[Ffmpeg-devel-irc] ffmpeg.log.20130806

Wed Aug 7 02:05:01 CEST 2013

[00:16] <roxlu> hi guys, I 'm trying to encode a png sequence but it fails with "pipe:: could not find codec parameters", when doing: cat dir/*.png | ffmpeg -f image2pipe -vcodec png -s 1280x720 -i - out.mp4
[00:16] <roxlu> Someone who knows why I get that error?
[00:18] <kevint_> How can I pipe in an audio stream (mp4 format) if I am already using stdin to pipe in video stream?
[00:20] <llogan> roxlu: also show the output of "ls dir/*.png"
[00:22] <llogan> kevint_: maybe you could try a named pipe
[00:22] <roxlu> llogan: https://gist.github.com/roxlu/76040ba166a155bcce85
[00:23] <mark4o> kevint_: or specify a file descriptor, e.g. pipe:3
[00:28] <roxlu> with jpegs it works. ... looks like  abug
[00:28] <roxlu> a bug
[00:28] <llogan> roxlu: you probably don't need cat: ffmpeg -i png/frames_recording/frame_%04.png output.mp4
[00:28] <llogan> also, worksforme. also, you never included your console output.
[00:28] <relaxed_> %04d
[00:29] <llogan> yes, thanks.
[00:29] <roxlu> llogan: yeah I need cat because I need to merge several dirs
[00:29] <roxlu>  using frame_%04.png works too
[00:29] <roxlu> it's just the combination image2pipe + png which doesn't work
[00:29] <relaxed_> it's never worked
[00:29] <roxlu> google gives me many other people mentioning it
[00:29] <roxlu> yeah
[00:29] <roxlu> : (
[00:30] <mark4o> roxlu: do you have your console output?
[00:30] <relaxed_> use `ln` to rename to a number sequence in one location.
[00:30] <roxlu> http://mplayerhq.hu/pipermail/ffmpeg-devel-irc/2012-December/001119.html  (search for image2pipe which give the same error)
[00:30] <roxlu> mark4o: Error while decoding stream #0:0
[00:31] <mark4o> :/ complete console output
[00:31] <llogan> i have no troubles using cat with png
[00:31] <relaxed_> I admit I haven't tried it in a while.
[00:32] <roxlu> mark4o: what is the filesize of your pngs?
[00:32] <roxlu> that might be related as I found in some posts
[00:32] <llogan> nobody can really help without your commands and console outputs
[00:33] <roxlu> full debug output: https://gist.github.com/roxlu/1327a657f396f4c439e4
[00:33] <llogan> that's not the complete console output
[00:34] <llogan> i want to confirm that you're actually using ffmpeg
[00:34] <mark4o> ...and the version number, and the configuration options
[00:34] <roxlu> ah I'm using avconv, but aren't those kinda identical ?
[00:35] <mark4o> lol
[00:35] <roxlu> btw, windows is cutting some of my selection :#
[00:35] <llogan> we don't serve their kind
[00:35] <llogan> (as in stuff from forks)
[00:36] <roxlu> :) I'm nobodies kind .. thought they were quite identical
[00:36] <llogan> use ffmpeg from FFmpeg or go to #libav
[00:36] <roxlu>  http://stackoverflow.com/a/9477756/1109017
[00:36] <roxlu> hmm don't you just love windows :)
[00:37] <llogan> http://ffmpeg.zeranoe.com/builds/
[00:44] <copious> hi all, I have a master file with 16 tracks of audio 1 track of video and I'm attempting to cut a sammple out of it. The sample cuts just fine, the problem is the channel position information in all the audio tracks is getting set to FC isntead of keeping the original information.
[00:45] <copious> I've attempted a bunch of different -map_metadata incatations and nothing seems to do the right thing.
[00:45] <llogan> copious: give the clients a mandelbrot instead: ffmpeg -f lavfi -i mandelbrot -t 10 output.mp4
[00:46] <copious> will do... be back shortly
[01:49] <copious> pebkac error apologies
[03:31] <Datalink> okay ffmpeg just finished compiling on the studio's Pi, http://illogicallabs.com/paste/00000003.txt now how much of this do I have to change if I switch from avconv to ffmpeg?
[03:34] <sacarasc> You can remove -vcodec libx264, as you've already declared it on the line before. Other than that, just change avconv to ffmpeg and I think it should work.
[03:36] <Datalink> sacarasc, it's throwing an error
[03:36] <Datalink> gimme a sec, updating the text file to show the updated code and error
[03:39] <Datalink> http://illogicallabs.com/paste/00000004.txt
[03:39] <Datalink> ugh, one of these times I'm afraid I'm gonna end up giving the stream credentials trying to get this working
[03:40] <sacarasc> Hmm, that is an odd error indeed.
[03:40] <Datalink> ah, found the option giving the error, -preset fast is right after the libx264, I'll remove that and try again
[03:40] <Datalink> ....
[03:41] <Datalink> ugh, it seems to not have x11grab now >.<
[03:49] <Datalink> oh ok, it appears that x11grab isn't enabled by default in the compiler
[03:50] <sacarasc> :(
[03:51] <Datalink> it's okay, I'm taking advantage of the recompile to enable the code from additional licenses including nonfree code, this is internal, and not for a distro, so I can safely use nonfree code
[04:56] <b6> i have a bunch of 16-bit grayscale images to encode as video. is there such a thing as a grayscale video? using a certain api based on ffmpeg, opening a video stream with pixel type gray16le is failing.
[05:47] <sacarasc> Datalink: How's it going?
[05:48] <Datalink> sacarasc, finished dephell to use the h264 library and x11grab, now it's compiling, this could take another 2 hours, I figure, since I'm compiling native on the Pi
[05:48] <sacarasc> Ah.
[05:49] <Datalink> I'll have to fix a known bug with the production scripts in this Pi, copy it's root partition, update it, and swap another pi for it when (if) I get this working
[05:50] <sacarasc> Wow.
[05:50] <sacarasc> That's a lot of hassle.
[05:51] <Datalink> yeah, which is why I'm annoyed libav has been so sloppy with various updates and also annoyed that the Debian branch includes one of those devs
[05:51] <Datalink> aka, having to do this
[05:51] <Datalink> after this, I have to figure out how to handle a streamname
[05:58] <Datalink> cool, it's in the h264 section
[05:58] <Datalink> for libavformat at least... this'll take... another couple of hours
[07:41] <Meuep> color me a noob, but where might I find the default location of the ffserver log?
[07:58] <Datalink> uh... /var/log?
[08:08] <Meuep> I think the one I was looking for was smack-dab in the home directory
[08:08] <Meuep> (I'm assuming just where you execute from?)
[08:10] <Meuep> I'm attempting to write an ffm file to an ffserver feed, but ffserver doesn't like what I'm doing, as I keep getting this: http://pastebin.com/kQLU1Uxw
[08:33] <t4nk962> hi all
[08:33] <t4nk962> the question about capture video + audio from USB webCam by ffmpeg
[08:34] <t4nk962> VLC can not play the stream , it says "PTS is out of range"
[08:34] <t4nk962> anyone have idea about "PTS is out of range"
[08:34] <t4nk962> the command I use to capture "ffmpeg -loglevel debug -f video4linux2 -r 30 -s 640x480 -input_format h264 -i /dev/video1 -f alsa -ar 48000 -ac 2 -i hw:0 -vcodec copy -acodec copy  http://localhost:8090/feed1.ffm "
[13:36] <Martijnvdc> hello. i am decoding an audio stream from a microphone, but the decoded frames just output junk. When reading raw packets, they contain PCM audio. But when i decode it, i just get junk
[13:45] <Martijnvdc> it's not a command line, it's coded in C
[13:45] <klaxa> ah
[13:46] <klaxa> well PCM is already decoded, it's raw audio you don't have to decode it, you have to set samplerate and bit depth to properly handle it though
[14:11] <Datalink> welp, compiled overnight, but still not working right: http://illogicallabs.com/paste/00000005.txt I think I'm down to my original issue in that there's a streamname that I don't know how to pass to ffmpeg
[14:12] <GoaLitiuM> Datalink: you had problems with ustream and ffmpeg?
[14:13] <Datalink> GoaLitiuM, rtmp stream, actually, but yeah
[14:14] <GoaLitiuM> adding flashver=FMLE/3.0\20(compatible;\20FMSc\201.0) right after the rtmp url made ustream streaming working for me
[14:18] <Datalink> this isn't ustream, and that didn't change the error :/
[14:19] <Datalink> I have a stream name which isn't in the code there, but is part of the authentication and I'm not sure how to add it
[14:23] <Datalink> oh hey, it's an adobe server, that'll help figure this out
[14:24] <Datalink> [rtmp @ 0x2d417e0] Server error: [ AccessManager.Reject ] : [ code=403 need auth; authmod=adobe ]
[14:30] <Datalink> yeah, this is getting frustrating but at least rtmp is throwing the error more usefully
[14:30] <Datalink> GoaLitiuM, no difference between flashver and no flashver
[14:30] <Datalink> since this isn't actually ustream, but an RTMP with similar auth credentials
[14:41] <Datalink> http://illogicallabs.com/paste/00000005.txt it seems to not even care about the flashver
[15:13] <Anarhist> hi, is it possible to drop all the even lines. i want to deinterlace, but not by combining them but rather by dropping them (i'm doing resize anyhow, so there'll be less quality loss)
[15:14] <Anarhist> i don't want to do an extra step in AVIdemux just to resize
[15:27] <vad_> Is there some option in ffmpeg to speed up a video by extracting one frame every 1 sec and using that to compose a new file?
[15:29] <Anarhist> vad_, in libav there's 'select' filter, i'm unsure if it is also in ffmpeg
[15:31] <vad_> "decimate" also looks interesting
[15:34] <vad_> as does "framestep"
[15:44] <Datalink> http://illogicallabs.com/paste/00000005.txt I'm looking at what I think is an adobe based stream server, I've ben supplied with a user, a stream name, the URL and a password, how would I enter the stream name?
[16:32] <Martijnvdc> hello, i'm trying to set up a decoder for a microphone, but after decoding, the data is corrupt. The raw packets from the audio device do contain valid PCM data though.
[16:32] <Martijnvdc> http://pastebin.com/T17gjUX6
[16:33] <Martijnvdc> would this be a bug, or am i doing something wrong?
[17:54] <tmkt> Using Mediainfo i have a piece of Media that says Mpeg AUDIO 2 Layer 3....but ffmpeg is crapping out on the file with Stream #0:1: Audio: none, 8000 Hz,
[17:54] <tmkt> and then Encoder (codec none) not found for output stream #0:1
[17:54] <tmkt> any ideas?
[17:57] <vad_> tmkt: and what do media players do with it? :)
[17:57] <tmkt> nm found the issue
[17:57] <tmkt> http://www.dilella.org/unsupported_codec_ffmpeg/
[17:57] <tmkt> solved it
[18:00] <tmkt> the last line
[18:00] <tmkt> extra-53
[18:55] <viric> hi!
[18:56] <viric> I've a video "720x576 [SAR 16:11 DAR 20:11]"
[18:56] <viric> If I scale it to a different resolution, do SAR and DAR keep the same?
[18:57] <viric> I don't really understand the difference between sar and dar
[19:03] <viric> So, getting it to squared pixels 16:9 would be nice.
[19:13] <Martijnvdc> hello, i'm trying to set up a decoder for a microphone, but after decoding, the data is corrupt. The raw packets from the audio device do contain valid PCM data though.
[19:13] <Martijnvdc> http://pastebin.com/T17gjUX6
[19:13] <Martijnvdc> would this be a bug, or am i doing something wrong?
[19:15] <viric> ok got it about dar/sar
[19:22] <vad_> Martijnvdc: what kind of PCM
[19:24] <Martijnvdc> 16le
[19:24] <Martijnvdc> the resulting decoded packet should also contain PCM, but it just contains junk
[19:25] <Martijnvdc> it's at 48000 samples per sec
[19:28] <vad_> if it's already LPCM (assuming it's LPCM), why would you need a decoder, or are you speaking of an ffmpeg decoder?
[19:30] <Martijnvdc> i'm decoding using an ffmpeg decoder, yes
[19:31] <Martijnvdc> well, the data shouldn't be corrupt after decoding it
[19:43] <viric> mh I'm playing with overlay... how can I make so when the shortest video finishes, it's not overlayed anymore? It looks like it stays overlayed forever
[19:59] <vad_> Anarhist: ffmpeg 1.0.6 has "select", but it always errors out when trying one of the examples :/
[19:59] <vad_> jfyi
[20:10] <viric> humm using "-ss", affects "movie=" sources in the vf chain too.
[20:10] <viric> annoying
[20:10] <viric> I want only the input skipping some seconds, not the vf movie= source
[20:45] <AndrzejL> damn it :D
[20:45] <AndrzejL> I always fall for the same stuff
[20:45] <AndrzejL> x264 not found
[20:45] <AndrzejL> Damn it
[20:45] <AndrzejL> Why is the aur package for aur created for x264 without the enable shared option DUH
[20:45] <AndrzejL> :D
[20:46] <AndrzejL> hehe it took me 10 mins to figure out... 10 minutes I will never get back :D
[21:39] <plm> Hi all
[21:41] <plm> people, how I do to receive video via rtsp in ffmpeg with auth? In VLC I do: "vlc rtsp://test1:p1234@192.168.0.30"
[22:08] <Martijnvdc> hello, i'm trying to set up a decoder for a microphone, but after decoding, the data is corrupt. The raw packets from the audio device do contain valid PCM data though.
[22:08] <Martijnvdc> http://pastebin.com/T17gjUX6
[22:08] <Martijnvdc> would this be a bug, or am i doing something wrong?
[22:08] <sacarasc> Would you need to decode it? Isn't it already decoded?
[22:09] <Martijnvdc> i can't put an AVPacket into an encoder after that
[22:09] <Martijnvdc> basically, i just want to encode an audio device with the opus codec
[22:10] <vad_> arecord -f - | ffmpeg -i - test.ogg  (for oggvorbis, for example)
[22:11] <vad_> just select the proper options for S16LE and Opus
[22:11] <Martijnvdc> yeah, but i'm doing it in C, it's for a VOIP application
[22:12] <Martijnvdc> video works perfectly, but i have encountered a very strange bug with audio
[22:13] <Martijnvdc> see, the code in that pastebin causes the decoder to have a corrupted output, but i don't know why
[22:13] <Martijnvdc> so i'm thinking it might be a bug...
[22:14] <vad_> if you want to produce Opus from LPCM, would you not need an *encoder* rather than a decoder?
[22:15] <Martijnvdc> how can i encode an AVPacket? wouldn't i need an AVFrame for that?
[22:15] <vad_> possibly (I don't know, I am just looking at this from a high level)
[22:16] <Martijnvdc> the packet does hold the raw PCM data, which i want to encode directly; but ffmpeg doesn't allow that, so i decode the PCM stream into an AVFrame
[22:17] <Martijnvdc> but the AVFrame holds corrupt data; which is very strange behavior
[22:18] <vad_> what if you feed some framed data (say oggvorbis) into the decoder, does the AVFrame then contain something useful? (either vorbis again or the decoded LPCM)
[22:20] <Martijnvdc> the data which i'm feeding the decoder is valid data. the resulting AVFrame never holds anything useful, no matter what i feed to it
[23:16] <mark4o> viric: -shortest, and setpts filter
[23:25] <viric> mark4o: hm I don't want to cut at shortest
[23:25] <viric> I used fade, at the end, + qtrle container for the overlayed movie
[23:28] <mark4o> viric: oic what you mean; you can also use enable option on overlay filter, e.g. overlay=...:enable=between(t\,100\,200) to enable between 100s and 200s only
[23:31] <mark4o> or if you want to fade you don't need to use a container with alpha, you can instead just add the alpha with ffmpeg if you prefer, e.g. movie=...,format=yuva420p:fade=...:alpha=1
[00:00] --- Wed Aug  7 2013