[Ffmpeg-devel-irc] ffmpeg.log.20121024

[burek]

[00:13] <klaxa> Endorgh: http://www.icecast.org/3rdparty.php
[00:13] <klaxa> that's the list of icecast source clients
[00:13] <klaxa> Endorgh: this is probably what you want: http://www.icecast.org/ices.php
[00:14] <Endorgh> but
[00:14] <Endorgh> is ices able to strean aac and acc+?
[00:14] <klaxa> hmmm... doubt it
[00:15] <klaxa> do you have to use icecast or do you just want an http/tcp stream? if you just want an http/tcp stream ffserver could do the job
[00:18] <Endorgh> the purpose is to set up a complete and reliable radio station. Icecast does it very well, Can ffmpeg do the same?
[00:19] <klaxa> i'm actually not sure
[00:20] <klaxa> i rarely use ffserver and stopped using icecast
[00:20] <klaxa> i bet there is a way to use ffmpeg as a source client for icecast
[00:20] <llogan> perhaps you can pipe from ffmpeg to oggfwd to icecast
[00:21] <klaxa> would that work with aac and aac+ as he wishes
[00:21] <klaxa> ?
[00:21] <llogan> as input, yes, but i was assuming vorbis output
[00:21] <Endorgh> I'm confused because major radio stations use icecast and stream aac+, but I don't know what kind of source client they use to stream aac
[00:23] <klaxa> Endorgh: this might be of interest: http://stackoverflow.com/questions/5215019/icecast-2-protocol-description-streaming-to-it-using-c-sharp/9985297#9985297
[00:23] <Endorgh> llogan: this process implies reencoding?
[00:23] <klaxa> as well as this: http://stackoverflow.com/questions/12885597/icecast2-two-sources-same-streaming (although the question is a bit different)
[00:24] <llogan> Endorgh: probably since i would guess oggfwd only works with ogg friendly stuff
[00:24] <llogan> you could always ask the radio stations what they use
[00:25] <klaxa> you could hack something together with shell scripting and netcat (here seems to be a quite good documentation of the icecast protocol: http://forums.radiotoolbox.com/viewtopic.php?t=74 )
[00:25] <klaxa> asking the stations how they do it would be the first thing i'd do before writing something myself... wait actually i'd try to write it myself before asking others :X
[00:26] <Endorgh> llogan: I don't think want tell to me this kind of things hehehe
[00:26] <Endorgh> *they want
[00:27] <llogan> then don't even try
[00:27] <klaxa> you could try asking in the icecast channel what source client they would recommend for your usecase
[00:28] <Endorgh> okok, thanks for the suggestions! :)
[00:29] <Endorgh> klaxa: they say that only can offer support for non-commercial audio formats
[00:29] <klaxa> mmh... understandable
[00:44] <buhman> ooh
[00:44] <buhman> it's tuesday
[00:49] <cbsrobot> Endorgh: what do you wanna do exactly ?
[00:53] <Endorgh> cbsrobot: I want to stream aac+ audio to an Icecast server
[00:53] <cbsrobot> one file ?
[00:53] <cbsrobot> playlist ?
[00:53] <cbsrobot> dynamic playlist ?
[00:54] <Endorgh> yes, directly from a file with no resampling (or playlist, perhaps)
[00:54] <cbsrobot> no microphone ?
[00:55] <Endorgh> no
[00:55] <Endorgh> only files
[00:55] <Endorgh> previously converted to aac or m4a container
[00:55] <cbsrobot> I'd go with liquidsoap
[00:56] <cbsrobot> but in my expereience it's better to use mp3 in the backend
[00:56] <cbsrobot> and convert the stream to aac+ at the end
[00:57] <Endorgh> exactly
[00:57] <Endorgh> but liquidsoap works with raw data, and it implies to reencode using CPU resources
[00:57] <Endorgh> for me, this is a problem
[00:58] <cbsrobot> not sure
[00:58] <cbsrobot> maybe there is a way to do it without reencoding
[01:00] <Endorgh> I haven't found out the way to do it. Any suggestions?
[01:04] <Kireji> I was using a voicerecorder to record a meeting at work and my HTC G2 ran out of power and shut down
[01:04] <Kireji> the resulting 3gp file is not playable with VLC or mac Quicktime - how can I go about repairing it and getting what audio I can from the file?
[01:04] <Kireji> http://www.3gprecovery.com/ has some videos and forum posts, but I have low confidence they are reputable
[01:04] <Kireji> when I open it with "ffmpeg  -i broken.3gp out.mp3"  I get broken.3gp: Error while opening file
[01:05] <Kireji> specifically "[mov,mp4,m4a,3gp,3g2,mj2 @ 0xdff260]moov atom not found"
[01:05] <Kireji> what can I do?  it's got 8.5MB of data we really want
[01:06] <cbsrobot> Endorgh: no it doesn't seem so
[01:07] <durandal_1707> Kireji: recreate moov atom
[01:08] <Kireji> ok, that sounds great, can ffmpeg do that?
[01:08] <Kireji> man'ing
[01:09] <durandal_1707> Kireji: no, what audio codec was used?
[01:17] <Kireji> I'm not sure, I think it's mp4
[01:17] <Kireji> it's whatever is default on an Android HTC G2
[01:18] <Kireji> I'm playing now with AtomicParsley and it's saying there are 0 tracks, I have no idae what I'm doing
[01:19] <durandal_1707> Kireji: it does not output anything instead of that?
[01:20] <Kireji> I'm looking now at working recordings made with the same program, and it outputs a bunch
[01:21] <Kireji> http://i.imgur.com/TGkoj.png
[01:21] <Kireji> working and broken .3gp files
[01:22] <Kireji> durandal_1707: I may have misunderstood your question
[01:24] <Kireji> a few different places (like http://blog.alwayshere.info/2011/01/ffmpeg-moov-atom-not-found.html) says this program may be able to add or fix moov atoms
[01:24] <dislo> Hey Everyone, I am trying to take a RTSP stream in and serve a RTSP stream back out for android phones and the apple http streaming protocol for iwhatevers. Can I do this using ffconv and ffserver?
[01:25] <llogan> what's ffconv?
[01:25] <dislo> llogan, ffmpeg sorry
[01:32] <durandal_1707> Kireji: Kireji only soluton i see it to learn mov format open hex editor and try to find and save audio data, to make use of saved data you will also need to know bitstream of audio codec and you will need to manualy mux audio packets into containter that support such codec
[01:33] <Kireji> I have a bunch of other examples of correctly recorded files from work, that should all have the correct headers
[01:33] <Kireji> like the other example in that picture
[01:34] <durandal_1707> muxing should not be hard...
[01:34] <Kireji> would it work to take another recording, rip out all the data and paste in the longer data from the broken file?
[01:34] <Kireji> I don't know what muxing is
[01:35] <durandal_1707> Kireji: moov atom is located at end of file
[01:36] <Kireji> ah
[01:36] <Kireji> crap
[01:37] <durandal_1707> you need to locate each audio packet in file and remux it
[01:37] <Kireji> this something a program can do, something like a repair program, or something that takes human effort?
[01:37] <durandal_1707> thats why it is imortant what audio codec was used
[01:38] <Kireji> if human work, not worth it, it was an important meeting, 8 of us were talking for 1.5h.  not going to spend sig $ on it
[01:38] <durandal_1707> it is probably amrwb...
[01:38] <Kireji> ah ok,  let me see if I can get codec from another successful recording from same program
[01:38] <Kireji> amrwb?
[01:39] <durandal_1707> Adaptive Multi-Rate WideBand
[01:39] <durandal_1707> or Adaptive Multi-Rate NarrowBand
[01:40] <durandal_1707> probaby some trivail parser could be written taking several assumptions ....
[01:40] <Kireji> can ffmpeg tell me the codec?
[01:40] <Kireji> man'ing
[01:40] <durandal_1707> ffmpeg -i input.file
[01:41] <Kireji> Stream #0.0(eng): Audio: samr / 0x726D6173, 8000 Hz, mono, s16
[01:42] <Kireji> is that it?
[01:42] <Kireji> or
[01:42] <durandal_1707> samr is fourcc
[01:43] <durandal_1707> for AMRNB
[01:44] <durandal_1707> and looking at code it is not hard to parse....., but that exercise is left to the reader
[01:44] <Kireji> ok
[01:44] <Kireji> thank you
[02:08] <RunawayDevil> which codec should I use to make a DVD HD (29.4 Mbps)?
[02:09] <RunawayDevil> using mpeg2video the maximum is 9.4mb (to make a vob)
[02:09] <klaxa> DVD standard is mpeg2 if i'm informed correctly
[02:09] <klaxa> hmm...
[02:09] <RunawayDevil> standard is that :/
[02:10] <RunawayDevil> it's because I have here many DVD DL, and it's very bad when I see just 3 gb full and 5 empty, it's like wasting
[02:10] <RunawayDevil> so i've decided to up
[02:10] <klaxa> i can imagine
[02:12] <RunawayDevil> anyone else?
[02:12] <klaxa> what is the source's bitrate?
[02:15] <RunawayDevil> https://www.mercadolibre.com/jms/mlb/lgz/login/otp?otp=044bcc791a300ccbb24069f2b9500477NOMCu1hVeb--RESTO_1-RESTO_2&go=http://produto.mercadolivre.com.br/MLB-437438618-100-dvd-r-maxiprint-printable-ate-o-eixo-consulte-frete-_JM
[02:15] <RunawayDevil> ops
[02:15] <RunawayDevil> sorry lol
[02:16] <RunawayDevil> klaxa it's 14mb
[02:16] <klaxa> so you want to push it up to 29.4 mbps?
[02:16] <klaxa> just to fill the dvd?
[02:16] <RunawayDevil> maybe yes, obviously i will have some gain in quality, no
[02:16] <RunawayDevil> ?
[02:17] <klaxa> you could gain some quality with post processing i guess... but just pushing up the bitrate usually doesn't increase quality
[02:17] <klaxa> but wastes space
[02:18] <klaxa> do you want to play it with your dvd player or put the dvd's in a computer?
[02:18] <RunawayDevil> dvd player in 1080p
[02:19] <klaxa> is that even possible?
[02:20] <RunawayDevil> im trying near as possible
[02:20] <klaxa> i haven't seen many dvd players, but none had a port that supports that resolution
[02:22] <klaxa> according to wikipedia (yay best source) the DVD format is a restricted mpeg2video definition as you already seem to know
[02:22] <klaxa> http://en.wikipedia.org/wiki/DVD-Video#Frame_size_and_frame_rate
[02:23] <klaxa> it's capped at 9.8 mbit/s
[02:23] <klaxa> probably (i'm almost certain) because of DVD player's read-speed
[02:26] <klaxa> http://en.wikipedia.org/wiki/DVD-Video#Data_rate <-- this appears to support my statement
[02:26] <cbsrobot> RunawayDevil: were you able to hardsub ?
[02:26] <klaxa> again wikipedia, probably not the most reliable source
[02:27] <RunawayDevil> cbsrobot: now yes, I'm to make a .vob first I convert .srt to .ass in jubler and so I hardsub with iFFmpeg
[02:28] <RunawayDevil> I use jubler to edit stuff too
[02:28] <RunawayDevil> cbsrobot http://s11.postimage.org/r51ucn1pv/Captura_de_Tela_2012_10_23_s_22_25_38.png
[02:28] <RunawayDevil> that's my config, what do you think?
[02:29] <cbsrobot> can you watch hd dvd ?
[02:29] <klaxa> what cable do you use to connect your dvd player to your tv?
[02:29] <cbsrobot> it's like eol for hd dvd: http://en.wikipedia.org/wiki/HD_DVD
[02:29] <cbsrobot> most use bluray nowadays
[02:30] <RunawayDevil> HDMI
[02:30] <cbsrobot> but you have a hd dvd player ?
[02:30] <RunawayDevil> yes
[02:30] <cbsrobot> ah ok
[02:31] <RunawayDevil> but for a simple DVD with good quality
[02:31] <RunawayDevil> do you think is that good? http://s11.postimage.org/r51ucn1pv/Captura_de_Tela_2012_10_23_s_22_25_38.png
[02:32] <cbsrobot> hmmm
[02:33] <cbsrobot> nah
[02:33] <cbsrobot> if you have a "HD DVD" Player
[02:33] <cbsrobot> I'd encode in avc
[02:33] <cbsrobot> not in mpeg2
[02:34] <RunawayDevil> hum
[02:35] <klaxa> was about to suggest that before my router decided to fuck up my wifi
[05:40] <dragunov_11> Hi, am having problem in transcoding a mjpeg stream from a camera to h264 stream over rtsp. http://pastebin.com/jnefyUiA
[05:41] <dragunov_11> says tcp connection failed, connection refused.
[09:34] <elkng> how to extract frames from 400 to 500 from video using frames number as parameter ?
[11:07] <jalalsfs> Hi all,
[11:08] <jalalsfs> I tried to record my desktop with ffmpeg but it doesn't work http://fpaste.org/BoLP/
[11:12] <elkng> jalalsfs: try this one: "ffmpeg -f x11grab -r 25 -s 1024x768 -i :0.0+100,200 output.flv"
[11:26] <jalalsfs> elkng, I looks the same look http://fpaste.org/Tm39/
[11:29] <elkng> jalalsfs: do you have radeon video card ?
[11:30] <Yulth> Hi everyone
[11:31] <gazzwi86> I have an issue with ffserver
[11:31] <gazzwi86> I'm trying to stream my webcam feed and can only get a download in chrome, one which doesn't download video
[11:32] <gazzwi86> from status.html it says its transferred 66bytes but no more
[11:34] <jalalsfs> elkng, no, it's Intel.
[11:44] <Yulth> How I can specify an output format when writing to stdout? This command line is giving me an error:  ffmpeg -i input.m4a -codec copy -f m4a  pipe:1
[11:53] <Yulth> any ideas?
[12:06] <Sashmo> hey guys, anyone have an idea on why I begin to get so much audio drift in my live encodings from transport stream?  I find that I have to restart my encoding every 5+ hours to fix the audio sync?
[12:21] <Yulth> any ideas? How I can to write to stdout m4a format?
[12:33] <brontosaurusrex> Yulth: perhaps ffmpeg -i input.m4a -acodec copy -f mp4 pipe:1
[12:38] <Yulth> Just at this moment I found out the solution: the format must be specified as "-f astd", because audio was aac
[12:38] <Yulth> sorry, "adts"
[12:39] <JEEB> that would be raw aac
[12:39] <JEEB> naturally, mp4 without fragments would not be open'able straight away, so it kind of makes sense
[12:42] <Yulth> mmm
[12:43] <Yulth> and is there any way to make a copy of m4a input file, without reencoding, to m4a stdout?
[12:43] <Yulth> preserving metdata, etc...
[12:51] <klaxa> Yulth: cat input.m4a | <next command to use the m4a from stdin>
[12:55] <Yulth> ok, here the pastebin:
[12:55] <Yulth> http://pastebin.com/FE1XdXWy
[12:58] <burek> pipe:: Invalid data found when processing input
[12:59] <burek> what are you trying to do?
[12:59] <burek> remux m4a to aac?
[13:00] <burek> try: ffmpeg -i Fisher_40k.m4a -acodec copy -f adts output.aac
[13:47] <jameshowe> ffmpeg 1.0 CLI - is there a way to pass the same encoder output to multiple muxers in a single command?
[14:18] <jalalsfs> I can't record from my desktop look at this http://fpaste.org/ZAOc/
[14:45] <burek> jameshowe http://ffmpeg.org/trac/ffmpeg/wiki/Creating%20multiple%20outputs
[14:45] <burek> maybe that can help
[14:56] <divVerent> burek: basically, the site says "no, ffmpeg can not do it, but the shell can" ;)
[14:57] <divVerent> one extra hint for jameshowe: try -f nut as format to be piped, because nut can contain "anything" unlike mpegts
[15:17] <teolicy> Hi. I'm trying to convert an mp4 stream to a raw YUV stream with pipes.
[15:17] <teolicy>  As you can see here (http://pastebin.com/EHAHHK1H), when I read the input file and write to an output file, my command line works.
[15:18] <teolicy> However, when I try to use pipes, my command line fails.
[15:18] <teolicy> The command line I used (with output) is in the paste, but I'll also copy it here:
[15:18] <teolicy> ffmpeg -i - -an -r 25 -f rawvideo -vcodec rawvideo -pix_fmt yuv420p - < input.mp4 > output.mp4
[15:18] <teolicy> And the input file (just a sample) can be retrieved here: https://s3.amazonaws.com/audish/input.mp4
[15:19] <teolicy> I tried playing a bit with analyzeduration and probesize, but not sure I used good numbers and didn't get better results.
[15:19] <teolicy> I suspect I should specify a -f and a -vcodec for my input, but not sure what to specify.
[15:19] <teolicy> Any help will be appreciated.
[15:52] <jameshowe> the trouble with doing it in multiple piped commands is that the MOV/MPG muxer can't deal with adts aac data unless it's just come out of the encoder
[16:21] <teolicy> I ran qt-faststart on my source and now it works fine through the pipe. Great!
[16:21] <ubitux> teolicy: you can remux with -movflags +faststart as well
[16:22] <teolicy> ubitux: you mean, when creating the original source, right?
[16:22] <ubitux> ffmpeg in-nofaststart.mp4 -c copy -map 0 -movflags +faststart out-faststart.mp4
[16:22] <ubitux> something like this
[16:22] <teolicy> aye, thanks.
[16:25] <jameshowe> just -movflags faststart, no +
[16:26] <ubitux> +faststart adds to the default flags
[16:26] <marlinc_> Any one in here who would like to try my live streaming application while it isn't fully ready yet
[16:26] <marlinc_> Need some suggestions and stuff
[16:30] <Simex> Someone?
[16:30] <dTal> you didn't exactly sell it :p
[16:31] <Simex> I know but it is true :p
[16:31] <Simex> Would you like try it?
[16:31] <Simex> It is open-source
[16:47] <rickbol> when transcoding a DV stream to mpeg4, can ffmpeg produce a "streamable" mpeg4? i.e. move the MOOV attribute (and what others) to the file header, etc?
[16:47] <rickbol> where can I find out how to construct the necessary parms to make this happen?
[16:47] <ubitux> -movflags +faststart
[16:48] <JEEB> of course that only means that it will become like that /after/ the encoding has finished
[16:48] <JEEB> if you need to have access to output ASAP
[16:48] <JEEB> you need the fragments feature
[16:48] <JEEB> I think libavformat's muxer should support that? at least the parser does.
[16:48] <ubitux> fragment? yes we have such thing
[16:48] <ubitux> and we also have a segmenter
[16:49] <rickbol> I would like to be able to view the trancode result (confidence monitoring) more-or-less in real-time.
[16:50] <JEEB> yeah
[16:50] <JEEB> fragments needed
[16:52] <rickbol> ultimately the resultant file will be posted for streaming from a website. Will fragments "complicate" that?
[17:04] <JEEB> rickbol, I wish I remembered the details of that (aka if the main index is generated or not) -- I know that some parsers probably are going to fail with just fragments
[17:46] <ogodon> total noob here, is using ffmpeg for converting .mkv HD into some SD format a bad idea?
[17:46] <ogodon> or is it like using a flamethrower to light a candle?
[17:51] <rickbol> ogodon: ffmpeg is among the most comprehensive tools for trancoding (converting) one format into another.
[17:52] <ogodon> thanks rickbol, I'll give it a shot then, I tried it some time ago and got frustrated, but I'm hoping for better results now
[17:54] <rickbol> the syntax isn't easy... and you have to know what your goal is, but if you work with audio\video,  ffmpeg is worth the effort it takes to learn it.
[17:57] <ogodon> thanks, any clues as to where to catch up with the syntax?
[18:07] <dragunov11> Hi, I needed help in transcoding a mjpeg stream to x264 stream, tried following loads of tutorials from google search, but doesn't work. can you guys help please?
[18:09] <rickbol> ogodon: google for your specific case. example... ffmpeg mkv to avi
[18:10] <rickbol> is ffmpeg changing name to avconv?
[18:16] <durandal_1707> rickbol: nope
[18:16] <ogodon> I did, didn't find it, but I think I worked it out... it's going to take a while to see if it worked though
[18:17] <rickbol> durandal_1707: ubuntu (ubuntustudio) is deprecating ffmpeg for avconv? wtf?
[18:20] <ubitux> rickbol: it's just propaganda, see above
[18:28] <ogodon> it works!!!!! thank you rickbol, you were right... :-)
[18:39] <Yulth> Hi everyone!
[19:01] <Yulth> It's known that HE-AAC is a non-free codec, and in all forums, chats, etc people say that one have to pay or adquire a valid license to use HE-AAC (or any variant) legally. It's all right?
[19:02] <durandal_1707> yes
[19:02] <Yulth> ok
[19:06] <Yulth> but reading the license requirements, etc, I've found out this webpage, on supposedly is shown all information about this regard. http://www.vialicensing.com/licensing/aac-faq.aspx
[19:06] <Yulth> and curiously, it shows literally:
[19:06] <Yulth> "License fees are due on the sale of encoders and/or decoders only. There are no patent license fees due on the distribution of bit-stream encoded in AAC, whether such bit-streams are broadcast, streamed over a network, or provided on physical media. "
[19:07] <Yulth> Does it mean that online radio stations are free to stream AAC channels without paying fees?
[19:07] <Yulth> Or how it works?
[19:08] <JEEB> IANAL and I don't know the full license, but it seems that if you don't distribute the decoder or encoder, you should be OK..?
[19:09] <JEEB> but you probably will need to poke them about getting the license itself
[19:11] <Yulth> So, Can an online radio station stream HE-AAC audio without problems?
[19:11] <Yulth> for me is very confusing...
[19:15] <Yulth> anybody have some knowledge on this matter?
[19:19] <JEEB> Yulth, ask via licensing if you want a proper answer
[19:19] <JEEB> but I would say that in most cases they really wouldn't give two derps, and IANAL
[19:20] <JEEB> I mean, it's funny how you actually ask that while everyone else just assumes that because H.264 can be used on the web for free if you distro it for free
[19:23] <Yulth> ok, I understand
[19:26] <durandal_1707> JEEB: so he can stream only if it stream from free OS?
[19:28] <JEEB> durandal_1707, uhh what
[19:28] <JEEB> what does that have to do with anything
[19:29] <durandal_1707> JEEB: nvm, i misread it
[19:29] <JEEB> the pasted point just notes that distro'ing the dec/encoders is payware, and otherwise you just have to get the paperwork done if you only distro the bit stream
[19:30] <JEEB> (I will guess they have the same thing as MPEG-LA, which means you have to get the paperwork done even if you don't pay)
[19:37] <k1ng__> hello
[19:38] <k1ng__> i am trying to convert mov to x264 mp4 but i am only getting audio
[19:38] <k1ng__> http://codepad.org/kAwwtI9P
[19:38] <dericed> k1ng__:  a guess, but try adding -pix_fmt yuv420p
[19:41] <k1ng__> ok let me try
[19:41] <k1ng__> cool
[19:41] <k1ng__> thanks a lot mate, its worked :D
[19:41] <k1ng__> dericed, you rock
[19:53] <dericed> k1ng__:  no problem
[19:53] <k1ng__> :)
[21:36] <jalalsfs> Hello there.
[21:37] <llogan> jalalsfs: hello
[21:38] <jalalsfs> Hello llogan, hwo are you?
[21:41] <jalalsfs> When I try to record desktop by FFmpeg it doesn't work look at this http://fpaste.org/CHfx/
[21:42] <llogan> ffmpeg tells you that when your -s exceeds the actual screen size
[21:42] <llogan> also, don't use sameq
[21:42] <llogan> http://superuser.com/a/478550/110524
[21:43] <llogan> and here is a good guide: http://ubuntuforums.org/showthread.php?p=8746719#post8746719
[22:07] <tr0m> could someone kindly have a look at http://pastebin.com/yjGRPHNE please, i cant script...
[22:08] <tr0m> maybe im missing something, the script works on fc systems but not so far on centos
[22:09] <llogan> 4mv is now mv4.
[22:10] <llogan> if you're running the same script on different ffmpeg versions you will likely experience syntax differences
[22:10] <tr0m> yes
[22:10] <tr0m> makes sense
[22:11] <wm4> what are libswscale's semantics about memory between end and start of a line? (i.e. that space that is used for alignment or when the image is a cropped view of a larger image)
[22:11] <wm4> i.e. the area between the pixels (w,y) and (0,y+1)
[22:13] <tr0m> llogan: any ideas on the (+aic) part
[22:13] <llogan> do you get a new error now?
[22:14] <tr0m> yah
[22:14] <wm4> it seems libswscale generally likes to stomp over some bytes past the last pixel of each scanline because of SIMD operations
[22:15] <tr0m> [msmpeg4 @ 0x89d3700] 4MV not supported by codec
[22:16] <durandal_1707> wm4: such bytes should be 0?
[22:16] <llogan> tr0m: remove it
[22:16] <tr0m> k
[22:16] <wm4> durandal_1707: it does write black to bytes it stomps over
[22:17] <wm4> durandal_1707: however, I want libswscale to write into a cropped image, without destroying the outside areas in the outer image
[22:17] <tr0m> [wmav2 @ 0x8517da0] output buffer size is too small
[22:17] <tr0m> audio encoding failed
[22:18] <durandal_1707> wm4: i doubt that is possilbe, because of simd stuff, but ask michaelni anyway
[22:18] <wm4> michaelni: see above
[22:19] <llogan> tr0m: i don't know. i've never encoded with msmpeg4/wmav2. use pastebin to show the command and the complete output.
[22:20] <tr0m> alright will inabit coffee & cig time
[22:20] <tr0m> thx
[22:23] <michaelni> wm4, i aggree it would be ideal if swscale didnt write into the alignment padding. a patch improving this is welcome if it doesnt slow thngs down
[22:49] <tr0m> llogan: http://pastebin.com/mQuE6epw
[22:50] <wm4> michaelni: I suppose ideally, libswscale would use the slow path for the last pixels in a scanline, but I certainly don't know swscale well enough to know if that's even possible
[22:53] <wm4> also, why does libswscale require a minimum image size?
[23:01] <mancha> hi folks. i am building mplayer svn at today with embedded ffmpeg git at today. i get lots of warning like this: ffmpeg/libavutil/libm.h:36:6: warning: "HAVE_ATANF" is not defined
[23:01] <mancha> any ideas?
[23:02] <mancha> seem i get 8 warning scroll by on most files (8 math functions, atanf atan2f powf, etc)
[23:03] <mancha> did a recent commit maybe mess up the headers a bit?
[23:08] <michaelni> wm4, honestly i dont remember, but you can remove the size check and debug the problems that occur with smaller sizes
[23:13] <mancha> i think 80521c1997a23e might be the culprit
[23:16] <llogan> burek: hows it going with the pi?
[23:20] <llogan> damn, tr0m left just as i had an answer for him
[23:20] <llogan> (probably bug 1104, try git master)
[00:00] --- Thu Oct 25 2012