[Ffmpeg-devel-irc] ffmpeg.log.20111214

[burek]

[00:11] <Shimmy> amv news?
[00:11] <Shimmy> adpcm_ima_amv news?
[00:32] <Shimmy> JEEB?
[02:23] <nooj> hi. can ffmpeg conver ac3 5.1 to aac 5.1? if so, how?
[02:25] <nooj> would something like -acodec libfaac -ac 6 -ar 48000 -ab 640k work?
[02:46] <burek> nooj, you can always use -vf aconvert or -vf aformat
[02:49] <burek> oops sorry :) didn't read very well
[02:49] <burek> -ac 6 would do the job ^^
[02:52] <burek> hehe, FOX aired rallies in Athens and told the viewers it was Moscow :)) http://rt.com/news/fox-moscow-fake-riots-281/
[04:09] <beastwick> hello, how can I find out what file my line-in for ALSA is? as in -f *** -i /dev/???
[04:14] <beastwick> oh I used cat /proc/asound/pcm
[04:14] <beastwick> because its now in /dev/sda format
[04:14] <beastwick> *not
[04:14] <beastwick> so how do I know which is line-in?
[04:14] <beastwick> ex) hw1,0 or hw0,1 ?
[04:47] <beastwick> could anyone help me with setting up my ffmpeg script for audio?
[04:52] <beastwick> 178 people and no help :(
[04:59] <Freakshow> what's up beastwick?
[04:59] <beastwick> ah hey freakshow
[04:59] <Freakshow> I'm on a conf call... but perhaps paste the command you're using (pastebin if it's too large) and I'll see if I can help
[04:59] <beastwick> sure
[04:59] <beastwick> hang on
[05:02] <beastwick> http://pastebin.com/tTRzV9qw
[05:02] <beastwick> and I explain my problem there
[05:04] <beastwick> i hope those alsa settings will grab sound from a microphone
[05:05] <beastwick> i will send u my cat /prc
[05:05] <beastwick> hang on
[05:07] <beastwick> http://pastebin.com/U6nEfBGH that is cat /proc/asound/pcm
[05:07] <beastwick> I am trying to locate the one for the LINE-IN mic as well
[05:09] <beastwick> iguess based on that its either hw0,0 or h0,2
[05:22] <beastwick> freak did you get all that?
[06:08] <serf_> i have these lectures in mp3 but whenever i play them they cause my speakers to click constantly even when i pause the player, what could i do to fix it?
[06:13] <blarson> When streaming an mp3 with my custom app or ffplay, av_read_frame() returns an error after a few minutes.  However, it works fine streaming with Chrome or playing the downloaded mp3 in ffplay.  Any thoughts or suggestions?
[08:07] <booi> hey guys, when i'm encoding some prores 422 with ffmpeg, I can encode 720x480 videos just fine but 1920x1080 videos don't play on Macs. is there something I need to know about encoding 1080p videos in prores?
[08:10] <hael> HELLO, this is my issue
[08:10] <hael> ffmpeg-0.9#  ./configure --enable-libx264
[08:10] <hael> libx264 is gpl and --enable-gpl is not specified.
[08:11] <hael> logfile http://pastebin.com/ipP4izX3
[08:14] <hael> hael 08:10:06
[08:14] <hael> HELLO, this is my issue
[08:14] <hael> ffmpeg-0.9#  ./configure --enable-libx264
[08:14] <hael> libx264 is gpl and --enable-gpl is not specified.
[08:14] <hael> 08:10:07
[08:14] <hael> logfile http://pastebin.com/ipP4izX3
[08:25] <kcm1700> hael: did you try "./configure --enable-gpl --enable-libx264" ?
[08:30] <hael> ok thanks it works
[08:42] <booi> okay i figured it out. seems like there was a bug because the same command on 0.9 fixes 1080p encoding in prores
[09:06] <tremby> How do I convert only a particular frame range of the source to the destination? (Ideally the corresponding audio range would be automatically selected, otherwise I can figure that out)
[09:41] <tremby> figured it out -- need to divide by framerate and use -ss and -t
[12:01] <burek> serf_, blarson, could you two please create a bug report and provide some samples?
[12:26] <sb1066> hello! I'm trying to transcode a dvb stream into webm for http streaming. The sound doesn't sound as it should, all I hear is cracks. Could someone take a look at my code?
[12:26] <sb1066> https://github.com/john-tornblom/tvheadend/blob/master/src/plumbing/transcode.c
[12:39] <serf_> huh burek ?
[13:15] <Abbadon> Hi, I'm trying to encode a video from a local file and send the output to justin.tv. ffmpeg also connects with the server, also starts to encode the file, but the fps drops during the encoding to about 9 fps. Of course with 9 fps it is not possible to watch a smooth video. CPU usage is at about 10%, upload about 400 kbit/s of 2.5 Mbit/s. Here is the command I use to encode the file: http://pastebin.com/A2vZk1Fx When
[14:02] <wmmd> Hi there, I would like to convert video files to MP3 and I am using this call: ffmpeg -i file.mp3 -acodec libmp3lame -qscale 4 -ab 320k -vn
[14:02] <wmmd> Now my question: If I were to convert the file to another video format, would the qscale effect the video quality too?
[14:03] <MT`> Abbadon: either pre-encode the file before using ffmpeg to send it, or tell x264 to encode it quicker (lower quality), or buy a faster computer
[14:03] <wmmd> Or should I use a different way to create a VBR MP3?
[14:04] <Abbadon> Hi, I have a Intel i5 2500k @ 4,5 GHz
[14:04] <Abbadon> and cpu usage is @ about 10%
[14:04] <Abbadon> so where is the hint that my cpu is to slow?
[14:04] <MT`> Abbadon: it takes too long
[14:05] <MT`> do you have HTT enabled? (8 cores or 4 cores)
[14:05] <Abbadon> 4 cors
[14:05] <sacarasc> Abbadon: Try adding -threads 0 to the output thingies.
[14:05] <Abbadon> cores
[14:05] <Mavrik> wmmd, no, ffmpeg is order sensitive when it comes to parameters
[14:05] <wmmd> Mavrik: What does that mean?
[14:05] <MT`> Mavrik: you sure about that? 
[14:06] <Mavrik> MT`, very sure :)
[14:06] <wmmd> Basically, what I want to do is: Convert video files to MP3 and preserve the best possible audio quality without bloating the files too much.
[14:06] <Mavrik> wmmd, meaning, for video you'd do -vcodec libx264 -qscale <whatever> -acodec libmp3lame -qscale 4
[14:06] <Mavrik> wmmd, and the first qscale would be for video, second for audio
[14:06] <wmmd> Mavrik: So I dont have to use -ab in addition to that?
[14:06] <sacarasc> wmmd: -acodec copy
[14:07] <Mavrik> um, don't know, that's libmp3lame dependent
[14:07] <MT`> Mavrik: if you do -c:v libx264 -profile main -c:a libfaac, ffmpeg will attempt to apply the main profile to libfaac (and fail)
[14:07] <wmmd> sacarasc: Nope, I need a specific target format.
[14:07] <Mavrik> MT`, file a bug (or update), it shouldnt.
[14:07] <MT`> using master
[14:08] <MT`> i'll file a bug tonight
[14:08] <Mavrik> wmmd, what do you mean "convert video to mp3"? discard the video?
[14:08] <Abbadon> -threads 0 doesn't help...
[14:08] <Mavrik> Abbadon, you have to put it in the -vcodec section
[14:08] <wmmd> Mavrik: Exactly. Thats why Im using -vn.
[14:08] <Mavrik> MT`, for most parameters ffmpeg is order sensitive (see -threads, -b etc.)
[14:09] <wmmd> Mavrik: But when I use -acodec libmp3lame -qscale 4, the output file will get much smaller than the original aac stream I get with -acodec copy.
[14:09] <wmmd> And VLC says the mp3 file has a 64kb/s bitrate.
[14:10] <Mavrik> wmmd, effect of -qscale is encoder dependant and I don't know how libmp3lame acts, so I can't really help you there
[14:10] <Mavrik> usually -qscale and -b parameters aren't compatible though
[14:10] <Mavrik> but as I said: it's encoder dependent
[14:10] <wmmd> Mhm. I have read that "The -b bitrate switch has no effect with -qscale. It will default to 200kbit and show 200kbit, but actual bitrates will be adjusted to match the q value!" (http://gasubasu.com/2010/02/12/troy%E2%80%99s-ffmpeg-recipe-book/)
[14:11] <wmmd> Is there any encoder-dependent documentation?
[14:11] <Mavrik> huh... dunno :\
[14:11] <Abbadon> I did it like this: http://pastebin.com/w1xHdxVV
[14:12] <Mavrik> wmmd, it seems for audio you should use either -aq or -q:a instead of -qscale
[14:12] <Mavrik> wmmd, for VBR
[14:13] <Mavrik> Abbadon, try saving the output to a file, just to make sure it's not the network that's the bottleneck
[14:13] <wmmd> Ah, Ill try that. I thought, -aq would only work with vorbis.
[14:14] <Abbadon> I wrote it to a file, there is no problem... But I also tried different broadcasting service, the same problem...
[14:19] <Mavrik> then you're obviously having a bandwidth problem
[14:19] <Abbadon> sometimes, ffmpeg.exe crashes...
[14:22] <Abbadon> hmm speedtest sais: 2.5 Mbit/s available
[14:33] <MT`> Abbadon: what fps do you get transcoding to a local file? What fps do you get using ffmpeg to send said local file to your network over rtmp? smth like ffmpeg -f flv -i input.flv -vc copy -ac copy "rtmp://...."
[14:34] <Abbadon> http://pastebin.com/w1xHdxVV 
[14:34] <Abbadon> my settings
[14:34] <MT`> Abbadon: thats great, its not what I asked :/
[14:35] <Abbadon> ahh ok, now i understand^^
[14:36] <Abbadon> mom, brb
[14:36] <Abbadon> quit
[14:37] <Bove> I'm getting what seems to be 709/601 related color shifts when converting DPX to Prores. Any tips on how to avoid this_
[14:43] <Abbadon> can't send that file via rtmp without encoding it
[14:44] <MT`> Abbadon: yes, so encode it first, and then send the encode file
[14:44] <MT`> I want you to encode it to local disk, so we can see how fast ffmpeg can encode it
[14:45] <MT`> I then want you to send the date using ffmpeg to the remote site over rtmp, so we can see how fast you can upload it
[14:45] <MT`> ffmpeg -f flv -i input.flv -vc copy -ac copy "rtmp://...." will not re-encode the data, so will be super fast, assuming your internet connection can handle that upload
[14:46] <Abbadon> ok, used the settings mentioned already 
[14:47] <Abbadon> min. 64 fps
[14:48] <Abbadon> hmm, really seems to be a bandwidth issue...
[14:49] <wmmd> Mavrik: Ah, now I know why I didnt use -aq with libmp3lame. It makes ffmpeg crash and abort quite often.
[14:50] <MT`> wmmd: -aq just means -qscale:a
[14:51] <MT`> wmmd: and -q just means -qscale
[14:51] <MT`> they're all equivalent, just -aq/-q:a/-qscale:a solely apply to audio codec, where as -q is applied to all codecs
[16:10] <bf4648> I'm trying to use php to execute an ffmpeg command & I can't get it to work...any ideas why?
[16:11] <Mavrik> nope.
[16:12] <Tjoppen> not if you don't at least describe what's wrong
[16:12] <bf4648> Well, I'm using php & exec ( 'ffmpeg -i  etc.') but nothing is happening?  
[16:12] <bf4648> www-data has the files on apache server
[16:13] <Tjoppen> well, pastebin some logs or something
[16:13] <Tjoppen> do you have sufficient permissions?
[16:14] <bf4648> www-data has the files & should be able to execute the command 
[16:19] <multiscan> Hi all. I know shouldn't enter a channel and imediately ask a question but I've been trying for the last two hours without success.... 
[16:20] <bf4648> yes you should multiscan...I do it all the time
[16:20] <multiscan> I am trying to join two m4v files (h264 video + aac stereo audio) with -i concat but I always get only the first file copied in the output 
[16:21] <multiscan> ffmpeg -i concat:"./00005.m4v|./00007.m4v" -acodec copy -vcodec copy aaa.m4v
[16:21] <multiscan> or various variant all similar to that
[16:21] <Tjoppen> that only works for formats designed for that, like mpegts
[16:21] <multiscan> ah ok. that's explains. but for ts it works also just cat
[16:22] <multiscan> so I have to remux the two files into ts and then concat them
[16:22] <relaxed> multiscan: you can probably concat them with MP4Box or mkvmerge
[16:23] <mystica555_> i second mkvmerge as being a good way
[16:23] <relaxed> why ffmpeg still can't do this I don't know.
[16:24] <multiscan> ok. I'll give mkvmerge a try. and eventually come back to you... :)
[16:42] <multiscan> wow! great mkvmerge worked super easy.
[16:42] <multiscan> thank you very much for the advice
[17:24] <Shimmy> Anyone please read this: https://ffmpeg.org/trac/ffmpeg/ticket/747
[17:29] <sacarasc> Shimmy: Is AMV also a container?
[17:29] <JEEBsv> it's basically AVI with some values in header derped
[17:29] <JEEBsv> http://wiki.multimedia.cx/index.php?title=AMV
[17:45] <jermy> How stable is an h264 decode expected to be when setting thread_count on the context?
[17:45] <jermy> We've got some content that works fine with a single thread, but fails to decode video with two or more
[17:46] <jermy> Or are we missing something - are we meant to handle a threaded codec context in a different way?
[17:46] <Ginks> any idea why -vcodec copy tells me I don't have the codec
[17:46] <jermy> (H264 error is 'Invalid mix of idr and non-idr slices')
[17:46] <Ginks> figured it would have been in one of the codec packages
[17:59] <Shimmy> What's the term 'derp' meant for?
[18:00] <JEEBsv> a very broadly used internet term for not something positive
[18:00] <Shimmy> Thanks
[18:01] <Shimmy> Now if u are a good C programmer please contact me on private
[18:10] <Shimmy> Anyone please help me out with this issue im desperate: https://ffmpeg.org/trac/ffmpeg/ticket/747
[18:11] <Shimmy> It's related to amv - adpcm_ima_amv formats
[18:13] <JEEBsv> If you want a diff of the adpcm.c in that GPL AMV ffmpeg thingy against the revision of ffmpeg it was originally made against
[18:13] <JEEBsv> I'll give you that when I get home
[18:13] <JEEBsv> but it's pretty derpy as far as I had Kovensky look at it
[18:23] <pasteeater> Zeranoe: did you get a chance to look at the frei0r issue?
[18:30] <Shimmy> @JEEBsv, of course I prefer it should be LGPL, but since my software is gonna freeware, GPL is OK
[18:31] <pasteeater> Ginks: more info needed
[18:31] <pasteeater> use a pastebin site to show your ffmpeg command and the complete console output
[18:51] <dericed> Given the shoutouts to open tickets. I opened http://ffmpeg.org/trac/ffmpeg/ticket/737. The new timecode feature of drawtext generates the wrong value when a dropframe timecode is provided.
[18:58] <dericed> re #1: i didn't get to it (just did the 4 that kate prioritized)
[18:58] <dericed> but I have a better model for adding these
[18:59] <dericed> re: #2. format is about container and codec is about codec
[19:00] <dericed> physical format is just a statement about the object, like 'betacam'.
[19:00] <dericed> 3: hmm ok, i can send the xsl. just worried that they would district because of all the bogus data
[19:01] <dericed> haha, wrong text window, so sorry to send uncontextual data
[19:13] <Zeranoe> pasteeater: I've talked to the guy, he brings up a valid point, is frei0r supposed to show up under filters?
[19:14] <pasteeater> Zeranoe: it does in linux
[19:14] <pasteeater> as frei0r and frei0r_src
[19:15] <Zeranoe> it doesn't seem to be for windows, even though it says it is compiled with it and doesn't fail with any errors
[19:16] <Zeranoe> ill look into it later 
[19:16] <pasteeater> weird
[19:23] <dericed> i think this is the same on a mac, that enable-frei0r will complete without error, but then it doesn't show as available
[19:58] <Zerodamage> I have an issue with capture from a v4l device. When i use ffmpeg it starts capture but just quits instantly with no errors. Any surgestions?
[19:59] <pasteeater> Zerodamage: are you using an old ffmpeg?
[20:00] <Zerodamage> I'm using the newest version available for Debian lenny on ARM
[20:00] <pasteeater> use a pastebin site to show your ffmpeg command and the complete console output (if any)
[20:04] <h_senior_paris> Hi could I get some help on rtmp protocol to red5 ?
[20:19] <pasteeater> h_senior_paris: try ffmpeg-user mailing list if you don't get help here.
[20:20] <h_senior_paris> ok thanks but as far the problem is not related to the command line they do not respond
[20:21] <h_senior_paris> i throw which here I could get help on real interface problem betwenn tools
[20:25] <gearaholic> If i need to convert a video to h264, webm and ogg (for html5 video)  would it be best to convert to h264 then use that file as the input for ogg/webm or encoder each from the original?
[20:25] <pasteeater> each from original
[20:26] <pasteeater> otherwise: input > lossy output > lossy output
[20:26] <gearaholic> thats what i was thinking
[20:26] <gearaholic> thanks
[20:26] <pasteeater> no problem
[20:47] <Zerodamage> Command and output: http://pastebin.com/GSq2dznv
[20:49] <Zerodamage> The problem may lie in the v4l part of things because the samme ffmpeg exec works fine on a different PC
[20:52] <relaxed> Zerodamage: your ffmpeg is from 2007! It's time to upgrade.
[20:53] <Zerodamage> Ok but does that advance v4l/v4l2 compatability?
[20:54] <Zerodamage> What i dont understand is why it quits with no apparent errors
[21:02] <Ginks> pasteeater, figured it out, thanks
[21:05] <Zerodamage> any verbosity flags and/or log options i can use to figure it out?
[22:07] <Shimmy> Hi! I'm pretty new to FFmpeg. I need to convert an HD video. I need it to much the parameters of this video (it's to be supported on a hardware player) providing the best quality possible.
[22:07] <Shimmy> Check it out: http://pastebin.com/gPGBBGuz
[22:11] <pasteeater> Shimmy: 1512.mpg is the file that works on your hardware player?
[22:12] <Shimmy> That is correct
[22:13] <Shimmy> I want to convert another HD file so it should have the same option, while still retaining the best quality
[22:14] <Shimmy> With 'best quality' I mean best quality possible under the new conversion
[22:15] <Shimmy> Any suggestion will be really appreciate
[22:15] <Shimmy> Please use http://pastebin.com
[22:23] <pasteeater> Shimmy: it would be useful to show info on the input too
[23:24] <bf4648> Alright, I'm executing an ffmepg command on an apache server...the person running the www-data...I've chmod -R 777 the entire folder but I still can't get ffmpeg to work...any ideas?
[23:42] <Shimmy> Any one can help me?
[23:43] <Shimmy> http://pastebin.com/gPGBBGuz
[23:43] <Shimmy> I want to convert videos to match the above video's settings while retaining the best quality possible
[00:00] --- Thu Dec 15 2011