[Ffmpeg-devel-irc] ffmpeg.log.20130215

[burek]

[00:10] <tds5016> so if I disable-all, what all would I need to enable for ffmpeg -i fs3 -c:v copy -f mpegts stream3.ts to work?
[00:10] <tds5016> ./configure --disable-yasm --disable-all seems not to work :-(
[00:12] <tds5016> correction: it works, but it doesn't output an ffmpeg binary anywhere :-(
[00:19] <ubitux> --disable-all is what it means&
[00:24] <tds5016> I'm now getitng osmething produced, but I'm missing something to make that command work...
[00:24] <tds5016> I keep getting invalid input.
[00:25] <ubitux> --enable-demuxer=mpegts --enable-protocol=file and stuff like that i guess
[00:26] <ubitux> (assuming you disabled everything)
[00:26] <ubitux> you'll likely need a few ones like this
[00:39] <stephanedev> hi everybody. i would like to stream a live video over rtp without any noticeable latency on a LAN. right now i have a latency ~1s.
[00:40] <stephanedev> i am already using some codec options to reduce latency
[00:41] <stephanedev> namely (("quality", "realtime"), ("lag-in-frames", "0")) for vp8 and ("tune", "zerolatency") for h264
[00:42] <stephanedev> this reduced the latency from a couple of seconds to around one second
[00:42] <stephanedev> even when using the loopback interface this latency happens
[00:44] <stephanedev> when streaming directly the output of the vp8 encoder without using rtp (and using my own code instead of letting ffmpeg lib handling the network bits), there is no noticeable latency
[00:45] <stephanedev> do you know of any other option in the ffmpeg C API to reduce this latency?
[00:46] <stephanedev> is it due to some internal buffering in the muxer or the demuxer?
[03:47] <lobotomius> hi ffmpeg!
[03:47] <lobotomius> how do i use this unsharpen filter
[03:47] <lobotomius> what are the params to sharpen the clip
[03:47] <lobotomius> ? tia
[06:33] <bigmac> i have a question about a movie collection and is there a way to convert them all to a smaller size streamable on my droid phone...
[06:33] <bigmac> maybe one command will work for all different formats, sizes and sizes?
[06:34] <bigmac> [700mb.mkv,  3gig.avi,  5gig.flv]
[06:34] <bigmac> those would be example source files
[06:36] <bigmac> src= mp4 avi mkv
[06:36] <bigmac> output= 300mb.flv
[06:36] <bigmac> or something
[06:38] <bigmac> or 300mb.swf
[06:46] <sacarasc> bigmac: Input matters relatively little.
[06:46] <sacarasc> Except for length, if you're wanting to have a specific size.
[06:49] <bigmac> i can batch script a directory but ultimatly i want to use a single ffmpeg command to get decent quility video i want to embed but download speed is my main concern
[06:50] <bigmac> maybe a example command if a embeddable output video?
[06:52] <bigmac> can i specify output file size? with width and hight?
[06:57] <bigmac> ffmpeg -i INPUT.mp4 -vcodec copy -acodec copy OUTPUT.flv
[06:58] <bigmac> thats same quality, im not sure how to loose file size
[07:01] <bigmac> -fs 3000
[07:20] <bigmac> i dont know what im doing lol
[07:21] <bigmac> how to reduce the output video size for better streaming
[10:20] <llogan> bigmac: see https://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide
[10:20] <llogan> two-pass if you're targeting specific output file size
[11:06] <barnseenio> relaxed: about?
[11:09] <relaxed> barnseenio: yes
[11:10] <barnseenio> relaxed: pm?
[16:17] <captainloc> Hi guys. Im broadcasting a live stream from one place to another and then i'm capturing this stream, cutting it into chunk. It's a mpeg-ts stream with H264 and AAC. I'm saving this stream locally, cutting it based upon a guide. It can appens that my live stream stop and restart but i keep recording  and appending in the same file. Then i transcode this file to MP4.  But randomly i get problems with Video : audio seems correct and plays but video stop and i
[16:22] <haole> hey there... I'm working in a digital TV application in which I need to send a private binary file over a MPEG stream... I need to develop a software to "break" my binary as MPEG sections with a possibly private type/descriptor/whatever... can ffmpeg help me achieving this?
[18:13] <alesan> hi
[18:13] <alesan> I am recording from three video sources at the same time and audio. So far I have captured the output of each camera, encoded in a different video file, and the audio in a .ogg
[18:14] <alesan> but it's a nightmare to synchronize everything
[18:14] <alesan> is there a way to record and sync right away multiple video sources? and maybe one or more audio sources too
[18:16] <klaxa> maybe synchronize recording devices clocks and add timestamps? :X
[18:17] <klaxa> *devices'
[18:21] <alesan> klaxa, the device is one and I am reading out three USB webcams
[18:22] <klaxa> are you using one ffmpeg instance or are you spawning multiple ones?
[18:22] <alesan> adding timestamps... but shouldn't the three videos be in the same file at that point? otherwise how can the timestamps relsato correctly to each other?
[18:22] <alesan> right now it's three different processes, each one records on a separate file completely indipendently
[18:23] <klaxa> you can put multiple video streams into one file and extract them for later processing
[18:23] <alesan> if they are in the same file, I do not want to extract them later :)
[18:23] <alesan> how do I put more than one video stream in the same file?
[18:23] <klaxa> they would be in the same file
[18:24] <klaxa> the matroska container for example supports multiple video streams
[18:24] <klaxa> you could extract them for post-editing
[18:24] <alesan> I know mkv supports more tracks, that would be awesome but... how to do that? I have searched for the past two days
[18:25] <alesan> I see no example at all, only joining several video files that already exist with mkvmerge
[18:26] <klaxa> ffmpeg -i <input 0> -i <input 1> -i <input 2> -map 0 -map 1 -map 2 everything_in_one_file.mkv
[18:26] <klaxa> i *think* that should work
[18:26] <klaxa> i never tried it myself
[18:27] <alesan> -map ? let me see what it does
[18:28] <klaxa> https://www.ffmpeg.org/ffmpeg.html#Main-options
[19:32] <Aktau> Hey guys
[19:33] <Aktau> I'm trying to convert an apple h.264 file (1080p) to a decent WebM file (libvpx and libvorbis)
[19:33] <Aktau> I can do that just fine with the libvpx-1080p.ffpreset
[19:33] <Aktau> But that sets the nitrate fixed
[19:33] <Aktau> In this case to 2M
[19:34] <Aktau> It looks nice and all but I would like to equal the nitrate of the original, which was a bit more than 16M
[19:34] <Aktau> Is there a nice way to tell ffmpeg to take the same nitrate?
[19:34] <Aktau> s/nitrate/bitrate/
[19:46] <Aktau> I guess I would like to specify quality and let the bitrate be determined by libvpx
[19:46] <Aktau> 't would be nice if anyone with experience with vp8 could enlighten me :) (using 1.1.2 on mac btw, freshly compiled)
[20:23] <john_____> hi, I'm using ffmpeg 0.11.2 with android 2.3.3, When I play h264 videos, there's this hiccup effect. With 0.10.8, everything was fine... Any idea ?
[20:24] <bahar> john_____ what do you mean by hiccup event?
[20:25] <john_____> bahar: the video plays normally, then suddenly, goes back few frames. Then proceeds forward, but jumps back few frame. and so on
[20:35] <llogan> john_____: does it occur with ffmpeg from git head?
[20:38] <john_____> llogan: tried to compile and run 1.0 but didn't succeed. We sticked to 0.x But to fix the deadlock in libestagefright, we need to use 0.11.1 or 0.11.2
[20:40] <llogan> development is active, and testing git head is always the first step in case the issue has already been resolved
[20:44] <john_____> llogan: ok, will try. Is it working with android 2.3.3 ? Does somebody has a recipe to build libstagefright with 1.x ?
[20:48] <peper03> Is there any way to get av_read_frame to return MPEG2 packets with 0x1bf (PRIVATE_STREAM_2) as start code?  I need them in sync with the normal packets to make improvements to DVD playback but mpegps_read_pes_header() actively filters them out.
[20:55] <rcombs> hey, all
[20:56] <rcombs> I'm trying to use FileSalvage and dd to recover a file
[20:56] <rcombs> it's an MOV
[20:56] <rcombs> I've been alternating between "invalid data" and "no MOOV atom found"
[20:57] <rcombs> in ffplay, which of those would come first?
[20:57] <rcombs> e.g. if I get "no MOOV atom", could it indicate that I'm not extracting enough data?
[21:00] <llogan> probably, especially if the moov atom is near the end of the file
[21:00] <rcombs> if the file was written by a DSLR, I'd expect the MOOV to be at the end, yeah?
[21:03] <llogan> i'm not sure. it's not something i'm really that familiar with
[21:04] <llogan> but an uneducated guess would be "yes".
[21:11] <rcombs> OK, which would indicate I've got the right start to the file: invalid data, or no MOOV?
[21:22] <qubitnik> hello!
[21:22] <Fjorgynn> cake?
[21:23] <qubitnik> thx!
[21:27] <qubitnik> I have an ffmpeg question (suprise, suprise), its about reusing outputs: I want one Single Input to be transcoded in multiple bitrates+resoultions (e.g. 720,480,240) but also pushed to other servers (e.g. rtmp), one quality should be saved to file, and the 3 qualities should be segmented for hls
[21:29] <qubitnik> I know how to do all the single Steps, but my Question is: What is the preferred way to provide each ffmpeg process with the already encoded streams, avoiding reencoding, and beeing able to use the streams any time not just at initial start
[21:34] <Mavrik_> qubitnik: ffmpeg really doesn't support this use case well
[21:34] <Mavrik_> (or at all) at the moment
[21:34] <qubitnik> hmm i made some experiments with a rtmp server in the middle
[21:34] <Mavrik_> qubitnik: if you have different resultions and bitrates you WILL have to reencode for each output
[21:34] <qubitnik> but i am not happy with that
[21:37] <qubitnik> yes i know, that is not the problem, but is there a way other ffmpeg processes can grab these outputs and use it again just copying audio+video and only doing remux for pushing,recording,segmenting?
[21:37] <llogan> maybe the tee muxer will be useful to you but i don't know if it will work with segment
[21:38] <Mavrik_> hmm
[21:38] <qubitnik> @llogan: yes, but it is a bit too static for me, using an rtmp server in the middle works, but adds another component+latency
[21:39] <Mavrik_> well another rather inefficient option is to just send UDP packets over localloop
[21:39] <Mavrik_> have one ffmpeg encode to MPEG2-TS and send to udp://127.0.0.1:<port>
[21:39] <Mavrik_> and other ffmpegs read from it and remux
[21:39] <Mavrik_> it'll add latency of course
[21:40] <Mavrik_> but for anything better you'll just have to pay for a "real" solution like Wowza :)
[21:40] <qubitnik> i also experimented with udp://127.0.0.1:5000 Streams and 1 Output is using them, and a second output is a copy to another port
[21:41] <qubitnik> i did so because this udp is point-2-point only one ffmpeg instance could use it, multicast is not an option in my scenario
[21:41] <Mavrik_> well as I said
[21:41] <Mavrik_> Wowza licenses aren't so expensive for pro usage
[21:42] <qubitnik> well, if you need one wowza license for each client, it is...
[21:43] <qubitnik> what i want to do tooks too much performance, its 1, maximum 2 liveshows per server
[21:43] <qubitnik> and they have a per server license
[21:43] <Mavrik> I had to develop my own solution based on libav* libs to do just what you want to
[21:44] <Mavrik-> everything else is just sub-par :\
[21:45] <qubitnik> yes using the libav* apis is the right way to go, but i am not good in c
[21:45] <qubitnik> more into java
[21:45] <Mavrik-> mhm, it's a problem then
[21:45] <qubitnik> yes
[21:45] <qubitnik> and libav* is not so well documented :)
[21:45] <Mavrik-> true ^^
[21:47] <qubitnik> what did you wrote with libav?
[22:01] <peper03> Is there a better place to ask about the PRIVATE_STREAM_2 issue?  I tried on the libav-user mailing list several days ago and got no response there either.  Just don't want to waste my (or anyone else's) time asking in the wrong place.
[22:02] <Mavrik_> well, you can ask
[22:02] <Mavrik_> but I doubt there's much people here that would know more than libav-user list
[22:07] <peper03> Ok.  Doesn't look hopeful, then.  I'm guessing there must be a reason for explicitly filtering those packets out but I'm blown if I can understand why.  Thanks anyway.
[22:47] <neXyon> hi
[22:48] <neXyon> what are the standard codecs (video, audio) for avi files? I know the container supports many codecs, but which ones are used and best supported from devices?
[22:49] <neXyon> mpeg4 + mp3? that's what ffmpeg uses by default
[22:51] <Mavrik_> something like that
[22:51] <Mavrik_> neXyon: what device are you targeting?
[22:51] <Mavrik_> usually choosing anything else than avi is a wise idea :)
[22:52] <neXyon> Mavrik_: yeah, the problem is, I don't know, that's why I'm gonna take my video in different formats and resolutions with me xD
[22:52] <Mavrik_> hmm
[22:52] <Mavrik_> mp4 with H.264/AAC is much more widely supported than anything in .avi really
[22:54] <neXyon> Mavrik_: yeah, that's the original format anyway
[23:15] <hi117> how do i extract a text stream from a video file?
[23:19] <hi117> er actually its the skeleton stream from a ogg
[23:23] <neXyon> ffmpeg -i short.mp4 -sameq -s 1280x720 short_hd.mp4 => gives an error, is that size not supported? o_O
[23:24] <Mavrik_> 1.) never use sameq
[23:24] <Mavrik_> 2.) what error?
[23:24] <neXyon> Mavrik_: what else? xD
[23:24] <neXyon> Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height
[23:25] <Mavrik_> look higher
[23:29] <neXyon> http://pastie.org/private/tuzgssgnsb8gc34qrypmg
[23:29] <neXyon> no clue why it writes none @ audio, and why not use sameq?
[23:30] <JEEB>  because sameq is "copy quants from source", and hasn't really worked even as such for years
[23:30] <sacarasc> -sameq uses the same quantizer as the source, not the same quality.
[23:30] <JEEB> yup
[23:30] <JEEB> also the reason why your line didn't work is written on line 31
[23:30] <neXyon> ah ok xD
[23:32] <neXyon> and how can I tell it to use the same quality?
[23:32] <JEEB> long story short, you can't
[23:32] <JEEB> you don't have thousands of chinese kids working in your computer
[23:32] <JEEB> the closest one has to such is libx264's -crf option
[23:32] <JEEB> which is the closest to "constant quality"
[23:32] <JEEB> default is 23
[23:33] <JEEB> higher is more compression, possibly worse quality, lower is less compression and possibly better quality
[23:33] <JEEB> the idea is to find the highest crf value that still looks good
[23:33] <neXyon> ah ok :) thanks
[23:33] <JEEB> which is the maximum amount of compression you can get while the video still looks good
[23:34] <JEEB> -c:v libx264 -crf 23 -preset medium <- these would be libx264's defaults
[23:34] <JEEB> preset is speed vs compression, while crf controls quality level
[23:34] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset <- all the presets listed
[23:34] <JEEB> so yeah, find the highest crf value that still looks good
[23:35] <JEEB> and then pick the slowest preset that is still fast enough for you
[23:35] <JEEB> and that's it :)
[23:35] <neXyon> k thanks :D
[23:35] <JEEB> you can use -ss and -t to cut 1-2 minute parts from the middle of a film or something
[23:36] <JEEB> for testing :)
[23:36] <JEEB> so you don't have to encode the whole thing
[23:40] <neXyon> JEEB: thanks, but I don't have enough time to do testing xD
[23:40] <neXyon> JEEB: have a very soon deadline :D
[00:00] --- Sat Feb 16 2013