
[02:59] <gfto> I expect the first extracted frame to be the 24-th frame (file.mpg is 25 fps) however
[02:59] <gfto> the output howver starts from the 12th frame (mpeg gop size is 12 so it skips the first 12
[02:59] <gfto> frames and starts decoding). Afair -ss when used as output should be frame accurate, not gop accurate
[02:59] <gfto> What am I douing wrong?
[03:00] <sam1> -ss  should  before  -i 
[03:02] <gfto> no it should not be there
[03:03] <gfto> when it is before input it just tries to seek and it is even more unaccurate
[03:04] <gfto> when it is after input (used as output option) ffmpeg decodes the output up to the frame and then starts to feed frames to the encoder (or at least it should do that :)
[04:12] <keeperofdakeys> gfto: you obviously don't understand how video encodiing works
[04:12] <gfto> probably
[04:12] <keeperofdakeys> you have key frames, which are totyou have kkeyframeees
[04:12] <keeperofdakeys> theseare a complete picture
[04:13] <gfto> thanks for the lecture :) the problem was av delay
[04:13] <keeperofdakeys> you then have p frames, which recordthe changes from the key frame
[04:13] <gfto> when I compensate for av delay by adding couple of frames to -ss everything works
[04:13] <gfto> so no need to explain to be about frame types, thankyouverymuch :)
[04:13] <keeperofdakeys> ah, okay
[04:15] <gfto> mediainfo reports Delay relative to video : -504ms, so to have "frame accurate" output -ss I have to add to ss time 504 / (504 % 40) + 2 /b-frames/ = 14) and voila...
[04:19] <relaxed> gfto: did you try -itsoffset?
[04:24] <gfto> can I apply itsoffset only to one of the streams stream in the input?
[04:28] <relaxed> yeah but you'll need to use the same input twice
[04:28] <gfto> ok, found it, very awkward..will try it
[04:28] <gfto> thanks for the suggestion
[04:29] <gfto> any idea how to find audio delay only with ffmpeg without using mediainfo?
[04:29] <relaxed> I don't know. Did you try ffprobe?
[05:02] <sam1> hi
[05:50] <bigjohno> Hi, I am using ffserver to stream a live video from an IP camera sending an mp4 stream. For maximum browser compatibility experimentation has lead me to stream via a flv format.  The only issue I have is when the feed file reaches maximum size after which it seems the browsers do not know how to show a live picture and the stream stops.  I am considering stopping ffserver and deleteing the feed file periodically as a work around but am wondering if there i
[05:50] <bigjohno> s a solution to this?
[06:58] <sam1> ffmpeg   -profile baseline/main/high  doesn't work ?   
[07:01] <sam1> paste  here 
[07:01] <sam1> http://dpaste.com/682608/
[07:02] <sam1> but i use  "-profile baseline     -acodec  copy    or  -an "   it works, why?
[07:07] <sacarasc> Put -profile baseline after -vcodec libx264...
[07:09] <sam1> http://dpaste.com/682610/  
[07:09] <sam1> doesn't  work 
[07:09] <sam1> Put -profile baseline after -vcodec libx264.. 
[07:14] <relaxed> sam1: -profile:v main
[07:14] <relaxed> or baseline
[07:14] <sam1> wait a  moment 
[07:15] <sam1> it works  thanks ,relaxed 
[07:16] <sam1> but i use  -x264opt  profile=main ,it doesn't work 
[07:20] <MrJones> hello
[07:20] <MrJones> http://cekirdek.pardus.org.tr/~ismail/ffmpeg-docs/libavcodec_2utils_8c.html#ee15efa14ae54308aed60d73000662ee talks a lot about requirements for the package input buffer
[07:20] <MrJones> now I simply get the package from av_read_frame()
[07:20] <MrJones> does that function care about all those alignment, padding size and whatever stuff which is apparently required?
[09:13] <j-b-r> Ok, I hope this is the right place to ask questions like this, but: can anyone give me a quick example of how to render styled subtitles from an mkv into a video with ffmpeg?
[11:38] <lytran> hi
[11:38] <lytran> please help me about ffmpeg usage
[11:41] <lytran> How can I create 10second ts segment with ffmpeg-0.9
[12:04] <saste> lytran: try the segment muxer
[12:18] <lavluda> is ffmpeg now supper stream to rtmp ?
[12:18] <lavluda> *support
[13:07] <xj> which is better for movie audio, faac or libaacplus?
[13:22] <j-b-r> As far as I know, both are pretty bad
[13:22] <j-b-r> I might be speaking out of my depth though. ¯\(º_o)/¯
[13:23] <xj> thanks
[14:49] <Mavrik> xj, libaacplus gives way better quality for bitrate, however it allows only up to 64k bitrate
[14:49] <Mavrik> xj, so if you want to encode up to 64k, libaacplus is way better, for everything else libfaac (or libvo_aacenc) is your only option
[14:50] <cbreak> or lame... :)
[14:50] <j-b-r> ...And that's why I use neroenc.exe
[14:50] <Mavrik> cbreak, hmm, lame isn't AAC (which can cause problems), not to mention it's not significantly better
[14:51] <Mavrik> j-b-r, of course, if you want piles of additional work on windows plaform :P
[14:51] <mohanohi> Hi. can i extract only odd frames with ffmpeg. I have footage for which i need to extract 1,3,5,7,9......... frames in image format
[14:52] <mohanohi> can anybody pls tell me if it is possible to do this?
[14:53] <xj> bummer
[14:53] <Mavrik> mohanohi, use "select" video filter
[14:54] <Mavrik> xj, what kind of videos are you encoding?
[14:54] <mohanohi> Mavrik : oh .. thank you :)
[14:54] <xj> just backing up my x-files collection to my new NAS
[14:55] <Mavrik> ah
[14:55] <Mavrik> xj, libfaac, 192k is more than enough ;)
[14:55] <Mavrik> or even less
[14:56] <xj> 200kb/s is the default in ffmpeg
[14:56] <xj> thanks Mavrik
[14:56] <Mavrik> (don't use defaults in ffmpeg) ;)
[14:57] <xj> i am only using the audio default
[15:08] <raven> hi
[15:09] <raven> how to check a (raw-dv) file with ffmpeg?
[15:12] <Mavrik> check?
[15:12] <raven> yes
[15:13] <raven> the file seems broken and after two days encoding of a 7 hour raw-dv file to mpeg4 i would like to know if this would work this time
[15:35] <raven> Mavrik?
[17:35] <flan_suse> When using cat + ffmpeg, the combined media file is only as long as the first input file. E.g, cat vid1.flv vid2.flv | ffmpeg -i - -vcodec copy -acodec copy combined.flv
[17:58] <raven> is ffmpeg able to combine two files and cut them simply?
[19:32] <MeCLovin> Hey guys, I've got a quick question! I'm trying to play a mp3 file silently with ffplay. But when I use the parameter -nodisp I don't get any audio :(! What am I doing wrong?
[19:44] <MeCLovin> nobody? I tried like 20 different builds :-/
[19:58] <Mavrik> um
[19:58] <Mavrik> you want to play an mp3 file silently.
[19:59] <Mavrik> and you have a problem that you can't hear audio?
[20:00] <MeCLovin> no, i want to hide the gui
[20:03] <Mavrik> hm, ffplay 1-18\ Tie\ Your\ Mother\ Down.mp3 -nodisp works fine for me
[20:03] <Mavrik> ffplay version 0.7.2-4:0.7.2-1ubuntu1, Copyright (c) 2003-2011 the Libav developers
[20:04] <MeCLovin> that's strange, I see that everything works (the console output looks just fine), but theres no sound..
[20:05] <Mavrik> hmm
[20:05] <Mavrik> which OS?
[20:05] <MeCLovin> win 7 x64
[20:11] <iive> Mavrik: that's not ffmpeg
[20:11] <Mavrik> iive, and?
[20:13] <iive> i mean, it is not ffmpeg project, its the libav fork.
[20:13] <Mavrik> hmm
[20:13] <Mavrik> you're right
[20:13] <MeCLovin> okay sorry then
[20:13] <iive> it would be good idea to compile recent ffmpeg release.
[20:14] <Mavrik> lemme go find my 0.9 deb
[20:17] <Mavrik> bah, didn't compile ffplay :(
[21:04] <markrose> I'm having some trouble with ffmpeg. It seems to be using an old version of libav*. When I compile ffmpeg without libavformat installed when I compile, ffmpeg output shows: "libavformat  53. 24. 0 / 53. 24. 0" . When I have the package libavformat-dev (version 0.7.2-1ubuntu1) installed and I compile, ffmpeg output shows "libavformat    53. 29.100 / 53. 29.100", which is newer. I thought the ffmpeg git repository contained the latest libav stuff?
[21:18] <glassresistor> is there a way to start an mp3 at an offset from a typical http source without having to download the audio from the first part of the offset(ie not wanting to download 1h of music to listen to the last 5min)
[21:19] <glassresistor> i expect the answer for this to be no, the mp3 is from 7digitals streaming server if that matters
[21:21] <Mavrik> hmm
[21:21] <Mavrik> it should be possible
[21:22] <markrose> if the mp3 is constant bitrate it should be possible (you'd have to download the first part, figure out where the audio starts, etc) then use HTTP 1.1's If-Range request
[21:23] <markrose> If-Range -> Range header, sorry
[21:34] <glassresistor> markrose: thanks I'm trying to do this in the browser using soundmanager2
[21:35] <glassresistor> so html5 and/or flash, it has an api for setting the offset, not sure if its taking advantage of that, im looking up 7digital now but i think its vbr
[22:18] <defaultro> hey folks, I made a dvd iso of our vacation using ffmpeg and dvdauthor. File size is about 8gb.  Mplayer plays the iso fine. I started preparing imgburn but it is complaining and saying "this image has not been mastered correctly for burning onto two layer disk."
[22:18] <defaultro> what should I do?
[22:19] <microchip_> jump off a bridge :P
[22:19] <defaultro> i already did during our vacation
[22:28] <defaultro> i shrink it down to 4gb. It's just vacation so it's ok :)
[22:34] <oimon1> hi. if i have a silent video file (2 mins) and audio file (1 minute long) and would like to mux them together without transcoding, can i do it without truncating the length of the video file? i tried "fmpeg -i out.ogg -i out.ogv -acodec copy -vcodec copy output.ogv" but the first section of video is truncated
[00:00] --- Sat Jan  7 2012