[Ffmpeg-devel-irc] ffmpeg.log.20121210

[burek]

[01:46] <wannabegeekster> I am trying to capture video from the screen.  I have the audio capturing now, but there is about a 5 second delay.  SHould I use -ss 00:00:05?  Or should I use something like -async 1?
[02:50] <smj> wannabegeekster: if you're using PulseAudio and figure out how to fix the delay, please let me know
[02:52] <juanmabc> i run pulse, yet can peek directly alsa
[02:58] <smj> juanmabc: how
[03:00] <juanmabc> ffmpeg -f pulse -ac 2 -i "alsa_output.pci-0000_00_1b.0.analog-stereo.monitor" -f x11grab -r 25 -s 600x450 -i :0.0+100,100 -qmin 3 -qmax 3 out.avi
[03:01] <klaxa> that's still pulse
[03:03] <juanmabc> "pactl list sources| grep Name" to get the device name
[03:03] <juanmabc> and oh, yeah still pulse
[03:42] <wannabegeekster> So do I need to capture audio and video separately and then mux them together afterwords?
[03:52] <zap0> which words?
[03:53] <wannabegeekster> well after the capture
[04:57] <lake> all, i am having massive trouble trying to convert my analog hi8 video to uncompressed avi. audio from my device is not in the output and the video turns out to be flakey, sometimes it works well and other times its a fail, with the same settings!
[04:57] <lake> i am using the video4linux2 driver with a dazzle dvc (svideo/component red, white cables for audio)
[04:58] <lake> ffmpeg -vsync 1 -async 1 -vc 1 -standard ntsc -f v4l2 -i /dev/video0 -f alsa -i hw:2,0 -ac 2 -t 02:45:00 -c:v rawvideo -c:a copy out.avi
[04:58] <lake> the above is the command i use to capture video and audio.
[04:59] <lake> my recording device plays back just fine when I "arecord -D hw:2,0 -f cd | aplay"
[04:59] <lake> however, when i run the ffmpeg command with the settings above, and then i try to run " arecord -D hw:2,0 -f cd | aplay", nothing plays back!
[05:00] <lake> it's like ffmpeg breaks my arecord
[13:27] <drc__> Hello
[14:01] <hackeron> hey, question about the -f segment option - I want to keep the last 10 minutes of video from an IP camera and whenever there is movement, to copy the last minute or so - I'm doing ffmpeg -i url -loglevel warning -analyzeduration 0 -map 0 -codec:v copy -codec:a libfaac -ar 44100 -ab 64k -f segment -segment_time 60 -segment_wrap 10 -segment_list_flags live -segment_list_size 10 -segment_list .csv -y _%02d.mkv -- which is working for the most ...
[14:01] <hackeron> ... part, but the resulting MKV file has the total duration since the start - so when I play the file, it shows 11:35:03 of 11:36:03 - anyway not to keep this original duration that increases endlessly?
[14:03] <hackeron> I think this might be relevant: https://ffmpeg.org/trac/ffmpeg/attachment/ticket/1425/0002-lavf-segment-add-reset_timestamps-option.patch
[14:05] <saste> hackeron, yes i'm going to apply it soon
[14:05] <saste> test and reports are welcome
[14:05] <hackeron> saste: trying the patch now, thank you :)
[14:11] <drc__> Can anyone help me with some transcoding ?
[14:31] <zap0> how?     you have not stated a problem
[14:35] <_learneR> Hi, im new to ffmpeg and am trying to develop a custom media player for ARM devices. I am not able to find any documentation on a few functions. I would like to know what happens in ffurl_open() and ffurl_alloc() functions and what are all the string manipulations for in ffurl_alloc() for?
[14:38] <zap0> get the source code
[14:45] <drc__> I will like to transcode with CBR
[15:53] <hackeron> saste: patch is working nicely so far :)
[15:54] <saste> hackeron: good, a day or two and it will be in master
[16:27] <ffmpeg_user> hi, I have a problem while applying a watermark: if I do so, I don't see the watermark: ffmpeg -i http://192.168.0.4/video.mp4 -vf "movie=water.png [watermark]; [in][watermark] overlay=0:0 [out]" http://localhost:9090/feed0.ffm ... while if I do so I see the watermark: ffmpeg -i http://192.168.0.4/video.mp4 -vf "movie=water.png [watermark]; [in][watermark] overlay=0:0 [out]" nameOfoutputFile.avi
[16:29] <Tjoppen> what happens if you drop the watermark stuff?
[16:29] <Tjoppen> also, what problem are you getting?
[16:37] <saste> ffmpeg_user, https://ffmpeg.org/trac/ffmpeg/ticket/1970
[16:44] <ffmpeg_user> Tjoppen: if I drop the watermark stuff nothing changes... it seems that the watermark stuff works only when the output is a file... if the output is a feed of ffserver it doesn't work
[16:47] <Tjoppen> in other words: it has nothing to do with watermarks, but probably relates to ffm. the trac ticket saste pasted seems relevant
[16:48] <saste> see also #1930
[17:52] <finics> Hello, what the notification here is meant by for x264 use "profile/preset/tune", as I want High profile but still not able to manage.
[17:53] <finics> ffmpeg -y  -i <input> -map 0:0 -r 25 -aspect 16:9 -pix_fmt yuv420p -flags +ildct+ilme -profile high -preset ultrafast -c:v libx264 -x264opts level=41:keyint=24:tff=1:cabac=1:ref=3:force-cfr:fps=25:nal-hrd=cbr:bframes=3:bitrate=12000:vbv-maxrate=12000:vbv-bufsize=6000 -c:a mp2 -b:a 192000 -map 0:7  -metadata:s:a:0 language=spa -map 0:8 -metadata:s:a:1 language=eng   -ss 0 -t 5 -loglevel info ...
[17:53] <finics> ...<output>.ts
[17:54] <finics> Above command gives Main at 4.1, Should I need to add anything else?
[18:01] <finics> Anyone?  Should I use 8x8 DCT command for High profile?
[18:11] <JEEB> hmm
[18:13] <JEEB> finics, libx264 default is "high profile" which then falls back to the minimum profile that your used feature actually needs
[18:13] <JEEB> although I'm surprised that it does override it even when setting it, but you really don't have to set the high profile
[18:13] <JEEB> you're most definitely not using high profile features with that preset and all :P
[18:14] <finics> ANother thing I read somewhere is, i am using here Interlaced , this may create problem
[18:14] <JEEB> it will just make x264 not able to use all the possible compression features, that's all
[18:15] <finics> Here
[18:15] <finics> --profile name
[18:15] <finics> The options will be baseline, main, and high, which do what you think they do. Note that these override all other options no matter what, with two exceptions.. Because interlaced and lossless coding are options with very explicit meanings, the encoder will terminate with an error if you attempt to use them with an incompatible profile.
[18:16] <finics> Can you help me with High profile feature, Here I tried "-partitions all" and I am getting error for this command "-flags 8x8dct"
[18:16] <JEEB> ...
[18:16] <JEEB> set a slower preset
[18:16] <JEEB> you are not using high profile features
[18:16] <JEEB> and those features make encoding slower
[18:16] <JEEB> finics, I think only baseline profile will error out if you try to set it with interlacing
[18:17] <JEEB> because baseline does not support interlaced
[18:17] <JEEB> :P
[18:17] <finics> Oh ok.
[18:18] <JEEB> if the profile gets downgraded, it just means that you are not using the features of that profile -> you might as well not flag the stream as the highest profile
[18:18] <finics> Oh yes, ultrafast will suppress some of features.
[18:18] <JEEB> "some"
[18:18] <JEEB> more like most
[18:18] <JEEB> it's one of those presets that you generally shouldn't be using
[18:19] <JEEB> generally you want to steer away from the slowest and the fastest presets
[18:19] <JEEB> because the fastest is for speed benchmarkes
[18:19] <JEEB> *benchmarks
[18:19] <finics> Yes, I understand that.
[18:19] <JEEB> and slowest doesn't offer enough compared to the addition slowness
[18:22] <finics> Yes, Right. It adds compression features but have very minor effects on quality.
[18:22] <finics> Voila, with slower preset I got High profile.
[18:23] <JEEB> also I'm not sure if your input is interlaced if you still need -flags +ildct+ilme
[18:24] <JEEB> also why are you specifically enabling cabac?
[18:24] <finics> My source is not always Interlaced. But I need Interlaced MBAFF at end.
[18:24] <JEEB> (I also have other comments about those x264opts)
[18:24] <JEEB> k
[18:24] <finics> CABAC is required.
[18:25] <JEEB> well if you use a slow enough preset that will be set automatically
[18:26] <JEEB> only ultrafast seems to be turning off cabac
[18:26] <JEEB> superfast and slower from that all have CABAC on
[18:26] <finics> Yes True.
[18:29] <finics> Also I think nal-hrd flag is not required.
[18:31] <JEEB> nal-hrd info is nice to have, but cbr depends on if you want padding or not
[18:31] <JEEB> nal-hrd cbr will basically pad it
[18:31] <JEEB> nal-hrd vbr will just put the nal-hrd info there
[18:35] <finics> Ok.
[18:47] <pierre-olivierro> Hi all, i'm currently using FFmpeg API for reseaarch stuff
[18:50] <pierre-olivierro> For this I'm using libx264 -preset and also extradata stuff in avpacket structure. So my question is simple, is there a size limitation in extradata use ? How can i increase this limit ? Or maybe its a decoding problem : with my decoder i got some problem (stream vui errors or other), but when i try with VLC, all seems ok
[18:50] <pierre-olivierro> Any ideas ?
[19:04] <D4rthB4n3> hi i have a .avi file with two audio tracks and want to change the order of these so that the original 2nd audio Track becomes the new first Audio Track what is the easiest way to do so
[19:04] <JEEB> -map
[19:05] <D4rthB4n3> -map ???
[19:05] <JEEB> map is the setting that maps streams
[19:06] <ubitux> -map 0:1 -map 0:0 -c copy
[19:07] <D4rthB4n3> ok i ll try that thx
[19:07] <JEEB> hmm, I thought it was <input file nr>:<input side track number>:<output side track number> ?
[19:08] <JEEB> so like -map 0:1:2 -map 0:2:1 if the audio streams are the streams 1 and 2 of input
[19:08] <JEEB> (streams start from zero)
[19:08] <JEEB> (and I basically guessed that track 0 was video)
[19:09] <D4rthB4n3> yeah i think u r right JEEB
[19:10] <D4rthB4n3> will try it the other thing didnt work
[19:11] <D4rthB4n3> didnt work as well :(
[19:12] <D4rthB4n3> this is my output
[19:12] <D4rthB4n3> http://pastebin.com/6mvW71Mt
[19:12] <JEEB> ok, that's libav
[19:12] <JEEB> switch to the avconv command
[19:12] <D4rthB4n3> ?
[19:12] <JEEB> also you need to add mappings to /after/ -i
[19:12] <JEEB> D4rkSilver, you are not using ffmpeg, you are using a fork :) They have the maintained application called 'avconv'
[19:12] <JEEB> so if you use libav, you use 'avconv'. if you use ffmpeg, you use 'ffmpeg'
[19:13] <JEEB> avconv -i test.avi -c: copy -map 0:1:2 -map 0:2:1 test1.avi
[19:13] <JEEB> like this
[19:13] <JEEB> uhh
[19:13] <JEEB> -c copy not -c:
[19:13] <D4rthB4n3> ok will try
[19:15] <StaRetji> Howdy folks. Has someone successfully captured dvb-t channed with usb card? I mean, I read entire google lol and couldn't find working example
[19:15] <finics> @D4*
[19:15] <finics> Stream #0.0: Video: mpeg4 (Advanced Simple Profile), yuv420p, 576x432 [PAR 1:1 DAR 4:3], 25 tbr, 25 tbn, 25 tbc
[19:15] <finics>     Stream #0.1: Audio: mp3, 48000 Hz, stereo, s16, 104 kb/s
[19:15] <StaRetji> I would appreciate a little help here, thx :)
[19:15] <finics>     Stream #0.2: Audio: mp3, 48000 Hz, stereo, s16, 102 kb/s
[19:15] <D4rthB4n3> ok did it with the command u posted but now i have an audio that no one can understadt just random noises ???
[19:16] <finics> You want to take the second stream in output?
[19:16] <JEEB> D4rkSilver, lolwut
[19:16] <JEEB> pastebin command and output
[19:16] <D4rthB4n3> D4rthB4n3 ;) and what ???
[19:16] <D4rthB4n3> ok
[19:16] <JEEB> finics, no he wanted to make stream 0.1 into out.2 and 0.2 into out.1
[19:16] <D4rthB4n3> http://pastebin.com/zfD3g90B
[19:17] <finics> He was talking about track, so I thought.
[19:17] <JEEB> lol
[19:17] <D4rthB4n3> sorry its the first time i m doing such kind of thing ;)
[19:17] <JEEB> D4rkSilver, try adding -map 0:0:0
[19:17] <JEEB> it didn't copy the video
[19:17] <JEEB> lol
[19:17] <D4rthB4n3> add it where ?
[19:18] <JEEB> after -i and wherever you like otherwise, I'd put it before -map 0:1:2
[19:18] <D4rthB4n3> ok
[19:19] <finics> ffmpeg -i <source> -map 0:0 -map 0:2 -map 0:1  <out>   , shouldn't this be
[19:20] <D4rthB4n3> ok now i got the video with it but its still noises and not the real audio
[19:20] <JEEB> finics, that just selects them all
[19:20] <JEEB> I think
[19:20] <JEEB> zero'th input, streams one two three
[19:20] <JEEB> D4rkSilver, once again pastebin time
[19:20] <JEEB> also we might switch to #libav at this point :)
[19:20] <JEEB> since you are using it
[19:20] <finics> Yes, he is talking about full track swap, not channel swap. Is that right Mr D4?
[19:20] <finics> Sorry, I forget that name.
[19:21] <JEEB> he wants track zero to be zero, track one be track two, and track two be track one
[19:21] <JEEB> aka, switch the order of the audio tracks
[19:21] <D4rthB4n3> D4rthB4n3 and i think thats what i m talking about
[19:22] <finics> Ok than JEEB, you are swapping channels, not tracks.
[19:22] <D4rthB4n3> http://pastebin.com/ZUbajJZT
[19:22] <JEEB> no
[19:22] <JEEB> -map doesn't switch channels
[19:22] <JEEB> only tracks
[19:23] <D4rthB4n3> in handbrake both audio traks r marked as ch2.0
[19:23] <JEEB> hmm, I guess I got the -map wrong then...
[19:23] <JEEB> because 0:1 becomes 0:1 of output
[19:23] <JEEB> and so forth
[19:23] <JEEB> meh
[19:23] <D4rthB4n3> i m trying finics way maybe that works
[19:23] <JEEB> the documentation of -map could really be better
[19:23] <ubitux> < JEEB> finics, that just selects them all // yes but it re-orders
[19:23] <finics> Ok, but for track swap 0:1 will take full 1st Audio and 0:2 will 2nd Audio track.
[19:23] <ubitux> -map 0:0 -map 0:2 -map 0:1 is fine
[19:24] <ubitux> afaict.
[19:24] <JEEB> oh
[19:24] <JEEB> so it takes them by order?
[19:24] <ubitux> yes
[19:24] <JEEB> that makes sense
[19:24] <JEEB> k
[19:24] <D4rthB4n3> trying that atm hope it ll work :D
[19:24] <finics> Right ubitux.
[19:25] <JEEB> what was the correct syntax for setting the input and output side track number with -map ?
[19:26] <JEEB> I see there's a ,sync_file_id:stream_specified
[19:26] <JEEB> *specifier
[19:26] <JEEB> so -map 0:1,0:2 ?
[19:27] <finics> I think we ned to use map_channel for that pupose.
[19:27] <JEEB> no
[19:27] <JEEB> map_channel is for channel switches
[19:27] <JEEB> not tracks
[19:27] <JEEB> afaik
[19:27] <D4rthB4n3> ok now tracks r changed right thx a lot for that but the video got really bad
[19:27] <D4rthB4n3> its really blurry ???
[19:28] <JEEB> you forgot -c copy?
[19:28] <finics> map_channel 0.1.1:0.1.0 map_channel 0.1.0:0.1.1 for swapping tracks? ?
[19:28] <D4rthB4n3> hm yeah that is probably it :D
[19:28] <D4rthB4n3>  avconv -i test.avi -map 0:0 -map 0:2 -map 0:1 test1.avi
[19:28] <D4rthB4n3> and then add the -c somewhere in there
[19:28] <JEEB> after -i set -c copy
[19:29] <ubitux> finics: map channel is for swapping channels
[19:29] <JEEB> ^
[19:29] <finics> ** map_channel 0.1.1:0.1.0 map_channel 0.1.0:0.1.1 for swapping channels.
[19:30] <ubitux> it will require a re-encode
[19:30] <D4rthB4n3> now i got the noises again :(
[19:30] <JEEB> go poke #libav
[19:30] <JEEB> possibly a bug in libav
[19:31] <finics> WHy you get noises, check the source.
[19:31] <finics> You are just copying audio.
[19:31] <D4rthB4n3> source fine
[19:31] <D4rthB4n3> http://pastebin.com/uyWfkd3F
[19:32] <ubitux> looks fine
[19:32] <D4rthB4n3> and without the -c copy it works just that the video quality gets really bad
[19:32] <JEEB> naturally, since any problem there might be with stream copy doesn't happen
[19:32] <ubitux> without -c copy it re-encodes
[19:32] <JEEB> audio quality is actually getting worse too
[19:32] <JEEB> you just don't hear it :P
[19:33] <D4rthB4n3> well thats not so bad just the video is important since its already not that well
[19:33] <relaxed> good
[19:33] <ubitux> the output says the copy is done, so it you shouldn't see any change to the audio
[19:33] <ubitux> unless the fork did some stupid things, which i doubt on this
[19:33] <finics> Random question, encoding of s302m LPCM wrapper is implemented?
[19:34] <JEEB> could be some bugs in general, that might have already gotten fixed on both sides or on one side
[19:34] <JEEB> or just weird stuff in avi
[19:35] <finics> As I saw some patch work.
[19:36] <D4rthB4n3> any idea on how to fix the video ?
[19:37] <StaRetji> folks, need example of piping mplayer to ffmpeg, thx
[19:37] <StaRetji> mplayer input | ffmpeg output
[19:37] <JEEB> D4rkSilver, first of all let's move to #libav so you might get heard by people who actually work on libav :)
[19:38] <D4rthB4n3> ohh sorry :D
[19:44] <StaRetji> hey burek mate, just the man I wanted to see :)
[19:44] <StaRetji> hehe
[19:44] <burek> :beer: :)
[19:44] <StaRetji> I am trying to restream dvb-t with ffmpeg, but would like to know if this is not possible, to give up :D
[19:45] <StaRetji> I tried mplayer, it plays, but I fail to pipe it to ffmpeg
[19:45] <StaRetji> system is headleass
[19:45] <StaRetji> headless*
[19:45] <burek> which cmd did you use
[19:46] <StaRetji> player -dumpstream -dumpfile recording.ts dvb://TF1 | /usr/local/bin/ffmpeg -i - -vf yadif -acodec libfaac -ar 44100 -ab 128k -coder ac -vcodec libx264 -b:v 536k -maxrate 728k -bufsize 2800k -profile:v baseline -preset:v veryfast -tune film -crf 20 -g 50 -f flv rtmp://X.X.X.X/live/livestream112012nomad
[19:46] <StaRetji> player -dumpstream -dumpfile recording.ts dvb://TF1  works just fine
[19:46] <burek> i dont use mplayer :/
[19:47] <StaRetji> I guessed, but I would like to use ffmpeg, mplayer is just here because I don't know how to capture dvb-t with ffmpeg
[19:48] <StaRetji> anyway, I am confused with piping, is there documentation on piping and ffmpeg in linux?
[19:56] <burek> well if you use -f mpegts, piping should be fine
[19:56] <burek> but i think mpegts is kinda broken or something, didn't check recently
[19:56] <burek> was broken several months ago when i was testing it
[19:59] <StaRetji> thx bro
[20:04] <maujhsn> Need some troubleshooting help on a new build attempt using ubuntu-11.10  http://pastebin.com/4KiWzigF
[20:06] <relaxed> maujhsn: make distclean, ./configure and build it again.
[20:07] <Mavrik> burek: mpegts works just fine.
[20:07] <Yulth> Hi folks! I've an existential doubt with mp3 and aac transcoding. I know that transcoding from mp3 to aac isn't a good idea because the quality obtained is generally worst. To here is all right, but I've been surprised transcoding mp3 at 40kbps aac because some song's pieces heard better than same pieces transcoded at 64kbps aac. What could be the reason for wich was happened?
[20:09] <maujhsn> relaxed what directory should I run this command in i.e /ffmpeg or /home?
[20:09] <relaxed> ffmpeg
[20:10] <maujhsn> I'll give it a shoot!
[20:10] <relaxed> shot
[20:10] <Mavrik> Yulth: which AAC encoder did you use?
[20:10] <Mavrik> Yulth: it might be it used HE-AAC which does some magic to increase percieved quality
[20:11] <maujhsn> releaxed "shot"
[20:12] <Yulth> Mavrik: of course :) I've used libfdk_aac, profile aac_he from 192kbps mp3 source
[20:14] <Mavrik> Yulth: yeah, that would be it
[20:14] <maujhsn> relaxed check this out: http://pastebin.com/ULtQhHpK
[20:15] <Mavrik> Yulth: iirc fdk-aac (like most encoders) switches to low-complexity profile over 64kbps
[20:15] <Mavrik> Yulth: so you could hear some difference around that line )
[20:18] <Yulth> Mavrik: mmm I understand. These differences remain at 72kbps but in most cases, seem to disappear at 80kbps and higher
[20:19] <maujhsn> Anybody willing to tackle this issue? http://pastebin.com/4KiWzigF then this: http://pastebin.com/ULtQhHpK
[20:21] <relaxed> maujhsn: No. Run "make distclean" and then follow the directions pertaining to building ffmpeg in the wiki.
[20:22] <relaxed> maujhsn: If that doesn't work, "rm -rf ffmpeg" and try checking out the source again.
[20:22] <maujhsn> So your talking about the multiple steps rule in that tutorial...right?
[20:23] <maujhsn> relaxed So your talking about the multiple steps rule in that tutorial...right?
[20:23] <Yulth> Mavrik: is there any way to fix it?
[20:23] <relaxed> maujhsn: http://goo.gl/DPrRY
[20:32] <finics> Hello, I want to get 5.1 ac3 audio in s16 format, but in ffmpeg ac3 encoder gives error that it does not support s16 format, only fltp.
[20:37] <maujhsn> relaxed I don't like building from source...I  like seeing build instructions...any thoughts?
[20:37] <maujhsn> http://goo.gl/DPrRY
[21:17] <achour12> hi im having a problem with ffmpeg usage
[21:18] <achour12> when i do : ffmpeg -i rtmp://URL.tv:1935/live?email=xman@email.com&password=96d723b568b071063cded3bdeffa7134/237.stream -f flv -c:v libx264 test.mp4 , i got error 'password' is not reconized as internal or external command
[21:20] <achour12> any one can help plz?
[21:21] <achour12> ...
[21:24] <achour12> hi all
[21:39] <michaelni> achour12, your command is wrongly quoted/escaped this is a basic unix thing, not related to ffmpeg
[21:48] <achour12> im on windows server 2008
[21:48] <achour12> not linux
[21:48] <achour12> so how should the command looks like ???
[21:55] <relaxed> achour12: quote the input
[21:55] <cbsrobot> damn relaxed was quicker
[21:55] <cbsrobot> maybe you have to url escae it too
[21:55] <cbsrobot> meaning & becomes &
[21:56] <achour12> well i dont have it on the url
[21:56] <maujhsn> I want to Compile & Install  TAR BZ2 Files...ffmpeg version N-47538-geaa47e7 it currently resides in my /Downloads folder! Need the commands...Thanks!
[21:56] <cbsrobot> also the end of the url is strange : .../237.stream
[21:56] <achour12> yes its wowza server
[21:56] <cbsrobot> I never saw an slash back there
[21:56] <achour12> it does exist
[21:57] <achour12> <break>=/
[21:57] <achour12> so i replaced the break with /
[21:57] <cbsrobot> break ?
[21:57] <achour12> <break>
[21:57] <cbsrobot> space ?
[21:57] <cbsrobot> what is break ?
[21:57] <achour12> hold on i will give you the real url
[21:58] <cbsrobot> ha - thats a good starting point :-)
[21:58] <achour12> rtmp://URL.tv:1935/live?email=xman@email.com&password=96d723b568b071063cded3bdeffa7134<break>237.stream
[21:58] <relaxed> maujhsn: It's an archive with binaries inside, meaning it's already compiled for you. tar jxvf filename.tar.bz2
[21:58] <achour12> so you see the <break>
[21:59] <cbsrobot> achour12: where did you find that url ?
[21:59] <relaxed> maujhsn: Then you can run the program like: /home/you/dir/ffmpeg -i (reset of the command)
[21:59] <achour12> i doesnt work
[21:59] <maujhsn> relaxed Okay
[21:59] <cbsrobot> meaning can you share the link you found that url ?
[21:59] <achour12> it gives me 'password' is not reconized as command
[21:59] <achour12> i gave the link
[21:59] <cbsrobot> isit a justin.tv link ?
[22:00] <achour12> no
[22:00] <cbsrobot> the link to the page you found the url
[22:00] <achour12> oh
[22:00] <achour12> its a paid service
[22:00] <achour12> provider of iptv
[22:00] <achour12> im trying torestream it to all my tvs
[22:00] <cbsrobot> but is <break> a special character ?
[22:00] <achour12> to re-stream *
[22:00] <cbsrobot> linebreak ?
[22:01] <achour12> well
[22:01] <achour12> i found many example
[22:01] <cbsrobot> or is it really written "<break>" ?
[22:01] <achour12> ppl changing <break> to /
[22:01] <achour12> and it works
[22:01] <cbsrobot> ah ok
[22:01] <achour12> but the problem here
[22:01] <achour12> is the authentification
[22:01] <cbsrobot> so when you quote it - doeas it work ?
[22:01] <achour12> no
[22:01] <cbsrobot> meaning putting "" around
[22:01] <cbsrobot> can you pastebin it ?
[22:02] <cbsrobot> like this:
[22:02] <achour12> ok wait
[22:02] <maujhsn> relaxed What's this crap: ubuntu at ubuntu:~$ sudo tar xvfj ffmpeg-linux64-20121206.tar.bz2 tar (child): ffmpeg-linux64-20121206.tar.bz2: Cannot open: No such file or directory tar (child): Error is not recoverable: exiting now tar: Child returned status 2 tar: Error is not recoverable: exiting now
[22:03] <maujhsn> relaxed What's this crap:
[22:03] <achour12> sorry i forgot the command to print log
[22:03] <achour12> -report???
[22:03] <achour12> or what
[22:03] <cbsrobot> I guess
[22:03] <cbsrobot> or just copy paste it
[22:03] <cbsrobot> make sure you scramble at least the password
[22:04] <achour12> of course
[22:04] <cbsrobot> if you dont' want to share it with the world
[22:04] <cbsrobot> btw
[22:04] <cbsrobot> did you try to replace the & with & ?
[22:06] <maujhsn> Relaxed What kind of stupid game is this? http://pastebin.com/kiqD0gME
[22:06] <achour12> here we go: http://pastebin.com/it8c4WTW
[22:06] <achour12> when i use ffplay = it works
[22:06] <achour12> with no probleme
[22:07] <achour12> but ffmpeg -i as input it gives me error
[22:08] <achour12> on the link or command line there's no &
[22:09] <cbsrobot> achour12: &l.com&password....
[22:09] <cbsrobot> see the & ?
[22:09] <achour12> hold on
[22:11] <achour12> i replaced & with &
[22:11] <cbsrobot> achour12: so put "" around
[22:11] <achour12> or just &amp
[22:11] <cbsrobot> and use \&
[22:11] <cbsrobot> eighter \&
[22:11] <achour12> i didnt get it
[22:11] <cbsrobot> or &
[22:11] <cbsrobot> so before password....
[22:12] <cbsrobot> eighter you write: ffmpeg -i "rtmp:// &&ur\&passord=&.."
[22:12] <cbsrobot> see the " ?
[22:12] <maujhsn> relaxed SORRY...MY BAD!
[22:12] <achour12> yeah it didnt work
[22:12] <cbsrobot> with the backslash ?
[22:12] <achour12> yeah
[22:12] <cbsrobot> or try
[22:13] <cbsrobot> sorry
[22:13] <cbsrobot> same error ?
[22:13] <cbsrobot> when using \& ?
[22:13] <achour12> when i do \& = 'password' is not reconized as command
[22:13] <cbsrobot> no
[22:13] <cbsrobot> not \& =
[22:13] <cbsrobot> jsut \&password=
[22:13] <cbsrobot> *just
[22:14] <achour12> when i replaced with &   amp is not reconized
[22:14] <achour12> gmail.com&password
[22:14] <achour12> gmail.com\&password
[22:14] <achour12> none works
[22:14] <cbsrobot> hmm
[22:14] <cbsrobot> always the same error ?
[22:15] <achour12> \& = password not reconized as command
[22:15] <pierre-olivierro> yes
[22:15] <achour12> & =     amp not reconized as command
[22:16] <cbsrobot> even wit hthe quotes around ?
[22:16] <cbsrobot> and using  &.gmail.com^&password=&.
[22:18] <achour12> ok i will try this
[22:21] <achour12> good step
[22:21] <achour12> gave some other errors to get the stream
[22:23] <achour12> cbsrobot
[22:24] <cbsrobot> achour12: ok
[22:24] <achour12> i will past you the next errors ok ?
[22:24] <cbsrobot> go ahead
[22:24] <cbsrobot> maybe other people will help too
[22:25] <achour12> http://pastebin.com/zsF9MP47
[22:25] <achour12> i feel that we are soooo close
[22:26] <cbsrobot> [flv @ 013d5780] FLV does not support sample rate 48000, choose from (44100, 220
[22:26] <cbsrobot> 50, 11025)
[22:26] <burek> [flv @ 013d5780] FLV does not support sample rate 48000, choose from (44100, 220
[22:26] <burek> 50, 11025)
[22:26] <cbsrobot> it's all written - you know
[22:26] <cbsrobot> hehe
[22:26] <burek> :)
[22:26] <JEEB> huuh
[22:26] <JEEB> that's not true btw
[22:26] Action: cbsrobot hands another beer to burek
[22:26] <burek> cheers :)
[22:26] <cbsrobot> JEEB: huh
[22:26] <JEEB> at least with AAC it supports other rates too
[22:26] <achour12> so whats the command for it
[22:26] <achour12> ?
[22:27] <cbsrobot> yeah - sounds fishy
[22:27] <JEEB> lemme link the spec
[22:27] <cbsrobot> achour12: read the manual :-)
[22:27] <achour12> hahahaha
[22:27] <cbsrobot> or add -r:a 44100 tothe command line
[22:27] <JEEB> I think ffmpeg might actually be limited to those
[22:27] <JEEB> but you should def. report it as a bug
[22:27] <JEEB> if it doesn't let you do 48kHz AAC
[22:27] <JEEB> in FLV
[22:27] <cbsrobot> well its liblame isnti t ?
[22:28] <JEEB> oh, mp3?
[22:28] <cbsrobot> yeah
[22:28] <cbsrobot> well
[22:28] <JEEB> lemme see how FLV specifies that
[22:28] <achour12> i choose mp4
[22:28] <Mavrik> the FLV specification does not allow 48000 samplerate
[22:28] <JEEB> Mavrik, except
[22:28] <Mavrik> only 11050, 22100, 44200 for AAC audio
[22:28] <JEEB> for AAC
[22:28] <JEEB> no
[22:28] <JEEB> incorrect
[22:28] <Mavrik> for AAC as well
[22:28] <JEEB> sec
[22:28] <cbsrobot> mp4 is teh container, mp3 the codec
[22:29] <cbsrobot> no 8000 ?
[22:29] <JEEB> Mavrik, re-read page 71 of http://download.macromedia.com/f4v/video_file_format_spec_v10_1.pdf
[22:29] <achour12> so -r:a 44100 didnt work
[22:29] <JEEB> "If the SoundFormat indicates AAC, the SoundType should be 1 (stereo) and the SoundRate should be 3 (44 kHz). However, this does not mean that AAC audio in FLV is always stereo, 44 kHz data. Instead, the Flash Player ignores these values and extracts the channel and sample rate data is encoded in the AAC bit stream."
[22:29] <Mavrik> JEEB: yeah, but you can't set proper flags
[22:30] <Mavrik> which is why ffmpeg muxer won't let you
[22:30] <JEEB> well it specifically says for AAC
[22:30] <maujhsn> Mavrik your a pretty smart guy...where is this folder? /ffmpeg-linux64-20121206/ffmpeg bash: cd: /home/ubuntu/ffmpeg-linux64-20121206/ffmpeg
[22:30] <JEEB> that you set it to 1 and 3 /always/
[22:30] <cbsrobot> :-)
[22:30] <JEEB> stereo, 44kHz
[22:30] <JEEB> and flash player then proceeds to ignore those values
[22:30] <maujhsn> Mavrik your a pretty smart guy...where is this folder? /ffmpeg-linux64-20121206/
[22:30] <Mavrik> hrm, I see
[22:30] <Mavrik> maujhsn: stop spamming me please.
[22:30] <JEEB> that thing has been in the spec for quite a while now
[22:30] <achour12> cbsrobot plz gift me for christmas the right command loll
[22:30] <Mavrik> time to file a bug then
[22:31] <maujhsn> Mavrik I am not trying to spamm you!
[22:31] <Mavrik> I've just reviewed flv muxer code in ffmpeg yesterday for 1.0 and it specifically fails if samplerate isn't 44100 :\
[22:31] <JEEB> yeah, that's incorrect behavior :)
[22:31] <JEEB> at least looking at the spec
[22:31] <JEEB> (at least for AAC)
[22:31] <cbsrobot> achour12: so wait another 13 days ?
[22:32] <achour12> looooool
[22:32] <achour12> u could gift me now
[22:32] <cbsrobot> ha - opening xmas gift before xmas is bad luck - isn't it ?
[22:32] <achour12> well im stuck now lol
[22:33] <cbsrobot> achour12: you need that combo: flv, hs64 nd mp3 ?
[22:33] <cbsrobot> *h264 and
[22:33] <achour12> yeah just shoot the command plz i tried -f flv -r:a 44100 -c:v libx264
[22:33] <JEEB> Mavrik, basically there seem to be three special cases, nellymoser, aac and speex. for nellymoser it's about low sampling rates, and pretty much the same thing for speex :) For AAC it's pretty much "ignore the container data lol"
[22:34] <Mavrik> yeah, love those kind of containers
[22:34] <Mavrik> "these parameters mean THIS but if those other 3 are set in this arcane combination DO THIS and ignore everything"
[22:35] <cbsrobot> try: ffmpeg -i rtmp://distribution03.arabtvnet.tv:1935/live?email=123654ur%40gmail.com^&password=96d72315463cded3bdeffa7834/576.stream -f flv -c copy cc.mp4
[22:37] <achour12> error
[22:37] <achour12> rtmp_readpacket, failed to read rtmp packet body. len:475
[22:42] <cbsrobot> try again ?
[22:46] <achour12> i did
[22:46] <achour12> but no clue
[23:56] <joelio> 2
[23:56] <joelio> doh
[23:57] <klaxa> 3
[00:00] --- Tue Dec 11 2012