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December 2013
- 1 participants
- 54 discussions
[01:33] <cone-414> ffmpeg.git 03Michael Niedermayer 07master:b3f44eafa5a8: avcodec/iff: warn about truncated input to decode_byterun() and clear remaining output
[01:33] <cone-414> ffmpeg.git 03Michael Niedermayer 07master:8fe06e7ae8ff: avformat/nistspheredec: check sscanf() success before using the result
[01:33] <cone-414> ffmpeg.git 03Jan Gerber 07master:200cfab8c4c2: remuxing .opus files to .webm codec->delay must be copied too
[01:33] <cone-414> ffmpeg.git 03Michael Niedermayer 07master:a165964f3c4b: avformat/sierravmd: Check avio_read return value
[13:11] <cone-64> ffmpeg.git 03Diego Biurrun 07master:8558595a5991: configure: Express atomics/thread deps through the dependency system
[13:11] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:3b6e8634c3be: Merge commit '8558595a59913a4667f57d5a4658b1340f1a4354'
[13:18] <cone-64> ffmpeg.git 03Diego Biurrun 07master:e1b9de4fe15c: atomics: cosmetics: Restructure ifdefs for greater clarity
[13:18] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:3f307d79d383: Merge remote-tracking branch 'qatar/master'
[15:32] <ubitux> i have a few h264 stream with 32 reference frames
[15:32] <ubitux> does this sound legit?
[15:32] <ubitux> (ffmpeg has a limit to 16)
[15:33] <Compn> ffmpeg has some bullshit limits
[15:33] <Compn> benchmark when changing limits to make sure it doesnt affect speed and then up it :)
[15:33] <Compn> i'm guessing it affects speed
[15:34] <ubitux> the code might not work at all with >16
[15:34] <Compn> only one way to find out
[15:34] <ubitux> it actually doesn't
[15:35] <Compn> oh ok
[15:35] <Compn> some other software plays such streams ?
[15:35] <Compn> quicktime ?
[15:36] <ubitux> doesn't seem so
[15:36] <kierank> the limit of 16 is in the spec
[15:36] <kierank> but i think there is no bitstream restriction
[15:36] <kierank> ubitux: are you sure it is not 32 reference fields
[15:37] <ubitux> i'm hiting the "too many references" message
[15:37] <ubitux> and sps->ref_frame_count is 32
[15:37] <kierank> where is this stream from
[15:37] <ubitux> encoded with an iphone i'd say but not sure
[15:38] <Compn> does it play and just throws messages ?
[15:38] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:241eccd62898: avcodec/mpegvideo_enc: fix integer overflow with -skip_exp >= 2
[15:38] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:34fe125f4bc8: avcodec/mpegvideo_enc: fix frame skipping with intra only codecs
[15:39] <ubitux> Compn: it can not be played; if i naively change the limit it decodes shit
[15:40] <Compn> so its either custom thing requiring a binary codec or its broken
[15:58] <ubitux> well probably a bug in the ios encoder
[15:58] <ubitux> can't find anything though
[16:11] <Compn> ubitux : did he use some 3rd party video capture app ?
[16:11] <Compn> did he rip it off his iphone with a 3rd party app ?
[16:43] <ubitux> Compn: 3rd party iphone app using the hw h264 encoder
[16:45] <Compn> yeah , that 3rd party app is doing it wrong
[16:46] <Compn> might be able to make a bsf to make the stream conform
[16:46] <Compn> if its not totally borken
[16:46] <ubitux> how can you be sure it's the 3rd party app?
[16:46] <ubitux> most of the video out of it are playable
[16:47] <ubitux> but once in a while, the encoder shits an unplayable brick
[16:48] <Compn> what country is 3rd party app made in ?
[16:48] <ubitux> not india
[16:48] <Compn> china ?
[16:48] <ubitux> no
[16:48] <Compn> how can i be sure? because iphones have millions of users , and they dont shit bricks
[16:48] <Compn> but 3rd party does
[16:49] <ubitux> well, some particular parameters might trigger the issue
[16:49] <ubitux> dunno
[16:50] <Compn> always blame the user
[17:13] <ubitux> the checks makes no sense btw
[17:13] <ubitux> if (sps->ref_frame_count > MAX_PICTURE_COUNT - 2 ||
[17:13] <ubitux> sps->ref_frame_count > 16U) {
[18:21] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:632fdec9f4eb: avformat/nistspheredec: initialize header_size to -1
[18:52] <saste> what happened to hue=s=0?
[18:52] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:b07a5e9b6be9: avformat/smacker: check for format mismatch more completely
[19:26] <cone-64> ffmpeg.git 03Stefano Sabatini 07master:70e74e007270: lavfi/hue: force table initialization for the first frame
[19:26] <cone-64> ffmpeg.git 03Stefano Sabatini 07master:028d8dd367f7: lavfi/hue: use av_clip_uint8() instead of av_clip_uint8_c()
[19:26] <cone-64> ffmpeg.git 03Stefano Sabatini 07master:d055a1395c23: lavfi/hue: show first decimal value of saturation
[23:11] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:5ec3c7b7c118: avformat/pva: Make sure the first byte of pes_header_data has been initialized
[23:11] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:eedd9148733f: avformat/pva: Make sure the header is large enough before reading the timestamp from it
[23:34] <cone-64> ffmpeg.git 03Mason Carter 07master:832e19063209: vc1: arm: Add NEON assembly
[23:34] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:69278d94c482: Merge commit '832e19063209a5f355af733d1a45f5051f49ce33'
[23:42] <cone-64> ffmpeg.git 03Mason Carter 07master:b254490bdabb: vc1: arm: Add NEON no_rnd chroma MC
[23:42] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:4fa91b88c618: Merge commit 'b254490bdabb21bd517c05b1a68717f9952ac8c4'
[23:51] <cone-64> ffmpeg.git 03Anton Khirnov 07master:dfc50ac85e9d: x86: mpegvideo: move denoise_dct asm to mpegvideoenc
[23:51] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:ce612fc18638: Merge commit 'dfc50ac85e9d68a771b556297b7c411650206f3b'
[23:57] <cone-64> ffmpeg.git 03Luca Barbato 07master:027712e851da: jvdec: Return EOF on end of file
[23:57] <cone-64> ffmpeg.git 03Michael Niedermayer 07master:d2344afb90db: Merge commit '027712e851da4d124a842c9e2802f95d50582553'
[00:00] --- Sat Dec 21 2013
1
0
[01:00] <pyBlob> I'm using ffmpeg to convert a video to raw yuv444p ... but it doesn't seem that the yuv-values are stored interleaved, like in rgb24
[01:00] <pyBlob> > http://www.pasteall.org/pic/show.php?id=64246
[01:00] <pyBlob> so what is the correct way to read the raw data?
[01:40] <Rivky> Is it possible to copy a given chapter's audio using the chapter's identifier? The docs only show it being done with -ss and -t and wants me to subtract the end time from the start time, and that just feels so silly.
[06:06] <clever> are there any test pattern video files, that have things like frame number embeded into each frame?
[06:06] <relaxed> you can create your own with the drawtext filter and a little scripting
[06:10] <clever> dont think mplayer has the same filter in it though, trying to create a vo module for that
[06:12] <Compn> hmmm
[06:12] <Compn> does ffmpeg have it ?
[06:12] <Compn> you might be able to enable mplayer libavfilter support
[06:12] <Compn> dunno if it works tho :)
[06:12] <clever> though i could maybe use ffmpeg in transcode mode to apply the filter to a new file, and then play that back
[06:13] <clever> but it might be simpler to just hard-code it to draw directly into the output buffer now that i think of it
[06:15] <Compn> sure either way
[06:16] <clever> ive got the bulk of the omx decode working, just some timeout problems now
[06:16] <clever> so that only leaves the output part, fbdev and x11 just plain suck
[07:47] <clever> for x in {0..300};do convert shot0001.png -background transparent -fill black -pointsize 40 caption:$x -gravity center -composite frame$x.png; done
[07:47] <clever> ah dang, Compn is gone
[07:48] <ubitux> why did you ignored relaxed suggestion?
[07:48] <ubitux> you can do it in one single ffmpeg command
[07:49] <clever> using mplayer not ffmpeg, didnt think to keep going down that route
[07:49] <ubitux> don't you just want to generate a video with numbered frames?
[07:49] <clever> yeah
[07:49] <clever> ffmpeg -f image2 -i frame%d.png -r 30 -q:v 2 sample.mp4
[07:49] <clever> this second step did that
[07:49] <ubitux> you can generate the numbered frame with ffmpeg as well.
[07:50] <clever> yeah, i could have probly used the video filters now that i think of it
[07:50] <ubitux> that was exactly what relaxed said&
[07:50] <clever> at the time, i thought he was saying to use them on ffplay
[07:54] <ubitux> ffmpeg -f lavfi -i color -vf "drawtext=text=%{n}:fontcolor=white:fontsize=70:x=20:y=20" -t 5 num.mp4
[07:54] <ubitux> ofc, ffmpeg built with --enable-libfreetype --enable-fontconfig
[07:55] <clever> looks like i turned it off, [AVFilterGraph @ 0x89caf20] No such filter: 'drawtext'
[07:56] <ubitux> you didn't --enable-libfreetype
[07:56] <ubitux> on which drawtext relies
[07:57] <clever> which seems to be the truetype flag in gentoo
[07:57] <ubitux> --eanble-fontconfig isn't mandatory but useful to avoid the need of specifying font path
[08:23] <clever> ubitux: yep, that also works nicely
[08:25] <clever> was going to ask about color, then i noticed fontcolor=white
[08:31] <clever> now to figure out why the file i made with image magick causes a segfault
[08:31] <zap0> is white 0xffffff ?
[08:32] <zap0> and is that value in-bounds for your colour space
[08:32] <clever> yuv420
[08:32] <clever> it sefaults when copying the v plane
[08:33] <clever> allocated 783360 bytes for a 960x540 image, padded to 960x544 buffer runs from 0xb3d88020 to 0xb3e47420
[08:33] <clever> draw slice 0 960/16 0/272
[08:33] <clever> copying 0xb3d77520(480) to 0xb3e47420(480), height 8
[08:33] <clever> at a glance, it appears to be using the end of the entire frame as a starting position
[08:34] <clever> so something is wrong with the plane pointers
[08:38] <clever> ah, i forgot to halve the y offset for u and v
[08:38] <clever> since u and v are 2x2 subsamples
[08:41] <clever> zap0: yep that was it, segfaults are gone
[08:42] <clever> still nothing but green out of it though
[08:45] <zap0> rgb(0,0,0) == yuv(green)
[08:47] <clever> only the y plane is working
[08:47] <clever> so i'm getting everything from dark green to light green, and all the shades between
[08:48] <clever> http://gallery.earthtools.ca/index.py/paste/DSCF0558.JPG would be a normal color sample image
[08:49] <zap0> who is doing the yuv/rgb conversion? it looks wrong
[08:49] <clever> the gpu hardware
[08:50] <clever> which is poorly documented
[08:50] <clever> the forums said the stride must be a multiple of 16, but i just found out today that 16 causes massive problems
[08:50] <zap0> ah.. blending edge hackery.. good luck.
[08:50] <clever> 32 works fine
[08:50] <clever> ive gotten the hardware h264 decode to work, and its returning yuv420 frames now
[08:51] <clever> but fbdev, x11, and sdl all do software rgb conversion
[08:51] <clever> so the cpu usage is too high
[09:08] <clever> zap0: bingo!, the out of order frames are caused by the decode engine
[09:08] <clever> its nothing to do with my double buffering at the output
[09:09] <clever> software decode is slow and perfect, hardware decode is completely out of order
[09:29] <Barvinok> hi all. I'm grabbing video from v4l2 : "ffmpeg -f v4l2 -r 30 -s 640x480 -i /dev/video0 ctest.mkv" but it encodes with about 10 fps. Ok then, I set "-r 10" but then it encodes with 5 fps :( how do I enforce input fps to match output fps ?
[09:30] <Barvinok> when I set 5, it degrades to 2.8 and to half what I set in general
[10:24] <pyBlob> with the blur filter you can blur a video in x-y-axis, is there also a temporal blur filter that overlays the video with previous frames?
[10:50] <DelphiWorld> hi
[10:51] <DelphiWorld> can someone help me converting a rtmp stream to rtp stream?
[10:52] <DelphiWorld> rtmp://radio.gmo.ps/live/livestream
[10:52] <DelphiWorld> unable to write stream 0...
[10:55] <DelphiWorld> i'm doing:
[10:55] <DelphiWorld> ffmpeg -i $stream -acodec copy -vcodec copy -f rtp rtp://localhost:6666
[10:55] <DelphiWorld> but is unable to work
[11:10] <DelphiWorld> hey odedo
[11:12] <bencoh> "-f rtp" is wrong
[11:13] <bencoh> (well it depends on how you want to mux it, but ...)
[11:13] <DelphiWorld> bencoh, want to kick out the video. keep the audio, AAC. that's all, rtp
[11:15] <bencoh> try -f mpegts
[11:15] <DelphiWorld> and use rtp:// ?
[11:16] <DelphiWorld> and how to kick the video?
[11:17] <bencoh> oh right, you want audio only ... "-vcodec copy" is definitely not gonna work ;)
[11:18] <DelphiWorld> then how to do audio only:)
[11:18] <bencoh> -vn should help
[11:20] <DelphiWorld> hey bencoh you rocks ;)
[11:22] <DelphiWorld> shit, need sdp file? mmmmmm
[11:22] <DelphiWorld> i'm not gonna use rtp, sory:)
[11:23] <DelphiWorld> bencoh, this is rtmp stream that couldnt be read by mpd, so wanna pass it to mpd using ffmpeg...
[11:23] <DelphiWorld> rtp need sdp file
[11:26] <bencoh> you don't always need rtp, depending on how you use it
[11:26] <bencoh> s/rtp/sdp/
[11:27] <DelphiWorld> lol bencoh ;-)
[11:27] <DelphiWorld> give me a sdp less solution:)
[11:29] <bencoh> you could send raw s16 to your mpd
[11:29] <bencoh> (raw s16 in rtp)
[11:31] <bencoh> read rfc 3551 :p
[11:33] <DelphiWorld> lol bencoh i'm not a developer ... i'm just a person trying to listen
[11:34] <bencoh> :)
[11:34] <bencoh> something like -acodec pcm_s16be -f rtp might work
[11:35] <bencoh> (or -f s16be, dunno)
[11:37] <bencoh> works with -f rtp
[11:38] <DelphiWorld> lol bencoh... ffprobe rtp://localhost:6666 dont give any audio codec:)
[11:38] <DelphiWorld> and complain about sdp:)
[11:38] <DelphiWorld> -acodec pcm_s16be -f rtp rtp://localhost:6666
[11:39] <bencoh> -vn -acodec pcm_s16be -f rtp ?
[11:39] <bencoh> works here
[11:39] <DelphiWorld> yeah, ffprobe complain about sdp and payload 97
[11:40] <bencoh> on the output stream ?
[11:42] <bencoh> dunno why ... vlc plays it just fine, and it's compliant to rfc 3551
[11:42] <DelphiWorld> bencoh: yeah, ffprobe refuse it
[11:49] <DelphiWorld> a=rtpmap:97 L16/22050/1
[11:49] <DelphiWorld> this is playable in ffplay
[11:53] <DelphiWorld> bencoh: i think, i'lle try to have it on http.
[11:54] <DelphiWorld> mmmmm!
[11:54] <DelphiWorld> got a good idea
[11:54] <bencoh> you could transcode to mp2 over rtp if you're that desperate
[11:55] <DelphiWorld> bencoh: lol hold on
[11:55] <bencoh> I tink this one does not need any sdp
[12:00] <DelphiWorld> didnt work LOL
[12:00] <DelphiWorld> i pushed the stream to a rtmpd
[12:00] <DelphiWorld> nginx rtmp module
[12:00] <DelphiWorld> and outputed it in hls
[12:00] <DelphiWorld> but mpd didnt read it
[12:18] <DelphiWorld> ok bencoh
[12:18] <DelphiWorld> i simplified my setup using nginx
[12:18] <DelphiWorld> ffmpeg push to nginx using rtmp & mpd fetch from nginx
[12:22] <bencoh> ohmy
[12:24] <DelphiWorld> lol
[12:24] <DelphiWorld> what else... didnt find a solution;)
[12:27] <DelphiWorld> my mpd is not decoding aac...
[13:27] <nyuszika7h> hi, I converted a MKV file to MPG so that my DVD player can play it, but I'm getting frame sync errors when pressing the arrow keys in ffplay to go back/forward, and in VLC and my DVD player, when I go to 20:00, it goes to 17:15 instead
[13:28] <nyuszika7h> http://ix.io/9sJ
[13:32] <nyuszika7h> the command I used:
[13:32] <nyuszika7h> ffmpeg -threads 2 -i 01\ -\ The\ Impossible\ Astronaut.mkv -target pal-dvd -q:a 0
[13:32] <nyuszika7h> er
[13:32] <nyuszika7h> ffmpeg -threads 2 -i 01\ -\ The\ Impossible\ Astronaut.mkv -target pal-dvd -q:a 0 -q:v 0 /run/media/nyuszika7h/STORE1/01\ -\ The\ Impossible\ Astronaut.mpg
[14:13] <relaxed> -q:v 0 isn't valid
[15:52] <nyuszika7h> relaxed: it isn't?
[15:55] <nyuszika7h> anyway, apparently ffmpeg is dropping frames
[15:55] <nyuszika7h> more than 1000
[19:36] <PowerCC> hello
[19:37] <PowerCC> trying to extract dvdsub > srt
[19:37] <PowerCC> ffmpeg -i moviefile -vn -an -codec:s:0.2 srt moviefile.srt
[19:38] <PowerCC> however getting: ffmpeg -i moviefile -vn -an -codec:s:0.2 srt moviefile.srt
[19:38] <PowerCC> error while opening encoder for output
[19:38] <PowerCC> i also tried to -map s:
[19:45] <JEEB> PowerCC, dvd subtitles are pictures
[19:45] <JEEB> srt is text
[19:45] <PowerCC> not possible?
[19:46] <PowerCC> must use mkvtools
[19:46] <PowerCC> in another word OCR
[19:49] <PowerCC> thx, it looks like ffmpeg can only extract subrip or other texts subtitles dvdsub is pictures that's the problem.
[19:50] <JEEB> PowerCC, use subtitle edit and such apps to OCR it first
[19:51] <PowerCC> thanks jeeb, i am getting old totally fogot dvdsub are pictures
[19:51] <JEEB> np
[19:53] <JEEB> dvb subtitles, dvd subtitles and pgs (presentation graphics, the naming used for blu-ray picture-based subtitles) are at least picture-based :)
[19:53] <clever> JEEB: do you happen to know how ffmpeg handles a decoder returning frames out of order?
[19:54] <clever> 30 37 36 37 35 41 for example
[19:56] <JEEB> I thought a libavcodec decoder should always output pictures in order? Also if you have the same PTS coming out multiple times you're most likely screwed already :P
[19:56] <PowerCC> JEEB yeah I remember using tools to convert it from dvdsub to srt long ago... been awhile.
[19:56] <clever> JEEB: ah, so its my job to fix that within my decoder
[19:56] <PowerCC> thx jeeb
[19:57] <clever> JEEB: the strange thing, is that i'm printing the pts out from the code and its in order, but the frames i see arent
[20:32] <alesan> re
[20:32] <alesan> how to calculate PSNR ?
[23:38] <FedoraUser> hello friends
[23:39] <FedoraUser> i am getting some strange glitches while trying to record from x11grab, it shows something like scan lines, where you can see the parts of window beneath the one i am recording
[23:39] <FedoraUser> any hints how to fix that?
[23:40] <FedoraUser> or how to properly test that
[23:57] <ph8> hi all! I have a 4:3 clip that i want to make a 768x432 thumbnail of (16:9) - i don't want to stretch it but i want black bars. Any idea on how i do this? At the moment i'm doing -filter:v scale="768:432" but that is stretching
[23:57] <ph8> i feel like a pad's in my future but i really don't get the syntax
[00:00] --- Sat Dec 21 2013
1
0
[00:47] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:947e40b9feb8: avformat/ipmovie: Check that OPCODE_SET_PALETTE size is large enough
[00:47] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:4e575adeff31: avformat/ipmovie: Check palette size in OPCODE_SET_PALETTE
[00:47] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:ff1e30c05938: avcodec/interplayvideo: Check that enough data is available for opcode_0xA
[00:47] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:66875798eb88: avcodec/interplayvideo: Check that enough data is available for opcode_0x9
[00:47] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:8eb76217d013: avcodec/interplayvideo: Check that enough data is available for opcode_0x7/8/D
[03:29] <cone-433> ffmpeg.git 03Martin Storsjö 07master:e121a725c38f: fate: Explicitly specify the pixel format for the pngparser test
[03:29] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:fde74d1b9d33: avformat/swfdec: check avio_read() return code
[03:29] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:d2d794f3aa55: Merge remote-tracking branch 'qatar/master'
[05:10] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:e72f5abbc62d: avformat/swfdec: check avio_read() return code
[05:10] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:f5d039840aca: avformat/swfdec: clear 4 bytes at the end of a packet if they are not initialized
[14:01] <cone-645> ffmpeg.git 03Lenny Wang 07master:d943330dc01f: lavfi/deshake_opencl: optimze transform filter
[14:34] <j-b> XvMC? Seriously?
[15:18] <Compn> j-b : we dont all have nvidia :P
[15:37] <vivienschilis> guys, any updates on supporting CLEF/TAPT atoms?
[15:57] <ubitux> oh, work on dvb sub
[16:49] <cone-920> ffmpeg.git 03Michael Niedermayer 07master:a33c7dd21362: avcodec/flashsv: check decode_hybrid() for overread
[17:11] <dorileo> michaelni: in your review do you mean a s/packet_size/packetsize/ ? or your comment implies something else?
[17:12] <michaelni> i was thiking of ts_packetsize but maybe theres a better name
[17:13] <dorileo> michaelni: if we want something specific and clearly related to mpegts ts_packetsize or ts_packet_size look good, but like your said ts_packetsize is more friendly in this case.
[17:19] <michaelni> i just want to avoid that there is ambiguity with the existing AVFormatContext.packet_size
[17:19] <michaelni> otherwise packet_size would have been fine
[17:20] <michaelni> maybe its not even so much an issue for the end user
[17:20] <dorileo> I see...
[17:21] <dorileo> michaelni: ok, I'm sending a new version with ts_packetsize
[17:21] <michaelni> ok thx
[18:53] <cone-920> ffmpeg.git 03Michael Niedermayer 07master:f6c95f4f8c2f: avcodec/dxa: add padding to decomp_buf
[18:53] <cone-920> ffmpeg.git 03Michael Niedermayer 07master:5a5c1b244281: avcodec/dxa: check for overread in decode_13()
[18:53] <cone-920> ffmpeg.git 03Michael Niedermayer 07master:ae09db102308: avformat/dxa: check avio_read() return
[20:07] <cone-920> ffmpeg.git 03Leandro Dorileo 07master:f9b997c33787: libavformat/mpegts: expose raw packet size
[20:07] <cone-920> ffmpeg.git 03Michael Niedermayer 07master:5eeeb4abf970: rpl: Dont fill the index with entries that failed to be parsed
[00:00] --- Fri Dec 20 2013
1
0
[01:04] <FunkyELF> are you not supposed to use id3 tags on mp4 files?
[01:04] <FunkyELF> I have a bunch of aac audio in mp4 files. When I run id3tag on them they no longer play in mplayer
[01:16] <bencc> is it possible to combine 25 video streams in real time?
[01:17] <bencc> all with the same resolution, size, bit-rate, fps
[02:42] <bencc> the following command just hangs ./ffmpeg -i test.webm test.mp4
[02:43] <bencc> do I have a syntax error?
[04:17] <jesseg> Hey I think I found a bug in ffplay - when laying an audio file, with the hystogram thing showing, if I resize the window to vertically zero height, it crashes. Anyone else notice that?
[10:11] <bencc> how can I convert webm to mp4 that can be played in browsers?
[10:11] <bencc> ./ffmpeg -i vid.webm vid.mp4
[10:11] <bencc> this just hangs
[10:19] <brontosaurusrex> bencc, full output
[10:25] <bencc> brontosaurusrex: http://dpaste.com/1512547/
[10:26] <viric> webm is the one that can be played in browsers
[10:26] <bencc> viric: not in IE
[10:26] <bencc> maybe the webm is corrupted
[10:26] <bencc> can I fix it instead of encoding?
[10:27] <brontosaurusrex> bencc, check mkvtoolnix, maybe there is way to remux it to new webm or mkv
[10:27] <brontosaurusrex> does that video.webm play in firefox well?
[10:28] <bencc> the webm palys in firefox but not in chrome
[10:28] <TheSchaf> ie does not support webm out of the box
[10:28] <TheSchaf> atleast not ie9, not sure about newer ies
[10:30] <brontosaurusrex> chrome should play webm stuff afaik
[10:31] <brontosaurusrex> bencc, a test http://video.webmfiles.org/big-buck-bunny_trailer.webm
[10:32] <bencc> the bunny plays both on Chrome and FF
[10:32] <bencc> I've created the webm with virtualbox video capture
[10:32] <bencc> maybe it creates bad index or something
[10:33] <bencc> how can I use mkvtoolnix to remux?
[10:33] <bencc> with mkvmerge?
[10:33] <TheSchaf> so what is you goal again? to play in ff and chrome?
[10:34] <viric> talking about IE
[10:34] <viric> I cannot find how to get into an https site whose cert is untrusted.
[10:34] <TheSchaf> ie11 doesnt play the bunny video for me
[10:34] <TheSchaf> maybe ie doesnt support webm at all
[10:34] <brontosaurusrex> bencc> with mkvmerge? < yes
[10:34] <bencc> play on Chrome, FF and IE
[10:34] <viric> I only get a page with "certificate error" and a button "close this webpage". No way to get in. Anyone knows?
[10:34] <bencc> IE doesn't support webm
[10:35] <brontosaurusrex> TheSchaf, this https://tools.google.com/dlpage/webmmf ?
[10:35] <TheSchaf> looks like there is no format that is supported by all 3 browsers
[10:35] <TheSchaf> well, mp4
[10:36] <TheSchaf> if yu have ff > 21
[10:36] <brontosaurusrex> TheSchaf, so mozilla is allready using that cisco thingy?
[10:36] <TheSchaf> http://www.w3schools.com/html/html5_video.asp
[10:36] <TheSchaf> according to that mp4 can run on chrome, ie and ff
[10:36] <TheSchaf> webm only chrome and ff
[10:38] <brontosaurusrex> i'd use AVC/mp4 and give insulting error messages to people running firefox without avc decoder, done.
[10:38] <lkiesow> TheSchaf: Afaik both mp4 and webm run in ie with the appropriate codecs installed, don't they?
[10:39] <TheSchaf> lkiesow: but mp4 runs out of th ebox, webm requires a third party plugin
[10:40] <brontosaurusrex> + ban ie users
[10:40] <lkiesow> Afaik. not with old Windows versions :)
[10:41] <lkiesow> Safest way would be to create both a WebM and a mp4
[10:42] <brontosaurusrex> <lkiesow> the encoders are not comparable, x264 is easier to use and much faster and much better and
[10:42] <brontosaurusrex> ...
[10:43] <TheSchaf> well i mean
[10:43] <TheSchaf> http://www.w3schools.com/browsers/browsers_stats.asp
[10:44] <TheSchaf> almost possible to ignore IE these days
[10:44] <brontosaurusrex> TheSchaf, give them flash player and same mp4/AVC file If you really must
[10:44] <viric> anyone knows how to get into this URL, bypassing the cert, in IE?
[10:44] <viric> https://84.89.61.89:41711/slkygg/prova.sh
[10:45] <viric> IE here offers only "close this website"
[10:45] <brontosaurusrex> viric, ask site owner and/or microsoft?
[10:45] <TheSchaf> my ie offers me "continue loading this page (not recommended)"
[10:46] <viric> TheSchaf: really? not mine.
[10:46] <viric> brontosaurusrex: I'm the site owner :)
[10:46] <TheSchaf> i have 11.0.1
[10:46] <viric> 11.0.9600 here
[10:46] <TheSchaf> as in http://support.microsoft.com/kb/2888505 that version
[10:46] <brontosaurusrex> viric, home made certificat?
[10:47] <TheSchaf> wait
[10:47] <TheSchaf> when i navigate to the page in a new window
[10:47] <TheSchaf> i get the continue option
[10:47] <TheSchaf> when i navigate from some other page, it doesnt give me continue
[10:47] <viric> brontosaurusrex: yes
[10:48] <brontosaurusrex> TheSchaf, ie has only 11% lately
[10:48] <brontosaurusrex> TheSchaf, so actually banning them would be no big deal.
[10:48] <TheSchaf> actually i think ie 11 is pretty good
[10:48] <TheSchaf> but i guess ie has just a really bad rep.
[10:48] <viric> TheSchaf: what is "navigate from some other page"?
[10:49] <brontosaurusrex> they had like 15 years to fix their c***, so ....
[10:49] <TheSchaf> viric, when i open a new ie and paste the url and hit enter i get the "continue" thing
[10:49] <TheSchaf> when i go to google.com then paste the url i only get "close htis page"
[10:49] <TheSchaf> so it differs if you want to go there directly or you are maybe "redirected"
[10:49] <viric> you paste the url in the google search box?
[10:49] <TheSchaf> no
[10:49] <TheSchaf> close all IE windows
[10:49] <TheSchaf> open new ie window
[10:49] <TheSchaf> enter url
[10:50] <TheSchaf> press enter
[10:50] <TheSchaf> then you should get continue
[10:50] <TheSchaf> then hit reload (navigate from "another page" to the evil page)
[10:50] <TheSchaf> and you wont get the continue
[10:50] <viric> hm no. I never get the continue option
[10:51] <TheSchaf> weird
[10:51] <viric> yes.
[10:56] <viric> TheSchaf: can you so download 'prova.sh' with IE, clicking that button?
[10:57] <TheSchaf> no it displays it
[10:57] <TheSchaf> some weird letters
[10:58] <TheSchaf> but i mean you can just use http instead of https? :D
[10:58] <viric> TheSchaf: weird letters? don't you see a short 'sh' script?
[10:59] <viric> DIED=0, ... ?
[11:00] <viric> this IE doesn't work at all. Maybe it's because "the site owner" forced ECDHE key negotiation, and IE doesn't support it? mh
[11:00] <TheSchaf>
[11:00] <TheSchaf> it shows chars like that
[11:00] <TheSchaf> no valid sh script
[11:00] <viric> it doesn't work at all for you then. Try with firefox :)
[11:01] <TheSchaf> same with ff
[11:01] <viric> Are you sure you typed https?
[11:01] <viric> and not http
[11:01] <TheSchaf> yeah
[11:02] <viric> ok then your browser situation is even worse than mine ;)
[11:03] <brontosaurusrex> viric, something is wrong http://paste.debian.net/plain/71740
[11:04] <brontosaurusrex> and it has to do with ssh/cert/something
[11:07] <viric> brontosaurusrex: wget --no-check-certificate ? can you try it?
[11:08] <viric> gnutls is not very verbose
[11:10] <brontosaurusrex> ok
[11:11] <brontosaurusrex> viric, same error
[11:17] <viric> auhm. Maybe you have an old gnutls/wget?
[11:17] <viric> gnutls 3.1.12 wget 1.14 work fine.
[11:19] <brontosaurusrex> GNU Wget 1.13.4
[11:19] <viric> maybe IE uses a old gnutls ;)
[11:19] <brontosaurusrex> haha, yeah
[13:23] <Jellicent> 2~
[13:23] <Jellicent> That was not supposed to happen
[13:24] <Jellicent> Hey guys, I'm aiming to stream (twitch.tv) but I'm having a little sound problem. I'm using alsa and the sound is awfully low. Has anything like this been reported? Couldn't find anything like this. Microphone works fine, just not the ingame recorded sound (what I hear basically).
[13:26] <Compn> Jellicent : probably have to adjust volume in amixer or whatnot
[13:26] <Compn> e.g. make speaker volume quieter and system volume louder
[13:26] <Jellicent> System volume is on 100%
[13:27] <Jellicent> That's the fancy part about it.
[13:27] <Jellicent> Everything's as high as it gets..
[13:29] <Jellicent> Compn: Would it be on any help if I pasted the script I'm using?
[15:37] <bencc> how can I fix a corrupted webm file?
[15:37] <bencc> it can be played in some browsers but not all
[17:19] <DelphiWorld> hi ffmpegifiers
[17:19] <DelphiWorld> this rtmp stream is giving me error evean i compiled with librtmp...
[17:19] <DelphiWorld> rtmp://radio.gmo.ps/live/livestream
[17:20] <DelphiWorld> giving me handcheck error
[17:26] <DelphiWorld> HandShake: client signature does not match
[17:28] <mootsadog> does anybody know if it's possible to force ffmpeg to finish encapsulating the output from a broken input file?
[17:29] <mootsadog> i have input from a hard drive camera's mpegTS that the camera recognized as "corrupted" and i believe i've hit that file because i'm getting "av_interleaved_write_frame()" input/output error" and unusable output
[17:31] <mootsadog> the input file plays fine in VLC. though you know what & it's possible this is happening because the server it's writing to is crapping out. sorry.
[17:42] <Compn> mootsadog : dang, thats bad luck, failing media and failing server
[17:43] <mootsadog> Hahah what else, right? ;) netatalk is giving me so many headaches.
[18:29] <DelphiWorld> ANYONE have a rtmp stream to try? (audio)
[18:29] <Compn> DelphiWorld : theres a list on http://wiki.multimedia.cx
[18:29] <CountryfiedLinux> I tried to install ffmpeg and it says no installation candidate
[18:30] <CountryfiedLinux> any suggestions? thanks in advance
[18:30] <Compn> what distro CountryfiedLinux :
[18:30] <DelphiWorld> Compn: what should i search?
[18:30] <Compn> rtmp ?
[18:30] <CountryfiedLinux> Compn, debian
[18:30] <Compn> DelphiWorld : http://wiki.multimedia.cx/index.php?title=RTMP
[18:30] <DelphiWorld> Compn: see that: rtmp://radio.gmo.ps/live/livestream
[18:31] <DelphiWorld> Compn: see if you can ffplay or ffprobe it
[18:31] <Compn> CountryfiedLinux : debian removed ffmpeg package
[18:31] <Compn> DelphiWorld : probably you are missing the swfurl or flashver parameters to access the stream
[18:31] <CountryfiedLinux> Compn, So is there a repository for it?
[18:31] <Compn> DelphiWorld : its rare you can play an rtmp stream without all the other referers
[18:31] <DelphiWorld> no need, its work perfectly in VLC
[18:31] <Compn> ok
[18:32] <DelphiWorld> CountryfiedLinux: bro, use git, you can compile it issueless
[18:32] <Compn> CountryfiedLinux : deb-multimedia repo has ffmpeg for apt get yes
[18:32] <Compn> or whatever its called now
[18:32] <bencoh> DelphiWorld: seems to ffprobe well here (both ffmpeg 1.1.5 and 2.1.1Ã
[18:32] <CountryfiedLinux> Compn, How do I install the multimedia repo?
[18:32] <Compn> CountryfiedLinux : http://www.deb-multimedia.org/
[18:32] <Compn> instructions there.
[18:33] <DelphiWorld> bencc: mmmmmmmmm, i'm using latest git ffmpeg and it's failing, are you using librtmp?
[18:33] <bencoh> not on this build no
[18:33] <DelphiWorld> bencoh: are you using git or pre-built?
[18:34] <Compn> fails in my nightly git build N-58288-g80e5859 built on Nov 19 2013
[18:34] <Compn> that rtmp url
[18:34] <DelphiWorld> Compn: also this one will fail: rtmp://sony.flash.internapcdn.net/sony_vitalstream_com/_definst_/mp4:NAB200…
[18:34] <bencoh> DelphiWorld: n2.1.1 tag from git
[18:35] <DelphiWorld> bencoh: would you tel me how to switch to it?
[18:35] <bencoh> from git ? git checkout n2.1.1
[18:36] <DelphiWorld> let's have some adventure
[18:36] <bencoh> DelphiWorld: http://pastebin.notk.org/pastebin.php?show=d62066243
[18:36] <DelphiWorld> LOL notk... crazy name:)
[18:39] <DelphiWorld> bencoh, the stream that worked for me is: rtmpe://wbads.fcod.llnwd.net/a2383/o25/mp4:thewb/video/thegeorgecarlinshow/…
[18:40] <saste> DelphiWorld, did you ever use ffserver?
[18:40] <DelphiWorld> saste: yes, i did and wrote a tutorial on using it;)
[18:40] <saste> DelphiWorld, where is the tutorial?
[18:40] <DelphiWorld> sec, saste
[18:41] <saste> i'm having a weird TCP issue with ffserver: http://thread.gmane.org/gmane.comp.video.ffmpeg.user/49400
[18:42] <DelphiWorld> saste, https://github.com/braice/MuMuDVB/blob/master/doc/TRANSCODE_EXTERNAL.txt
[18:43] <bencc> DelphiWorld: no. I'm talking about webm
[18:43] <DelphiWorld> sory bencohbut could you please give me the git cmd again?
[18:43] <DelphiWorld> bencc: lol, sory but my msg was to bencoh :-P
[18:47] <DelphiWorld> saste: isnt you're a ffmpeg dev?
[18:50] <Compn> DelphiWorld : this is also where ffmpeg devs ask for help
[18:50] <Compn> ffmpeg devs dont know the whole codebase, just like linus torvalds doesnt know all of linux :)
[18:50] <DelphiWorld> Compn: yeah i know, to get improvement offcource, but i just asked;)
[18:50] <Compn> ah
[18:50] <DelphiWorld> HEHEHE
[18:51] <DelphiWorld> i wish ffserver get some improvement and stability, it's fully buggy right now
[18:51] <DelphiWorld> also i dont like to fetch streams using ffmpeg and push to ffserver, it's better to let ffserver do the trick itself (without any external process
[18:52] <DelphiWorld> bencc2: did you get my pm?
[18:53] <CountryfiedLinux> missed part of the instructions I see now sorry brb
[18:54] <odedo> I'm having trouble converting Hebrew SRT subtitles to ASS: http://bpaste.net/show/160259/\
[18:56] <ubitux> it's an input option
[18:56] <spaam> try putting -sub_charenc before -i
[18:56] <spaam> and what ubitux said :D
[18:57] <DelphiWorld> spaam: you lighttpd developer?
[18:57] <ubitux> "enc" (encoding) is related to the "char" (character), not "encoding" in the sense of ffmpeg
[18:57] <ubitux> it's an input option used to specify the character encoding of the input
[18:57] <ubitux> so you put it before the -i as spaam said
[18:58] <spaam> DelphiWorld: you like my host? :)
[18:58] <DelphiWorld> spaam: seriously are you lighttpd dev?
[18:58] <spaam> DelphiWorld: yeah you can say that :)
[18:58] <odedo> Thanks ubitux and spaam, it's working
[18:59] <spaam> odedo: great :)
[18:59] <DelphiWorld> spaam: did lighttpd get any streaming functionality like nginx?
[18:59] <CountryfiedLinux> still says no installation candidate
[18:59] <DelphiWorld> odedo: where you live in IL?
[18:59] <spaam> DelphiWorld: "streaming functionality" ?
[18:59] <DelphiWorld> CountryfiedLinux: did you apt-get update? did you install deb-multimedia-keyring?
[19:00] <odedo> DelphiWorld, Tel Aviv, and you? ;o
[19:00] <DelphiWorld> spaam: push live http stream to lighttpd, push / get rtmp, hls...
[19:00] <saste> DelphiWorld, to me it looks more like a TCP buffering or network issue
[19:00] <DelphiWorld> odedo: i'm algerian, but comming there soon:)
[19:00] <CountryfiedLinux> DelphiWorld, Yes I did.
[19:01] <DelphiWorld> saste, would you describ your setup?
[19:01] <saste> since i'm relatively new with using ffserver, you may have already encountered the problem
[19:01] <DelphiWorld> odedo: i will come to Telaviv to buy a audio encoder:)
[19:01] <saste> DelphiWorld, i did in the post i linked
[19:01] <DelphiWorld> saste: show me please again
[19:01] <spaam> DelphiWorld: no. never heard of it .
[19:01] <DelphiWorld> spaam: nginx do rtmp...
[19:01] <saste> basically it's a one stream ffserver instance, streaming a synthetic video, nothing fancy
[19:01] <odedo> DelphiWorld, I hope you'll have a nice time :) I wouldn't hurry though, it's kind of really cold for the time being (although middle-eastern cold)
[19:02] <saste> http://thread.gmane.org/gmane.comp.video.ffmpeg.user/49400
[19:02] <spaam> DelphiWorld: i guess you are talking about https://github.com/arut/nginx-rtmp-module ?
[19:02] <DelphiWorld> spaam: YUP YUP
[19:02] <DelphiWorld> odedo: yeah i know, maybe i'lle be there in february
[19:03] <CountryfiedLinux> Here's my source list http://pastebin.com/45ZS9KHH
[19:03] <CountryfiedLinux> DelphiWorld, ^
[19:04] <DelphiWorld> CountryfiedLinux: using wheezy?
[19:04] <CountryfiedLinux> DelphiWorld, no
[19:04] <CountryfiedLinux> DelphiWorld, Should I pin that one?
[19:04] <spaam> DelphiWorld: anyway. it wont happen in lighttpd 1.4.x maybe if someone add it to lighttpd 2.0
[19:04] <DelphiWorld> what version?
[19:04] <CountryfiedLinux> DelphiWorld, Testing
[19:04] <acovrig> I am encoding a stream from a v4l device, to a m4p (libx264, libfdk_aac). I know the issue with no moov atom for streaming, but is there any way for me to see what it is getting in?
[19:04] <DelphiWorld> spaam: want me to help you to compile it ?
[19:07] <DelphiWorld> bencoh: thank dude, work in 2.0
[19:08] <DelphiWorld> bencoh: do you have librtmp enabled in your build?
[19:09] <Compn> acovrig : nope. mp4 cannot be played incomplete, unless you build some kind of custom player with custom moov headers
[19:09] <Compn> or reindex the file with qt_faststart
[19:09] <Compn> (after its done being madE)
[19:11] <acovrig> Compn: what format/codecs do you recommend that can be played partially through, and is the most compatible?
[19:15] <CountryfiedLinux> Is it normal for ffmpeg to not be able to be installed in Debian Testing?
[19:16] <sacarasc> ffmpeg isn't in Debian any more.
[19:17] <CountryfiedLinux> sacarasc, not even the multimedia repository?
[19:17] <sacarasc> It got replaced by libav.
[19:18] <CountryfiedLinux> thanks
[19:18] <ubitux> Compn: stop recommending qt-faststart if the user is muxing the file, we have -movflags +faststart now
[19:18] <ubitux> qt-faststart is a hack
[19:19] <DelphiWorld> ubitux: :)
[19:19] <ubitux> qt-faststart is basically doing a memsearch for the chunk offset list
[19:19] <ubitux> which can easily break
[19:20] <DelphiWorld> ubu2.0 solved my rtmp issue
[19:21] <DelphiWorld> but now i'm enabling librtmp, to see if the issue cam through latest git or issue is in librtmp
[19:32] <DelphiWorld> ok, my conclusion:
[19:32] <DelphiWorld> librtmp cant play the stream...
[19:37] <Jellicent> Anyone using ALSA having the problem that the sound is super low (and barely hearable)? My system volume's on 100%, can't go past that.
[19:39] <viric> I don't have that problem
[19:39] <viric> but check *all* volume bars.
[19:40] <viric> "the system volume" may not map well to hardware mixers
[19:42] <Jellicent> You mean master? That's on 100%, too.
[19:42] <Jellicent> My sound is fine when I'm listening to it, I can regulate it on my headphones, that's why I have it on 100% in ALSA.
[19:43] <Jellicent> When I'd pull it all the way up on my headphones my ears would bleed.
[19:43] <Jellicent> And the sound would still be barely hearable when recorded.
[19:43] <Jellicent> Which is so weird.
[19:43] <viric> recorded?
[19:43] <Jellicent> Well, not recorded, but. Uh, I'm bad with words in this.
[19:44] <Jellicent> See, I'm trying to stream to twitch, maybe that helps. And when I watch the actual broadcast the sound is super low, 1/10 as loud as I heard it when streaming for example.
[19:44] <Jellicent> And I can't find out why.
[19:45] <Jellicent> Someone just suggested I boost my mic. But would that really help? I'm not recording through a physical microphone. I have never streamed on linux, so I have no idea if there's a mic entry for that.
[19:47] <viric> if you don't record, no idea what do you stream.
[19:48] <Jellicent> I planned on streaming StarCraft II on twitch.tv.
[19:48] <viric> ok, I might be the one who knows least about that.
[19:59] <DelphiWorld> bencoh: here?
[20:05] <Jellicent> viric: Okay. But well, if I was just going to stream my music, as a stupid example, it still wouldn't be any better.
[20:34] <Compn> ubitux : then put that +faststart in a message whenever anyone encodes to mp4
[20:34] <Compn> :)
[00:00] --- Fri Dec 20 2013
1
0
[02:37] <Zeranoe> does frames:v go off of intput frames or output? If I'm also changing the fps, how do I know how many frames if it is off the output
[02:52] <Zeranoe> it goes off the output, which makes it hard to set with -r without double converting
[03:07] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:645c94b61bfd: Revert "Merge commit '3bc2e89c76e88ae6f1fd5287e0b11abcfc3c601c'" (Bump libavutil major version to account for the LLS API/ABI changes.)
[05:33] <cone-446> ffmpeg.git 03Anssi Hannula 07master:e10fccf62a36: avcodec/aacdec: default to non-wide 7.1 in non-strict mode
[09:49] <ubitux> wbs: you probably need to fix the type of the chunk_length as well
[09:49] <ubitux> which was changed to "int" for some reason (???)
[09:52] <wbs> ubitux: hmm, possibly, even if that's a separate issue
[12:39] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:d8fd183683b7: vc1: Fix mb_height for field pictures
[12:39] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:09d6ceac4d10: Merge commit 'd8fd183683b7495566b7e510a6536ae2efe8dfed'
[12:47] <cone-433> ffmpeg.git 03Martin Storsjö 07master:b3189aff8b47: png_parser: Fix parsing on big endian
[12:47] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:b8f8f7c7fe37: Merge remote-tracking branch 'qatar/master'
[13:32] <wm4> ubitux, saste: can avfilter_graph_parse() be fixed before the Libav 10 release? I know, normally this should wait until the next lavfi major bump...
[13:34] <saste> wm4, what's the problem with avfilter_graph_parse()?
[13:34] <wm4> saste: it's incompatible with Libav 10, and has been marked as deprecated
[13:35] <wm4> but only the incompatible use is deprecated
[13:35] <wm4> after the bump, it's not deprecated anymore, and works like in Libav
[13:38] <nevcairiel> wasnt lavfi just bumped
[13:39] <wm4> yes, but not really: "All scheduled API changes are deferred to the next bump."
[13:39] <nevcairiel> then undeffer one
[13:39] <saste> wasn't the bump reverted?
[13:39] <nevcairiel> :D
[13:39] <nevcairiel> that was avutil
[13:40] <saste> i lost track of all the version bump hacks
[13:40] <nevcairiel> in any case, this version thing is all w rong anyway, i'm sure i can find a combination of versions that still fails, yet is considered "valid"
[13:40] <wm4> this version situation is quite comical
[13:40] <wm4> I'll just stand here and laugh about the chaos that's causing for the same people who defend it
[13:40] <saste> well guys feel free to send patches, I honestly lost any interest in that
[13:41] <wm4> :)
[14:23] <study> hi there, is there any way to get full file length from AVFilter context
[14:23] <study> hi there, is there any way to get full file length from AVFilter context?
[14:27] <wm4> study: no
[15:20] <cone-433> ffmpeg.git 03Stefano Sabatini 07master:8937af7b6fca: lavf/segment: increase logging level for message with segment start information
[15:20] <cone-433> ffmpeg.git 03Stefano Sabatini 07master:aeccb522ebb6: lavf/concatdec: fix crash in case a file in script cannot be opened
[16:02] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:1935173fd1de: avcodec/dirac_dwt: initialize support value for fidelity wavelet
[18:22] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:4a2570f50a2f: avcodec/nuv: zero buffer padding
[19:47] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:d8b4ba3c5a9c: avformat/isom: zero MPEG4AudioConfig to avoid leaving fields uninitialized
[20:04] <cone-433> ffmpeg.git 03Anton Khirnov 07master:fe533ffdcffa: Prepare release notes for Libav 10 "Eks"
[20:04] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:03a5c44faece: Merge remote-tracking branch 'qatar/master'
[20:19] <j-b> 'evening
[20:39] <clever> aha, so thats why my draw_slice is doing nuts
[20:39] <clever> its not giving me a full frame!
[22:24] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:cb079b1b2bb1: avcodec/pngdec: fix signed overflow
[23:24] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:8faabf3bd29c: avformat/ipmovie: Check that the OPCODE_INIT_AUDIO_BUFFERS size is large enough
[23:24] <cone-433> ffmpeg.git 03Michael Niedermayer 07master:5f0d552c9b4c: avformat/ipmovie: remove superflous ()
[00:00] --- Thu Dec 19 2013
1
0
[00:20] <Overflip> anyone?
[00:21] <bencc> what video formats are lossless?
[00:21] <bencc> webm? h.264?
[00:22] <Overflip> I only have libx264.a
[00:22] <Overflip> not libx264.so
[00:22] <Overflip> hmm
[00:22] <Overflip> sorry I'm not the greatest with linux
[00:24] <Vandalite> one second, was waiting for that.
[00:25] <Vandalite> http://pastebin.com/pjiA209j
[00:25] <Vandalite> shows both versions of the same command, run as different users on the same server.
[00:25] <llogan> bencc: http://superuser.com/a/347434/110524
[00:26] Action: llogan failed to paste URL correctly...
[00:26] <llogan> ...i mean your url, Vandalite...anyway i'll look at it
[00:28] <llogan> Vandalite: unrelated, but "using cpu capabilities: none!" is not good
[00:28] <Vandalite> that's intentional
[00:28] <llogan> why?
[00:29] <Vandalite> The server doesn't have YASM
[00:29] <llogan> install it
[00:29] <Vandalite> can't. not my authority to do so. That's not the problem here though
[00:29] <llogan> one step at a time
[00:30] <Vandalite> you'll note the 'root' version gets past that without problems.
[00:30] <Overflip> llogan if you could give me a hand after that would be great =)
[00:30] <fonso1> hey there, I have installed ffmpeg in a amazon EC2 instance following a CentOS tutorial, but now if I run the command : whereis ffmpeg it returns me nothing, but doing $ffmpeg it founds. Thanks
[00:30] <llogan> Vandalite: why are you not using the presets? there is no reason to attempt to use a million encoding options
[00:31] <Vandalite> it is using a preset... (ipod320)
[00:31] <llogan> that's not a real preset
[00:31] <llogan> https://trac.ffmpeg.org/wiki/x264EncodingGuide
[00:31] <llogan> anyway, start simple. does it work with: ffmpeg -i input -vcodec libx264 -an output.mp4
[00:31] <Vandalite> but actually to answer your question, except for the '-y -v verbose' at the end, this is exactly the output the script for this user generated.
[00:31] <llogan> the script is bad
[00:32] <Vandalite> well let's see if the simple one works
[00:34] <Vandalite> same problem.
[00:34] <Vandalite> as the user, failure.
[00:34] <Vandalite> as root: success.
[00:35] <Vandalite> i even tried putting 'strace' on it to see if it was running into fileystem errors.
[00:35] <Vandalite> ... nope, except for a few attempts at locating the ffpreset file (which it eventually found)
[00:36] <Vandalite> want me to update the pastebin with the new outputs?
[00:37] <llogan> ok
[00:38] <Overflip> I'm so lost why I'm getting ERROR: libx264 not found
[00:39] <Overflip> I just installed it from source
[00:39] <Overflip> why cant it find it?
[00:39] <llogan> Overflip: see "man whereis": whereis has a hard-coded path, so may not always find what youre looking for.
[00:39] <Vandalite> llogan: http://pastebin.com/60N4gT3r
[00:40] <Overflip> whereis libx264 gives me libx264: /usr/local/lib/libx264.a
[00:41] <llogan> Overflip: --extra-libs="-ldl"
[00:41] <Vandalite> whups, user-version of the output didn't have '-v verbose'
[00:42] <Overflip> Thank you llogan
[00:42] <Overflip> works
[00:42] <Overflip> what does --extra-libs="-ldl" mean?
[00:42] <llogan> Vandalite: options don't go after the last output. global options are the first options
[00:42] <Vandalite> llogan: heh, that's funny, cause it totally takes them at the end too.
[00:42] <Vandalite> -v debug works fine there... and so does -y
[00:44] <llogan> options may work fine, but you should follow the documentation so things will work as expected
[00:45] <Vandalite> same output putting -v and -y at the beginning didn't change the result.
[00:45] <llogan> i'm just pointing it out
[00:46] <llogan> does the verbose output show anything useful?
[00:48] <Vandalite> It added exactly one line... doesn't seem to be anything but info about the x264 encoder settings: "[graph 0 input from stream 0:0 @ 0x8c37e0] w:1920 h:1080 pixfmt:yuv420p tb:1/2997 fr:2997/100 sar:0/1 sws_param:flags=2"
[00:49] <llogan> i don't know why it's not working for you. you could just download a recent static binary and use it instead
[00:50] <llogan> http://ffmpeg.org/download.html#LinuxBuilds
[00:50] <Vandalite> the odd thing is, in verbose mode, the libx264 parts of the output have two lines that do not appear when running as the user.
[00:51] <rager> howdy
[00:51] <Vandalite> right after the 'using cpu capabilities: none' line are two more lines that *do not* show up on user mode.
[00:52] <rager> I'm working on a java wrapper for ffmpeg, and I've got this last error while building: "error: invalid application of 'sizeof' to incomplete type 'AVDictionary'"
[00:52] <fonso1> Hey, the ffmpeg is saying me : permission denied, I set up 777 on the dir and all the files which ffmpeg are installed...
[00:53] <llogan> Vandalite: are you encoding for iOS?
[00:54] <Vandalite> well, the profile originally was baseline 1.3, but that changed when we started using your 'simplified' ffmpeg command
[00:54] <Vandalite> now it says profile high level 4.0
[00:54] <Vandalite> but the point is, that 'profile' line doesn't show up at all when run as the user.
[00:54] <llogan> the simple command was to rule out that any options or whatever were causign the issue
[00:55] <llogan> i was going to give you a non-ancient command but i don't know why you're encoding in the first place
[00:56] <Vandalite> the original invocation I had was how clipshare was configured.
[00:56] <llogan> so that means it's right?
[00:56] <llogan> this is #ffmpeg.
[00:57] <Vandalite> i ruled out clipshare as the cause when I isolated this as a problem running ffmpeg as the user, and not root.
[00:57] <Vandalite> i'm trying to figure out what the difference is. every file accessed by ffmpeg during the conversion is accessible by both root, and the user.
[00:58] <llogan> did you compile it?
[00:58] <Vandalite> yes, this is compiled from source.
[00:58] <rager> are you invoking ffmpeg with just "ffmpeg" or the full path to the binary?
[00:58] <Vandalite> I even tried updating x264 with the latest stable snapshot, no help.
[00:58] <Vandalite> full path.
[00:58] <llogan> https://trac.ffmpeg.org/wiki/CentosCompilationGuide
[00:59] <rager> oh.. ffmpeg on centos? that's not fun.
[01:02] <Samus_Aran> is there any way to add silence to a video that doesn't produce corrupted audio packets?
[01:03] <Samus_Aran> I'm trying with the -f lavfi -i aevalsrc=0 method, but if I try to use the resulting file with ffmpeg, it has garbage audio (mostly white noise)
[01:03] <Samus_Aran> such as re-encoding it
[01:05] <llogan> i can't duplicate the issue
[01:05] <Samus_Aran> I'm looking for a way to add audio to a video created with images that has proper headers
[01:05] <Samus_Aran> so I can then use the concat filter to put it with other video clips
[01:05] <Samus_Aran> but if I try to do anything at all with the resulting file, it turns the proper silence into white noise
[01:05] <llogan> you didn't provide enough information to duplicate the issue. using aevalsrc=0 works for me.
[01:06] <Samus_Aran> it plays fine in mplayer or whatever, silence
[01:06] <Samus_Aran> but ffmpeg can't re-encode it
[01:06] <llogan> so show your commands and the console outputs so we can try to duplicate the issue
[01:07] <llogan> you've been here enough to know this
[01:07] <rager> Samus_Aran: you could just remove the audio track?
[01:07] <rager> passthru the video stream and don't use the audio
[01:08] <Samus_Aran> rager: some of the clips have audio, some don't. I thought when using concat it needed to be the same on all of them?
[01:08] <Samus_Aran> llogan: I don't have working commands, I've been trying to work out the commands for hours
[01:08] <llogan> i meant for you to show your commands that *are not* working
[01:08] <rager> sorry, didn't read enough
[01:09] <llogan> i don't understand how you got white noise from aevalsrc
[01:09] <llogan> *aevalsrc=0
[01:09] <Samus_Aran> llogan: I didn't. I got white noise by re-encoding the clip that was produced using aevalsrc
[01:09] <rager> llogan: I've got some issues with some code I'm compiling against the libs: http://hastebin.com/biboruhahe.coffee
[01:09] <llogan> Samus_Aran: so show these commands.
[01:09] <llogan> jesus
[01:10] <rager> I'm not really exactly sure what the error message means, but it seems like I'm misusing AVDictionary
[01:10] <llogan> rager: sorry, i don't know.
[01:10] <rager> no worries
[01:11] <rager> I wish I knew C properly
[01:11] <llogan> is jjmpeg something you're making?
[01:11] <llogan> i mean is it something you wrote
[01:12] <rager> naw, it's a project I found that wraps ffmpeg in java
[01:12] <rager> I'm trying to make it work with ffmpeg 2.x rather than 1.x
[01:13] <rager> the original author seems to have gone to lengths to avoid using AVDictionary, though, and I'm not really sure how I should go about this
[01:13] <llogan> we don't support 3rd party stuff here
[01:13] <rager> I'm not trying to get support for 3rd-party stuff
[01:13] <rager> I'm trying to be the 3rd-party stuff.
[01:14] <rager> AVDictionary is not 3rd-party.
[01:17] <llogan> obviously i was referring to jjmpeg
[01:23] <Samus_Aran> llogan: http://pastebin.com/raw.php?i=2dNrwH07
[01:24] <Samus_Aran> I had to use -frames:v 356 because -shortest was non-functional
[01:24] <Samus_Aran> but I may have done something wrong. anyhow, it was 356 frames so that worked.
[01:24] <llogan> thanks. can you also include: ffmpeg -i out.mkv -i Public_Domain_Dedication_With_Audio.mkv
[01:25] <Zeranoe> Is there any FFmpeg command that strips the last few frames off of the end of a video?
[01:25] <llogan> Samus_Aran: although i see that the silent audio is pcm_f64le and the other is pcm_s16le
[01:28] <llogan> oh...i read that wrong.
[01:29] <Samus_Aran> http://pastebin.com/raw.php?i=ddPEhr8Y
[01:29] <Samus_Aran> it's at the end
[01:30] <Samus_Aran> Zeranoe: there are various methods for this. if you know the number of frames, you can use -frames:v with a re-encode or stream copy.
[01:30] <Samus_Aran> Zeranoe: stream copying is not exact on most codecs, though, due to keyframes or whatever. so you may need to re-encode to get exact
[01:31] <llogan> Samus_Aran: there is a difference in audio rate though: 48000 vs 44100
[01:32] <llogan> oh, and one is stereo and one is mono
[01:32] <Samus_Aran> shouldn't that automatically be handled? I mean, I am re-encoding to pcm_s16le in that example
[01:32] <Zeranoe> Samus_Aran: How do I get the number of frames? Do i have to do a pipe with encoding to get it like this example http://stackoverflow.com/questions/2017843/fetch-frame-count-with-ffmpeg or do you know another way?
[01:32] <Samus_Aran> but I will enforce it to the correct rates and try again
[01:35] <llogan> and -shortest works for me for the aevalsrc command
[01:35] <Samus_Aran> Zeranoe: I'm not sure what the proper way is, personally I would just dump tiny images to get the file count. you can probe the number of frames, but this isn't always accurate
[01:36] <Zeranoe> Samus_Aran: On Windows here... Hopefully cmd/batch can provide what I need. Thanks anyway, I'll figure it out
[01:38] <Samus_Aran> Zeranoe: I was just suggesting an ffmpeg command, nothing specific to Linux
[01:38] <Zeranoe> Samus_Aran: Well, you would still need to count the # of files if you wanted to automate it
[01:38] <Samus_Aran> it spits out numbered frames
[01:38] <Samus_Aran> so just look at the last file
[01:39] <Samus_Aran> i.e. ffmpeg -i foo.mp4 -vf scale=160:-1 -an "frame-%05d.bmp"
[01:41] <Zeranoe> Samus_Aran: That would work. But you still need to find the last file number
[01:41] <Vandalite> not sure if this is gonna do any good, but reconfiguring ffmpeg and x264 to use yasm.
[01:41] <Zeranoe> then extract the number, and feed it as frames
[01:43] <Samus_Aran> Zeranoe: sorry, I didn't get that you were trying to script this
[01:44] <Samus_Aran> it's always removing a fixed number of frames?
[01:44] <Samus_Aran> if you dump all the files, in a Batch script you can do a for loop on them with a variable counting up by +1
[01:44] <Zeranoe> Samus_Aran: I just want to strip the last few frames from the end. It makes sense I need a frame count, and on Bash I would have no issues scripting. On Windows it's not as fun
[01:46] <Vandalite> llogan: the compile guide you linked me to was for a local non-system version of ffmpeg. I need to fix the *system* one.
[01:47] <llogan> then move the binary to whever your PATH points to
[01:47] <Samus_Aran> llogan: do I need to use filters to convert the mono to stereo, or does aevalsrc have some way to specify that?
[01:47] <Vandalite> the build i'm running right now already writes to the system version
[01:47] <llogan> or change the prefix to wherever you want
[01:48] <Vandalite> i'm just updating it to include yasm
[01:48] <llogan> Samus_Aran: aevalsrc="0|0"
[01:49] <Samus_Aran> I just tried -ac 2 and it seems to have worked, though I don't know what it did
[01:49] <Samus_Aran> llogan: thanks
[01:49] <llogan> -ac works too. or you could use amerge+pan
[01:50] <Samus_Aran> does -ac 2 on a mono file duplicate that channel to the others?
[01:50] <llogan> not amerge+pan..i was thinking 4 channels to 2
[01:51] <Samus_Aran> *to the other
[01:51] <llogan> i don't know
[01:54] <Samus_Aran> is there some way to choose the order of the streams in an output file? some of my files have audio first, some video first, which makes mkvmerge a huge pain
[01:56] <llogan> Samus_Aran: switch your -maps so vidoe is listed first
[01:56] <llogan> vidoe and audoi and subtitlse
[01:57] <Vandalite> no luck. using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2, still fails with no error.
[01:59] <Samus_Aran> llogan: I am still getting white noise when the formats match
[01:59] <Samus_Aran> with the PCM errors
[01:59] <llogan> can you provide the files?
[02:00] <llogan> the inpus
[02:00] <llogan> *inputs
[02:00] <Samus_Aran> what does the default in this mean:
[02:00] <Samus_Aran> Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s (default)
[02:00] <Samus_Aran> the other audio stream doesn't have that
[02:01] <Samus_Aran> with the ffmpeg -i one -i two
[02:01] <rager> found my issue
[02:02] <rager> in libavutil/struct.h, AVDictionary does not have its implementation shown
[02:02] <rager> so I had to just add one to my project so that sizeof(AVDictionary) would work
[02:02] <Samus_Aran> llogan: can I PM you the links?
[02:02] <llogan> sure
[02:06] <Samus_Aran> just a moment, need to poking at Apache.
[02:08] <Zeranoe> Samus_Aran: My solution: http://pastebin.com/hFhcRw8i
[02:10] <Samus_Aran> Zeranoe: nice. Batch is a lot uglier than Bash. :p
[02:10] <Zeranoe> Samus_Aran: Tell me about it
[02:14] <Vandalite> llogan: want to hear odd? If i change the encoder format from libx264 to another codec (I picked libxvid) it works.... even in user mode
[02:19] <llogan> Samus_Aran: i ran out of time to look at it now. i have to leave, but you can try asking on ffmpeg-user mailing list
[02:20] <Samus_Aran> llogan: okay, thanks anyhow.
[02:27] <Samus_Aran> llogan: I still don't have it working with ffmpeg concat filter, but I do have it working with mkvmerge now.
[02:27] <Samus_Aran> yay.
[02:28] <Samus_Aran> that's good enough for me. I just need a finished product that works. :)
[03:18] <Zeranoe> Does ffmpeg ever print multiple lines of "frame= " when encoding? If so, how could I reproduce that
[03:41] <DeadSix27> it actually does print multiple
[03:41] <DeadSix27> wait, zeranoe?
[03:41] <DeadSix27> just a coincidence?
[03:47] <relaxed> Zeranoe: for i in {01..10}; do printf "%s\r" "frame $i";sleep 1;done
[04:51] <Samus_Aran> where on the command line is -shortest supposed to appear?
[04:52] <Samus_Aran> I can't get it to work. I have a 13.00 second clip of images I am encoding to AVC, generating silence ith aevalsrc. it stops at 39 minutes or more
[04:52] <Samus_Aran> using -frames:v doesn't help, the audio keeps going
[05:49] <relaxed> Samus_Aran: after the input. show me your command
[05:54] <Samus_Aran> ffmpeg -r 25 -i .rvs_temp-23737/frame-%03d.tiff -f lavfi -i aevalsrc=0|0 -frames:v 324 -shortest -c:a pcm_s16le -ar 48000 -ac 2 -strict experimental -codec:v libx264 -pix_fmt yuv420p -preset veryfast -crf 18 -sn Public_Domain_Dedication.mkv
[05:55] <Samus_Aran> this produces a 40 minute video
[05:55] <Samus_Aran> but the images stop after 324 frames
[05:57] <Samus_Aran> presumably the audio keeps playing for the rest, not sure, it's silent, heh
[05:57] <relaxed> use either -frames:v or -shortest
[05:59] <Samus_Aran> I'm pretty sure I've tried with both, with neither, and with each one by itself, but I'll try again
[06:00] <Samus_Aran> with only -shortest it is still running away
[06:00] <Samus_Aran> 19 minutes encoded so far
[06:01] <relaxed> it should stop when the frames run out.
[06:02] <Samus_Aran> frame= 322 fps=2.5 q=23.0 size= 2978kB time=00:38:26.69 bitrate= 10.6k
[06:02] <Samus_Aran> it is stuck there
[06:02] <Samus_Aran> hasn't increased the time for a couple minutes
[06:02] <relaxed> why aevalsrc=0|0 and not aevalsrc=0 ?
[06:03] <Samus_Aran> llogan suggested using that for stereo. I also tried -ac which worked, forgot to remove one of them
[06:03] <Samus_Aran> not sure if they are identical or not
[06:04] <Samus_Aran> Duration: 00:38:26.77, start: 0.000000, bitrate: 1540 kb/s
[06:04] <Samus_Aran> final file
[06:05] <Samus_Aran> there are 324 TIFF images on the input. that output was stuck at 38 minutes with 322 frames
[06:05] <Samus_Aran> it never got to the final frames
[06:11] <Samus_Aran> my only guess is that converting the audio rate is altering the syncing with the video frames.
[06:12] <Samus_Aran> I'm trying to generate 48KHz/16bit/stereo
[06:12] <Samus_Aran> I guess I will try generating the silence with sox and muxing it
[06:15] <relaxed> you just need silence?
[06:15] <Samus_Aran> I tried removing the -ac and -ar, but it still ran away
[06:16] <Samus_Aran> I'm trying to add an empty audio channel to the files I generate from images
[06:16] <relaxed> tried using /dev/zero for the audio?
[06:18] <Samus_Aran> I'd prefer to not use something Linux-specific, as it's part of a set of scripts
[06:18] <Samus_Aran> sox is fine, if it works remuxing
[06:18] <Samus_Aran> I've had nothing but problems for weeks with FFmpeg.
[06:18] <relaxed> scripts for windows?
[06:19] <Samus_Aran> I was thinking other Unix systems, but Windows as well, sure. Bash and common tools are avail.
[06:19] <Samus_Aran> I'd be surprised if sox wasn't available for Win
[06:20] <Samus_Aran> also, the file produced from 324 frames at 25 FPS is 13.04 seconds, according to ffprobe
[06:20] <Samus_Aran> but the input of the frames says 12.96
[06:20] <Samus_Aran> 2 frames behind.
[06:21] <Samus_Aran> not sure what that's about, but random frame shifts seem to be standard with all ffmpeg cutting commands
[06:21] <Samus_Aran> including the frame-accurate ones
[06:25] <relaxed> if you know how many frames there are then you're done
[06:25] <Samus_Aran> relaxed: it doesn't stop
[06:25] <Samus_Aran> or if you're talking about the other, the time of the video is longer than the actual frames allow for
[06:25] <relaxed> does it stop without -i aevalsrc=0|0 ?
[06:26] <Samus_Aran> without audio, it stops fine
[06:26] <Samus_Aran> other than being 2 frames longer on time
[06:26] <relaxed> you've tried with git master?
[06:26] <Samus_Aran> I'm using git from when I started messing around, a couple weeks ago I think
[06:27] <Samus_Aran> ah, it's more than that, a couple months behind
[06:27] <relaxed> update to latest and see if it's fixed. if not, file a bug report
[06:32] <Samus_Aran> seems to be working fine using sox to generate silence and remuxing.
[06:33] <relaxed> still needs to be fixed
[06:48] <relaxed> Samus_Aran: did you quote aevalsrc=0|0?
[06:48] <relaxed> it works fine here
[06:48] <relaxed> -i 'aevalsrc=0|0'
[07:09] <Samus_Aran> relaxed: aevalsrc=0|0 was quoted, and it worked fine
[07:09] <Samus_Aran> the problem was it not stopping output for 40+ minutes
[07:10] <Samus_Aran> 0|0 without -ac 2 did stereo
[07:20] <relaxed> it stop here using ffmpeg version N-45667-ge10fccf
[07:20] <relaxed> stops*
[07:27] <lucaswang> hi, i have a question. what is the different with AVDISCARD_BIDIR and AVDISCARD_NONREF?
[07:27] <relaxed> Samus_Aran: does this stop? ffmpeg -f rawvideo -pix_fmt yuv420p -framerate 15 -video_size 320x240 -i /dev/zero -f lavfi -i 'aevalsrc=0|0' -map 0 -map 1 -frames:v 200 -c:a:1 pcm_s16le -ar 48000 -ac 2 -y output.avi
[07:30] <Samus_Aran> relaxed: will try, moment.
[07:32] <Samus_Aran> frame= 200 fps=112 q=2.0 Lsize= 261kB time=00:00:13.33
[07:32] <Samus_Aran> stops quickly
[07:32] <Samus_Aran> Duration: 00:00:13.40, start: 0.000000, bitrate: 159 kb/s
[07:33] <relaxed> odd... change the encoding part to match your workflow
[07:35] <Samus_Aran> I'm trying to generate an .mkv with silent audio and avc from images.
[07:35] <Samus_Aran> but when I used -frames:v it didn't help because it wasn't getting to the end frames
[07:37] <Samus_Aran> my next issue is that the .mkv file I create from the images doesn't return a duration with ffprobe
[07:37] <Samus_Aran> so I have no way to automatically generate the silence with sox.
[07:37] <relaxed> you have -sn in your command
[07:37] <Samus_Aran> it says N/A in the output
[07:37] <relaxed> pastebin your command and all output
[07:37] <Samus_Aran> to remove subtitles
[07:44] <Samus_Aran> I just assembled the paste, then I noticed that the issue is only in ffprobe and not in ffmpeg itself
[07:44] <Samus_Aran> ffprobe sees the duration as N/A, ffmpeg sees it as 13.04
[07:45] <Samus_Aran> I was using ffprobe to grab the duration. works on most files.
[07:45] <Samus_Aran> I guess I'll switch to parsing ffmpeg output for the duration.
[07:48] <Samus_Aran> correction: ffprobe video.mkv also returns 13.04, but when I use: ffprobe -sexagesimal -show_streams -print_format compact, it changes to N/A on this file generated by ffmpeg.
[07:49] <Samus_Aran> works fine on my other random video files I have laying around
[14:27] <bratao1> Help ! I encoded in ffmpeg a raw opus audio
[14:28] <bratao1> Help ! I encoded in ffmpeg a raw opus audio ( run avcodec_encode_audio2 and saved the packet directly) . It do work with other formats, but I canŽt get anyone to read this file. How can I read this file back in ffmpeg ?
[15:09] <grkblood13> is there a way to transcode with libspeex without wrapping it in an ogg container?
[15:09] <grkblood13> I want the speex output without a container
[15:12] <Samus_Aran> grkblood13: either specify the output format, or use a file extension for the format
[15:13] <grkblood13> well, speex isn't listed in -formats and I'm not outputting to a file. I'm piping stdout via pipe:1 to another process.
[15:14] <Samus_Aran> what's wrong with having a header?
[15:16] <grkblood13> Well, it's additional overhead and the wrapper+header just use more bandwidith. Right now I'm stripping that stuff out so I wanted to see if I could avoid having to do that.
[15:18] <grkblood13> If I can avoid cutting out the audio data from the container with additional code I would like to do that
[15:19] <Mavrik> grkblood13, it seems that currently ffmpeg just doesn't support speex output without muxing it into something
[15:20] <grkblood13> real quick, when people say muxing what do they mean? I'm not affluent in the audio lingo.
[15:21] <Compn> grkblood : you know avi, mov, mp4, mkv right ?
[15:21] <grkblood13> and thanks Mavrik, I guess I'll continue cutting out the audio data myself.
[15:21] <Compn> inside those 'containers' can contain 'codecs'
[15:21] <grkblood13> right, i got that
[15:21] <Compn> you can output speex with -f rawaudio probably
[15:21] <Compn> if you want raw speex
[15:21] <Compn> or -f raw something , better check manual
[15:22] <Compn> so muxing is when you put a codec into a container
[15:22] <grkblood13> yea, i grepped specially for "raw" and nothing nothign about speex
[15:22] <Compn> remuxing is moving a codec from one container (mp4) to another (mkv) etc
[15:22] <Compn> its a generic raw muxer
[15:22] <Compn> so it supports all codecs...
[15:23] <grkblood13> -f rawaudio fails for me
[15:23] <Compn> whats your command line
[15:23] <grkblood13> with speex atleast
[15:23] <grkblood13> also, rawaudio is not listed in my ffmpegs list of supported formats. rawvideo however is though.
[15:23] <Compn> ffmpeg -formats for a list
[15:24] <grkblood13> ya, thats what I did
[15:24] <Samus_Aran> can you just do -codec:a copy -vn file.speex ?
[15:24] <Samus_Aran> grkblood13: and which raw ones do you have listed?
[15:24] <grkblood13> Samus_Aran: its pretty lengthy. you want me to dump it in here?
[15:25] <grkblood13> https://gist.github.com/grkblood13/72122a9fd40b2bff6680
[15:26] <grkblood13> ^ supported raw formats
[15:26] <Compn> yeah dont see rawaudio
[15:26] <Compn> weird
[15:26] <Samus_Aran> on mine I see rawvideo, but not rawaudio
[15:26] <Compn> could use mplayer -dumpaudio to dump it
[15:27] <Compn> after its in .ogm or whatever
[15:27] <Compn> :)
[15:27] <grkblood13> I don't want to have to bring along mplayer. That's worse than what I'm doing right now in my opinion.
[15:28] <grkblood13> It's not difficult to cut out the speex data from the ogg container. I just don't want to do it if I don't have to as it seems very unnecessary.
[15:28] <Samus_Aran> grkblood13: did you try the copy line I suggested? might want to try .raw in case ffmpeg handles that.
[15:29] <grkblood13> gotta do a man on -vn first. I've never used that
[15:29] <eckesicle> Hi I'm returning today with the same problem as yesterday. avprobe_input_format2 returns null and avformat_open_file returns -1. Here's the relevant gist: https://gist.github.com/cfeckardt/0fc7f5e0c2e4cafd144c
[15:30] <grkblood13> -codec:a is the same as -acodec correct?
[15:33] <grkblood13> if I you copy I'll no longer be encoding with speex
[15:33] <grkblood13> s/you/use
[15:41] <Samus_Aran> <grkblood13> -codec:a is the same as -acodec correct? << yes
[15:42] <Samus_Aran> -vn means null for the video
[15:42] <Samus_Aran> -an is no audio, -sn for no subtitles
[15:43] <Samus_Aran> grkblood13: sorry, didn't realise you were encoding the speex with ffmpeg, thought it was just passing through your pipe
[16:32] <eckesicle> Okay so I have pinpointed my problem
[16:34] <eckesicle> avformat_open_input return AVERROR_INVALIDDATA (in av_probe_input_buffer lin 299 in libavformat/utils.c - I'm trying to load an mp4 file. What could be the problem?
[17:14] <k-s> is there an easy (or recommended) way to use an OpenGL FB as source for encoding? We have a compositor in EFL/Enlightenment and we'd like to generate screencasts from its output
[17:15] <k-s> we could have the x11 compositor to output in sw (rgb-pixels in RAM) but it would be much slower
[19:02] <Samus_Aran> night
[19:23] <pyBlob> Hello there!
[19:24] <pyBlob> I want to use ffmpeg and java to capture images from my webcam, this worked nice by using the system-exec-stuff to call ffmpeg from java, but there are some performance drawbacks
[19:24] <pyBlob> do you know any good ffmpeg bindings for java?
[19:25] <pyBlob> currently I've found jjmpeg and xuggle ... and both should work for my problem
[20:20] <lkiesow> pyBlob: What about just starting a separate process (execute the ffmpeg binary passing an appropriate commandline)
[20:21] <pyBlob> that's what I've already done
[20:21] <pyBlob> and it seems that this is the only thing that really works xD
[20:22] <lkiesow> And what is the performance drawback with that?
[20:22] <lkiesow> I'm an Opencast Matterhorn developer and that is exactly what we do, too :)
[20:25] <stieno> HI, I have an mp4 container containing 2 h264 video tracks. I'll want to stream those two streams at the same time via rtp to 2 different clients. A requirement is that they are played in sync at the client side. How can this be achieved? I tried with tee to split into 2 different rtp streams, but this failed.
[20:31] <stieno> please see http://pastebin.com/GWVtbXij for the error message I get.
[21:02] <pyBlob> ... I give up, using ffmpeg librariers with java on win64 doesn't seem to work, so I'll continue to just execute the ffmpeg binary
[21:04] <stieno> nobody can help me how to sync two video rtp streams on different clients?
[21:09] <pyBlob> is this a stereo-video-stream?
[21:09] <pyBlob> "[rtp @ 0x285b380] Only one stream supported in the RTP muxer"
[21:10] <stieno> no, actually I cut a video into 2 pieces, and want to display each piece on a different client. But they need to be in sync of course.
[21:12] <pyBlob> ok ... as you see, there are 2 streams in one mp4-file, and as the error-messages says, you can't send that to the clients using the RTP muxer
[21:12] <pyBlob> but you could try merging both streams into one bigger video: e.g. side-by-side
[21:13] <pyBlob> so there is only one stream with size 1024x576 instead of 2 streams
[21:14] <stieno> yes, the initial video file was one stream (1024x576). But I've split this into two pieces to send to 2 different clients. I do not want on the client side to decode the full resolution (1024x576), but only half of it (512x576).
[21:15] <pyBlob> or do you want to show piece 1 on client 1, and piece 2 on client 2?
[21:15] <stieno> yes, exactly
[21:16] <pyBlob> you used map, so you're halfway there ^^
[21:17] <pyBlob> try: ffmpeg -re -i output.mp4 -f rtp -map 0:1 -c copy rtp://localhost:5554 -f rtp -map 0:0 rtp://localhost:5556
[21:17] <pyBlob> oops, there's another -c copy missing for the second output
[21:18] <stieno> great! sending out the stream is working now! Are they sent out in sync?
[21:19] <pyBlob> they *should* be sent out in sync
[21:19] <stieno> Of do I need to modify timestamps in the rtp stream?
[21:20] <stieno> Ok. At the receiving side, is it obliged to use the sdp file, or is there another method not using the sdp file?
[21:21] <pyBlob> don't know about that :/
[21:32] <pyBlob> I'm using ffplay to view a video, why does the sound get muted, when the window isn't focuesd?
[21:36] <stieno> pyBlob, On a single pc it seems indeed to be in sync. I'll try later on 2 different pc's to see if they are still in sync. Thanks for the great help!
[21:42] <pzich> hmm, I'm trying to create a video (to play in HTML5) that has sections to loop until the user interacts to go to the next section. Is there support in MP4 for an index of seek positions, to speed up jumping to these predefined times? and is it possible to feed these into ffmpeg?
[21:47] <pyBlob> you might want to split the video into seperate files
[21:49] <pzich> pyBlob: hmm, we were hoping this would cut down on the number of network requests and inherent HTTP overhead
[21:50] <pyBlob> when using videos, the HTTP overhead shouldn't be that big compared to file size
[21:50] <pzich> yeah, there will just be a lot of videos, and there's more latency caused by making a request for each than having one, but it's probably still the better way to go
[21:51] <pyBlob> have you considered making use of browser caching and preloading?
[21:52] <pyBlob> another solution would be to use javascript to detect and set the looping position
[21:52] <pyBlob> (while the video is playing)
[21:53] <pzich> well right now I have a if (video.currentTime >= loopEnd) video.currentTime = loopBegin
[21:53] <pzich> I'm just trying to optimize seeking back to the beginning if possible
[21:56] <pyBlob> perhaps this helps too: http://stackoverflow.com/questions/8616855/how-to-output-fragmented-mp4-wit…
[22:03] <pzich> hmm, interesting, I'll see if fragments would help with seeking
[22:07] <Jellicent> Hey guys, I'm aiming to stream (twitch.tv) but I'm having a little sound problem. I'm using alsa and the sound is awfully low. Has anything like this been reported? Couldn't find anything like this.
[22:09] <Jellicent> By the way, I'm not talking about ingame sound and music for example. Microphone works fine.
[22:59] <bencc> is there a difference in performance on the client (browser) when redering a big video which shows several small videos and rendering separate small videos?
[23:00] <rager> number of threads involved for playing 1 video vs playing 4 videos is probably slightly different
[23:04] <bencc> what about 25 small videos vs 1 big video?
[23:04] <SLAYBoz> does ffmeg automatically convert all interlaced video to progressive?
[23:38] <pyBlob> I want ffmpeg to output it's data to a named pipe on windows, but it says the file doesn't exist:
[23:38] <pyBlob> http://pastebin.com/D1N6uHww
[00:00] --- Thu Dec 19 2013
1
0
[00:06] <cone-86> ffmpeg.git 03Diego Biurrun 07master:11bb5e10c365: build: Define __printf__ to __gnu_printf__ on MinGW*/gcc
[00:06] <cone-86> ffmpeg.git 03Michael Niedermayer 07master:a67387913eb0: Merge commit '11bb5e10c36539bcc303ceaac6f88d9ecb66e07f'
[00:37] <cone-86> ffmpeg.git 03Diego Biurrun 07master:0d882e9e7b78: avutil: Remove deprecated intfloat_readwrite code
[00:37] <cone-86> ffmpeg.git 03Michael Niedermayer 07master:196f7da7c818: Merge remote-tracking branch 'qatar/master'
[01:14] <cone-86> ffmpeg.git 03Michael Niedermayer 07master:ec464c96831a: avcodec/utils: av_register_codec & hwaccel() that work in O(1) time
[01:14] <cone-86> ffmpeg.git 03Michael Niedermayer 07master:49f10c9cb185: avformat/format: av_register_output_format() and av_register_intput_format() that work in O(1) time
[01:35] <Zeranoe> Is there any more information about pp=fq? http://www.ffmpeg.org/ffmpeg-all.html#pp it just says fq/forceQuant[|quantizer] without any options for quantizer. I'm trying to get fixed tables somehow
[01:37] <cone-86> ffmpeg.git 03Stoian Ivanov 07master:5fe415f0f415: UDP: join multicast group on the interface specified with &localaddr=
[01:38] <michaelni> Zeranoe, a number between 1 and 31 or so should work as quantizer
[01:40] <Zeranoe> michaelni: Thanks, will this help at all with "fixed rather than variable quantisation tables". Is there any option for that?
[01:44] <michaelni> iam not sure i understand your question
[01:47] <Zeranoe> michaelni: I'm back to the amv issue, and I'm trying to get a file working with a device. the amv format (http://en.wikipedia.org/wiki/AMV_video_format) call of fixed quanisation tables. I'm not sure why my settings don't work with some input videos, but other videos do work. I encode both videos with the same settings, so something must be passing through
[01:48] <michaelni> sounds unrelated to -vf pp
[01:49] <michaelni> you use qscale instead of bitrate ?
[01:49] <Zeranoe> michaelni: Yup
[01:51] <Zeranoe> michaelni: I guess the player has issues at the end of the file, I don't know if that narrows it down at all. I even tried dumping it to uncompressed and then encoding that, but still got the end error
[01:54] <Zeranoe> It seems that converting mp4 files downloaded from youtube will work on the device after being encoded with ffmpeg and amv-codec-tools. So something about those input files makes the output work, I have no idea why
[01:55] Action: michaelni has no idea either
[01:55] <Zeranoe> :(
[02:58] <cone-86> ffmpeg.git 03Michael Niedermayer 07master:745c40a47f88: avcodec/gifdec: check that the correct number of bytes was decoded
[03:47] <clever> Daemon404: you about?
[04:11] <Compnn> we should chip in to buy zeranoe a real mp4 player ,not some amv thing :D
[04:12] <clever> Compnn: i think i just figured out the codec config problems
[04:13] <clever> all packets that omxplayer sends in, are prefixed with a 4 byte size
[04:13] <clever> my initial checks for that kind of thing where thrown off, because the first 'packet' actualy contained 2 such packets
[04:17] <clever> now i just have to deal with byte order issues...
[04:34] <clever> Compnn: i think its working, but now i'm back to a corrupt stack!
[05:00] <Compnn> using str versions of functions ?
[05:00] <Compnn> did you dump omxplayer to assembler and just reverse engineer that ?
[05:08] <clever> Compn: i just compiled omxplayer from source
[05:08] <clever> *** glibc detected *** /160g/rpi/mplayer/mplayer: double free or corruption (out): 0xb0f3d020 ***
[05:11] <clever> still trying to figure this out now, ive been over the code multiple times, and it should only have freed that pointer once
[05:11] <Compn> grep -i free *
[05:11] <clever> i already did a search in the file, only one line ever deals with at pointer and free
[05:11] <clever> av_free(inframe->v);
[05:11] <clever> no other line anywhere mentions the bad pointer and free
[05:12] <Compn> what if you remove that line
[05:12] <clever> and acording to gdb, it crashes the first time i run that line
[05:12] <clever> i'll comment it out and see what happens
[05:12] <Compn> you're thinking of a 'double free'
[05:12] <Compn> oh nm
[05:12] Action: Compn cant read lol
[05:13] <clever> v is allocated in 2 places
[05:13] <clever> decoder->frame->v = av_malloc(newframe->vstride * (decoder->height/2));
[05:13] <clever> decoder->frame->v = av_malloc(pic->f.linesize[2] * (decoder->height/2));
[05:13] <clever> both look fine to me
[05:18] <clever> Compn: with the free commented out it works
[05:18] <Compn> of course it works
[05:18] <Compn> because i am genius
[05:18] <clever> and its likely leaking a whole v plane on every frame
[05:18] <clever> 500kb per frame, 30 fps
[05:18] <Compn> likely ?
[05:18] <Compn> we dont use words like that here
[05:18] <Compn> we use words like valgrind
[05:19] <clever> and valgrind uses words like 'wtf is omx, that memory was never initialized!'
[05:19] <clever> and 'oh, let me eat all you disk space with debug symbols'
[05:20] <Compn> well then run top when its playing
[05:20] <Compn> double free means its being freed elsewheres
[05:20] <clever> except its imposible for it to be free'd anywhere else
[05:20] <clever> valgrind loading up...
[05:21] <clever> ==3087== Conditional jump or move depends on uninitialised value(s)
[05:21] <clever> ==3087== at 0x4837A54: memcpy (mc_replace_strmem.c:838)
[05:21] <clever> ==3087== by 0x4AE159B: ilcs_pass_buffer (in /opt/vc/lib/libopenmaxil.so)
[05:21] <clever> all over the place! :D
[05:22] <clever> --3087-- VALGRIND INTERNAL ERROR: Valgrind received a signal 11 (SIGSEGV) - exiting
[05:22] <clever> ok, that didnt go very well
[05:22] <Compn> ehe
[05:22] <Compn> impossible you say
[05:22] <clever> lets see, 48mb
[05:23] <clever> 57mb
[05:23] <clever> 63mb
[05:23] <clever> yeah, its leaking!
[05:23] <Compn> oom is fun :P
[05:23] <clever> except when you have swap on SD cards
[05:24] <clever> let me double check the log files i just made...
[05:24] <clever> copying 0xb13db220(960) to 0x35d6cb0(976), height 536
[05:24] <clever> copying 0x35d6cb0(976) to 0xb540eea8(976), height 536
[05:24] <clever> deleting frame 0x2840040 y0xb0061020 u0x30d97d0 v0x35d6cb0
[05:24] <clever> ok, it populates the v plane, then copies the v plane to ffmpeg, then tries to release it
[05:24] <clever> and never mentions the same pointer again
[05:26] <clever> so its not a double-free when its commented out, and nothing is reusing frames
[05:27] <Compn> are you using memcpy or av_memcpy ?
[05:27] <Compn> av_fastmemcpy
[05:27] <Compn> whatever
[05:27] <clever> memcpy, via libcofi_rpi.so
[05:27] <clever> which i have to comment out, valgrind claims its an illegal instruction
[05:27] <Compn> you do what now
[05:27] <Compn> lol
[05:28] <clever> its a special lib that replaces the stock memcpy with one that gets much better performance
[05:28] <clever> some kind of broadcom extension
[05:28] <clever> which valgrind doesnt know about
[05:28] <Compn> could you use standard memcpy and see if the free works then ? or is it completely diff ?
[05:28] <clever> the standard memcpy also works, just wont be as fast
[05:29] <clever> but i'm already getting 2 fps, so who cares!
[05:29] <Compn> i'm just guessing at straws, so my advice isnt great :)
[05:29] <clever> https://github.com/simonjhall/copies-and-fills has more info
[05:30] <clever> stock memcpy, a single free call, and it screams double-free
[05:30] <Compn> you sure ffmpeg isnt freeing somewhere
[05:31] <Compn> because ffmpeg does that
[05:31] <Compn> lots of free
[05:31] <clever> i never pass ffmpeg a copy of this pointer
[05:31] <clever> its a custom frame struct, to hold things between the async callback and the avframe
[05:32] <clever> the only time it leaves my file is memcpy, printf, and av_free
[05:33] <Compn> are you free'ing inframe or v ?
[05:33] Action: Compn dont know coding
[05:33] <clever> both, in order
[05:33] <clever> Compn: http://pastebin.com/CsGkuPWv
[05:34] <Compn> oh easy
[05:34] <Compn> put the last free above the first 3 :)
[05:35] <clever> then the memory for the struct will be 'free' and another thread may overwrite the pointers
[05:35] <clever> before i have a chance to free them properly
[05:35] <clever> and its c, it isnt capable of recursively releasing a complex struct
[05:35] <clever> thats the kind of hidden bug that valgrind is great for
[05:38] <Compn> well im out of idears
[05:38] <Compn> wait for smart people :)
[05:39] <Compn> gdb wont tell you where the second free is ?
[05:39] <Compn> or is corrupt
[05:39] <clever> gdb wont even give me a backtrace
[05:39] <clever> corrupt stack it claims
[05:39] <clever> and free is used too much to just put a breakpoint on it, thats what valgrind is for
[05:43] <clever> Compn: ok, i'll just disable the entire frame mess!
[06:07] <clever> Compn: there, it now works without my double buffering hack
[06:07] <clever> and yay!!, it works on other files
[06:07] <clever> without having to recompile!
[06:09] <clever> and 720 is almost at a usable fps
[06:38] <Guest61581> Is there some kind of prioritization system for the Trac issues?
[06:50] <clever> Compn: getting 20 fps now!, but there is some frame drop
[06:50] <clever> and thats with a software pixel format change!
[06:50] <clever> fbdev doesnt like yuv420, and i havent fixed it yet
[06:54] Action: Compn afk
[06:55] <clever> A: 71.8 V: 66.5 A-V: 5.277 ct: 0.430 0/ 0 20% 68% 14.9% 1868 0
[06:55] <clever> ok, thats enough for one night
[06:55] <clever> except for the frame drop and xv replacement, its nearly done
[06:57] <clever> and ass subs are fully working
[06:57] <clever> hard to say how well it will run once scaled up, but its looking good
[06:58] Action: clever heads to bed
[10:18] <llogan> saste: any reason why @code is used for options and such instead of @option in texi?
[10:21] Action: llogan goes to bed but will look for replies, if any, in the morning
[10:26] <saste> llogan, where?
[11:07] <saste> [h264 @ 0x23ec080] non-existing PPS referenced
[11:07] <saste> what does this ^^ mean?
[11:44] <michaelni> saste, that a slice refers to a pps that doesnt exist
[11:45] <saste> michaelni, what is a PPS?
[11:45] <michaelni> picture parameter set
[11:46] <saste> michaelni, alright, but can you say why it happens?
[11:46] <saste> for the long story, see my mail on ffmpeg-devel
[11:47] <nevcairiel> usually such messages only show up briefly at the beginning of a badly-cut stream, when the decoder has not seen the PPS yet
[11:47] <michaelni> pps wont be repeated for every slice so when you start decoding at a random point you can run into slices that refer to a pps that hasnt bee seen
[11:48] <saste> michaelni, even if the starting packet is a key-packet?
[11:49] <saste> what's wrong with my assumption: key packet -> decodable I frame
[11:49] <nevcairiel> just because a packet contains a keyframe, does not mean the SPS/PPS are necessarily available yet. Ideally they should always preceed a I/IDR frame, but thats no guarantee
[11:51] <michaelni> also you can use -debug 1 to see whats in the h264 stream from the decoders point of view
[11:51] <saste> nevcairiel, so basically there is no reliable way to split a H.264 stream at key-packets?
[11:51] <nevcairiel> you could go digging for the PPS in question and manually ensure its present at the cut point
[11:53] <saste> nevcairiel, yes but not very practical
[11:53] <nevcairiel> in many streams the SPS/PPS is the same through the whole stream
[11:53] <michaelni> or just dump the pps&sps before the key frame you choose
[11:53] <nevcairiel> so you could just chekc extradata
[11:54] <nevcairiel> and put it into the stream
[11:54] <saste> can this be done programmatically with some already existing bsf?
[13:07] <kierank> 10:50 AM <"nevcairiel> just because a packet contains a keyframe, does not mean the SPS/PPS are necessarily available yet. Ideally they should always preceed a I/IDR frame, but thats no guarantee
[13:07] <kierank> they should for an IDR otherwise it's out of spec
[13:08] <nevcairiel> as if everyone produces files that are correct
[13:09] <iive> I think that they do for IDR, but IDR is rarely used.
[14:08] Action: ubitux wonders why trac makes any difference between search & query
[14:09] <ubitux> one is completely useless and the most simple to find
[14:09] <ubitux> and the other is pretty relevant and hard to find
[14:30] <michaelni> ubitux, i wondered about that too
[14:30] <michaelni> someone should maybe improve the user interface
[14:30] <ubitux> i don't actually even know how to access the /query link
[14:30] <ubitux> i just have it in my history since carl pointed it out
[14:31] <michaelni> view tickets, custom query
[14:31] <michaelni> but its impossible to find if one doesnt know where to look
[14:31] <ubitux> mmh ok
[14:32] <michaelni> also if someone has a patch to some template or ini file for trac to improve this ... send it to me
[14:41] <cone-410> ffmpeg.git 03Michael Niedermayer 07master:6aed7bfd8453: avcodec: use av_frame_unref() to set frame defaults
[14:41] <cone-410> ffmpeg.git 03Michael Niedermayer 07master:5abdda214df5: avcodec/utils: implement avcodec_alloc_frame() through av_alloc_frame()
[15:52] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:c5f7c8f9a9a1: avcodec/svq3: set last/next frames to defaults before use
[15:52] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:fca7943850ec: avcodec/diracdec: avoid depending on sizeof(AVFrame)
[16:54] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:c81234651f76: avcodec/msvideo1enc: drop dependancy on sizeof(AVFrame)
[16:54] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:5b3f4b3ef590: avcodec/mjpegenc: drop dependancy on sizeof(AVFrame)
[16:54] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:1458f0647ca0: avcodec/libopenjpegenc: drop dependancy on sizeof(AVFrame)
[17:18] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:8443b27072a0: avcodec/j2kenc: drop dependancy on sizeof(AVFrame)
[17:18] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:b8f4410ff60b: avcodec/flashsv2enc: drop dependancy on sizeof(AVFrame)
[17:18] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:bf1c87ee7ab1: ffprobe: drop dependancy on sizeof(AVFrame)
[17:38] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:3c8b085764ed: avcodec/libvorbisenc: drop dependancy on sizeof(AVFrame)
[17:38] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:c90f31146e8b: avcodec/utils: drop 2 dependancies on sizeof(AVFrame)
[18:09] <cone-446> ffmpeg.git 03Peter Holik 07master:6fd99e78def3: png: add a standalone parser
[18:09] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:47cc61686494: Merge commit '6fd99e78def3c795bdd0bc31f3ae0998d24bc94c'
[18:14] <cone-446> ffmpeg.git 03Vittorio Giovara 07master:eddff165b4cd: fate: add utility function to test parser, demuxer, and decoder
[18:14] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:b22989da47be: Merge commit 'eddff165b4cdc1f064245a4bad6f4265581c12b1'
[19:04] <cone-446> ffmpeg.git 03Vittorio Giovara 07master:cfb4ee309777: fate: add a pngparser test
[19:04] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:682b28b41098: Merge commit 'cfb4ee30977732674d30c20e93a761c33c743972'
[19:11] <cone-446> ffmpeg.git 03Luca Barbato 07master:3a149e23d37f: hevc: Refactor decode_nal_sei_frame_packing_arrangement
[19:11] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:4e1dc600a16e: Merge commit '3a149e23d37f06d49ab0bb013a3b9cb0224bba5c'
[19:35] <llogan> saste: in decoders.text for example: @code{--enable-lib}
[19:35] <llogan> texinfo says: you should use @code for an expression in a program, for the name of a variable or function used in a program, or for a keyword in a programming language.
[19:38] <cone-446> ffmpeg.git 03Vittorio Giovara 07master:3d9fc21e593a: h264: Refactor decode_frame_packing_arrangement
[19:38] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:d6acf45ec280: Merge commit '3d9fc21e593aa89b5877bd739268499b079052f2'
[19:54] <cone-446> ffmpeg.git 03Vittorio Giovara 07master:bd316109b333: apidoc: fix warning from stereo3d.h
[19:54] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:9be63be1b285: Merge remote-tracking branch 'qatar/master'
[19:58] <saste> llogan, what do you suggest to use instead?
[19:58] <saste> also i consider the option in a script as a sort of "expression in a program"
[19:59] <saste> the real alternative is probably @option{...}
[20:03] <llogan> saste: yes, @option
[20:03] <saste> llogan, patch welcome :)
[21:13] <dorileo> hi guys... I need to determine if an mpeg ts file's packet size is TS_PACKET_SIZE || TS_DVHS_PACKET_SIZE || TS_FEC_PACKET_SIZE but as far as I could see mpegts.c doesn't expose it anywhere, is there any "correct" way to get this information?
[21:14] <michaelni> its found during probing but IIRC not exportet
[21:15] <dorileo> btw, AVContextFormat structure has a packet_size field but I couldn't see where it's set as well, I'm thinking about sending a patch changing mpegts.c to set raw_packet_size to AVContextFormat->packet_size.
[21:15] <dorileo> michaelni: do you know if AVContextFormat->packet_size is set anywhere? I couldn't find it.
[21:16] <dorileo> michaelni: what do you think about setting AVContextFormat->packet_size = MpegTSContext->raw_packet_size ?
[21:24] <michaelni> probably possible, someone would have to test it and submit a patch, alternatively it could be exportet as private AVOption
[21:24] <dorileo> michaelni: sounds better using AVOption.
[21:25] <dorileo> michaelni: I can cook it up.
[21:25] <michaelni> ok
[21:41] <dorileo> michaelni: by "exported as private AVOPtion" do you mean setting adding it to AVFOrmatContext's metadata dict?
[21:49] <dorileo> michaelni: don't worry, I got it.
[22:08] <cone-446> ffmpeg.git 03Michael Niedermayer 07master:3e626548ce6d: avcodec/vc1dec: use av_mallocz for luma_mv
[23:56] <kierank> what a stupid discussion
[23:56] <kierank> it's clear ttx in wtv is not working but carl wants him to use a transport stream
[00:00] --- Wed Dec 18 2013
1
0
[00:29] <bri> does anyone know if features from ffmbc would ever be merged into ffmpeg?
[00:31] <saste> bri, ffmbc is GPL, so we can't merge its features in ffmbc without relicensing
[00:31] <saste> in practice it means that all feature should be implemented in ffmpeg from scratch
[00:31] <iive> only baptiste could say that.
[00:31] <bri> i noticed that. (at least in comparing the mxf demux/mux)
[00:32] <bri> is it permissible to change the license like that?
[00:32] <saste> not without original author explicit consent
[00:33] <bri> weird.
[00:37] <bri> i know there's definitely some good features in there, like proper support for reading opatom mxf
[00:42] <klaxa> licensing is weird, right? :)
[00:44] <bri> klaxa: yes. i don't like dealing with it ;(
[04:01] <llogan> DeadSix27: -g
[06:00] <SLAYBoz> i was just told that faac license is not GPL compatible so why does ffmpeg include faac
[06:18] <grepper> it and various other libraries are included if you use --enable-nonfree
[06:33] <allengreen> how can I force the rtsp's rtp use tcp only?
[15:21] <eckesicle> Hello, I'm getting an "Operation Not permitted" error when calling avformat_open_input on an mp4 file. I do not believe it is a permissions error because I am able to read and write to the same file using stdio. What could cause this?
[15:22] <ubitux> return -1 in the code
[15:22] <eckesicle> Yes the method returns -1
[15:23] <ubitux> it's often mapped to ENOPERM
[15:23] <ubitux> but it was just meant to be a negative return value originally
[15:23] <ubitux> can you share a sample?
[15:24] <ubitux> does it happen with ffmpeg/ffprobe?
[15:24] <eckesicle> i am developing using libavforamt
[15:26] <eckesicle> https://gist.github.com/cfeckardt/0877ce3b15c26fc06e00
[15:27] <koljjhi> In trancoding a DV file to MPEG the resulting left audio channel has a buzz. How do I fix that?
[15:27] <koljjhi> Here's the code I use- $ffmpeg -i Input.DV -target ntsc-dvd Output.mpeg
[15:28] <eckesicle> ubitux: it could well be that I am missing some library that needs to be linked with libavformat to read mp4 files
[15:28] <ubitux> libavformat has native support for mp4, that's not it
[15:29] <ubitux> your ffmpeg should use the same libavformat, that's why i'm asking if the probing work with it
[15:29] <ubitux> it's just a first step to figure out if it's a problem with the sample, or with your code
[15:29] <eckesicle> the file's on an embedded device
[15:29] <eckesicle> a camera to be specific
[15:30] <eckesicle> but ill see if i can extract a sample file
[15:30] <eckesicle> ffmpeg is compiled with minimum libraries
[15:31] <ubitux> check if it has mp4/mov demuxer then
[15:31] <eckesicle> --enable-decoder=mpeg4 \
[15:31] <ubitux> demuxer, not decoder
[15:31] <JEEB> if you have a minimum config within libav* then make sure you have all of the protocols you need enabled, file among others, as well as mov demuxer and various other parsers you might need
[15:31] <eckesicle> no I don't see it in the compilation options
[15:32] <ubitux> --enable-demuxer=mov
[15:33] <JEEB> eckesicle, just see the list you get after you run configure with your options, if it starts lacking stuff then you know you're disabling too much :P
[15:33] <JEEB> also this is one of the problems I see the most
[15:33] <JEEB> you start minimizing your build BEFORE you actually have the code working
[15:33] <eckesicle> hmm we have to
[15:34] <eckesicle> have very little memory to spare
[15:34] <JEEB> that makes it more understandable but you should really keep an eye on what's actually enabled in your builds :P
[15:34] <eckesicle> :D
[15:35] <JEEB> you have no idea how many people have had similar problems because of just disabling features they actually needed :D
[15:35] <eckesicle> :)
[15:35] <eckesicle> anything else i need for mp4 files with x264?
[15:35] <JEEB> the h264 parser and decoder, mp4 and mov demuxers, file and other protocols
[15:37] <eckesicle> what's the command for parser
[15:37] <eckesicle> enable-parse=h264 _
[15:37] <eckesicle> ?
[15:37] <eckesicle> enable-parser=h264
[15:37] <JEEB> just look at the configure line when you run it without any options :P
[15:37] <JEEB> as in the output of it
[15:37] <JEEB> it lists all the things buildable with LGPLv2
[15:38] <eckesicle> ah ok
[16:11] <eckesicle> Nope
[16:11] <eckesicle> didn't work
[16:57] <eckesicle> Okay so I enabled all relevant, demuxers, decoders and parsers
[16:58] <eckesicle> still getting -1 returned when calling avformat_open_input
[16:58] <eckesicle> https://gist.github.com/cfeckardt/6cac7cd854c098654cd2
[17:21] <eckesicle> Hmm
[17:21] <eckesicle> it segfaults on av_probe_input_format2 too.
[18:04] <DrSlony> Hi, I want to transcode a "-preset ultrafast" screencast recording to "-preset slower" and I want to set the framerate at 15fps, where do I put the -r? before the -i or after? does it make a difference?
[18:04] <DrSlony> code: http://paste2.org/YEAVY7zD
[18:04] <sacarasc> It's an output option, so you put it after.
[18:12] <DrSlony> thank you
[18:13] <DrSlony> Is there anyone here who would like to take a shot at stabilizing (deshaking) a video clip?
[20:32] <caribou|> hello
[20:33] <caribou|> i'm trying to make a video out of a sequence of images and i have troubls matching files like screen_1387009721.5.jpg screen_1387009731.51.jpg screen_1387009741.51.jpg screen_1387009751.5.jpg
[20:33] <caribou|> (this is the sequence)
[20:34] <caribou|> i think i can't use the sequence pattern so probably with glob_sequence ?
[20:34] <caribou|> (as i can't use glob on windows)
[20:34] <caribou|> again i tried stuff but it doesn't seem to work
[20:36] <Compn> start renaming it :)
[20:36] <Compn> if you are on windows, i recommend oscars' renamer program
[20:36] <Compn> fairly easy to use
[20:39] <caribou|> i wish i could have done it with ffmpeg but yeah i guess you removed the potential hassle of finding a proper rename software
[20:39] <caribou|> i'll try it, thanks :)
[20:41] <Compn> but uh
[20:41] <Compn> what is your command line caribou| ?
[20:41] <Compn> there is a special image input option to fix image names like that
[20:42] <Compn> %3d or something
[21:20] <llogan> caribou|: just use the glob pattern
[21:20] <llogan> http://trac.ffmpeg.org/wiki/Create%20a%20video%20slideshow%20from%20images
[21:20] <llogan> http://ffmpeg.org/ffmpeg-formats.html#image2-2
[21:21] <llogan> no need to rename
[22:04] <caribou|> Compn, isn't %3d just 3d digits ?
[22:04] <caribou|> i have [fixed digits number].[one|two(digits)]
[22:05] <caribou|> llogan, i don't have the glob pattern
[22:05] <caribou|> only glob_sequence :/
[22:24] <klaxa> is %3d even a valid strint format?
[22:25] <klaxa> it is
[22:25] <klaxa> *string even
[22:25] <DeadSix27> yep
[22:26] <DeadSix27> Mainly used for image sequences (gets replaced by an integer), in fact i don't know any other usage.
[22:29] <klaxa> well it paddes with spaces so that's bleh
[22:29] <DeadSix27> shouldnt
[22:29] <klaxa> padding with zeros is smarter and that would be %03d
[22:29] <DeadSix27> the 3d, makes it zero appended
[22:30] <DeadSix27> i mean
[22:30] <klaxa> %03d prepends zeros %3d prepends spaces
[22:30] <DeadSix27> yes.
[22:30] <DeadSix27> theres also the option, -start_number
[22:30] <DeadSix27> to define from where it starts.
[22:30] <DeadSix27> counting
[22:31] <DeadSix27> in case you dont want 0.
[22:31] <DeadSix27> or want 0 w/e
[22:48] <Overflip> Unknown option "--enable-libdirac". using the current ffmpeg version. Does anyone know the last version that his library is supported?
[22:50] <braincracker> lemon juice is good with salt ?
[22:53] <Compn> Overflip : current, you mean git ?
[22:53] <Compn> dirac should still be there
[22:53] <Overflip> ffmpeg-2.1.1
[22:53] <Overflip> not from git
[22:54] <Compn> oh nevermind
[22:54] <Overflip> I'm about to try 1.2.4
[22:54] <Compn> it was removed by libav (and remove-merged to ffmpeg) in 2012
[22:54] <Compn> 2012-04-24
[22:54] <Compn> get an ffmpeg before that date
[22:55] <Compn> or maybe michaelni will undo that merge ?
[22:55] <Overflip> ty =)
[23:06] <Overflip> what about libx264?
[23:08] <Overflip> ah nm
[23:20] <Overflip> libavcodec/libavcodec.a(libx264.o): In function `X264_init'
[23:20] <Overflip> ffmpeg-2.1.1/libavcodec/libx264.c:569: undefined reference to `x264_encoder_open_138'
[23:20] <sacarasc> That sounds like you have 2 copies of x264 installed.
[23:29] <Overflip> well I'm getting ERROR: libx264 not found
[23:29] <Overflip> with running ./configure
[23:29] <Overflip> which x264
[23:30] <Overflip> shows it in /usr/local/bin/x264
[23:32] <klaxa> you need libx264 in a place where configure finds it or you tell configure where to look for it
[23:32] <klaxa> regardless you need libx264 (apparently libx264.so.138)
[23:33] <Overflip> how do I tell configure where it is?
[23:34] <klaxa> --extra-ldflags=-L/path/to/lib/dir --extra-cflags=-I/path/to/include/dir
[23:34] <klaxa> those directories hold libx264.so and x264.h respectivley
[23:35] <Overflip> thanks I'll give that a try
[23:45] <Vandalite> has anyone ever run into a situation where ffmpeg works perfectly fine when running as the root user, but fails with no clear error when run as another user on the same system?
[23:47] <Overflip> I can only find x264.o and x264.h, do I need to rename thos?
[23:48] <Vandalite> I mean, ffmpeg runs, it appears to be able to read the file, but breaks setting up the encoders with: Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
[23:51] <Overflip> ya I'm confused
[23:53] <Overflip> How do I tell ffmpeg where x264 is installed?
[00:00] --- Wed Dec 18 2013
1
0
[01:25] <cone-75> ffmpeg.git 03Yu Xiaolei 07master:1c67ad9d9303: swscale: NEON optimized unscaled rgba to nv12 conversion
[02:27] <cone-75> ffmpeg.git 03Michael Niedermayer 07release/1.2:414b37746274: swscale/utils: remove useless ()
[02:27] <cone-75> ffmpeg.git 03Michael Niedermayer 07release/1.2:af7cbdf470e4: swscale/utils: check chroma width for fast bilinear scaler
[02:27] <cone-75> ffmpeg.git 03Michael Niedermayer 07release/2.0:9de71b0eb2d2: swscale/utils: remove useless ()
[02:27] <cone-75> ffmpeg.git 03Michael Niedermayer 07release/2.0:bd339d488285: swscale/utils: check chroma width for fast bilinear scaler
[02:27] <cone-75> ffmpeg.git 03Michael Niedermayer 07release/2.1:5b52b4962e22: swscale/utils: remove useless ()
[02:27] <cone-75> ffmpeg.git 03Michael Niedermayer 07release/2.1:85ea84658021: swscale/utils: check chroma width for fast bilinear scaler
[03:28] <cone-75> ffmpeg.git 03James Zern 07master:7a994d777a0a: configure: (vp9) ensure bitstream compatibility
[03:28] <cone-75> ffmpeg.git 03Tomer Barletz 07master:fe285b04bbad: avcodec/mpeg12dec: Don't create side data when current picture ptr is NULL.
[03:54] <cone-75> ffmpeg.git 03Michael Niedermayer 07master:f618cb1a4be8: swscale/x86/rgb2rgb_template: try to fix build failure with avx disabled
[04:44] <cone-75> ffmpeg.git 03Michael Niedermayer 07master:4ecac816780d: avformat/psxstr: zero packet to prevent uninitialized data to leak through to the decoder
[13:42] <sdr_> Hi all, I'm trying to submit a patch to avformat/udp.c whos maintainer is Luca Abeni, but he seems inactive in the ffmpeg-devel mailing list. How shall I proceed? Keep waiting, resubmit?
[13:51] <saste> sdr_, send it to the mailing list
[13:51] <saste> ping it if you got no reply
[13:53] <sdr_> saste, It's in the mailing list http://ffmpeg.org/pipermail/ffmpeg-devel/2013-December/151938.html but I got no replay. What do you mean by "ping"?
[13:53] <nevcairiel> answer your own mail with the text "ping"
[13:53] <nevcairiel> :)
[13:54] <sdr_> Ok I'll ping it
[13:56] <TimNich> Ooooh VLC Rincewind won't play ffv1 files that Twoflower will&.
[14:29] <BBB> ubitux: also a prototype of 4x4 sub16x16idct on github, it's a little broken and 2x2 will come soon
[14:29] <BBB> ubitux: so I can almost work on the crasher in trac and parallelmode bug
[14:36] <cone-86> ffmpeg.git 03Michael Niedermayer 07master:e2742d6ad695: avcodec/wmalosslessdec: completely initialize contains_subframe
[15:25] <cone-86> ffmpeg.git 03Michael Niedermayer 07master:a8d702859b8b: avcodec/mlpdec: check that streamtype matches the codec
[16:45] <ubitux> BBB: oh, cool
[16:46] <ubitux> BBB: a ticket was opened recently
[16:46] <ubitux> i think it's related to frame parallel as well
[16:46] <ubitux> https://trac.ffmpeg.org/ticket/3228
[17:09] <cone-86> ffmpeg.git 03Torsten Jager 07master:48d39c8786d2: avcodec/libx264: Set colorspace details correctly
[18:26] <cone-86> ffmpeg.git 03Stefano Sabatini 07master:023693d786c2: lavfi/buffersrc: set min and max values for pix_fmt option
[19:21] <cone-86> ffmpeg.git 03Stefano Sabatini 07master:e3eb0988c662: lavu/dict.h: extend/clarify av_dict_get() doxy
[19:21] <cone-86> ffmpeg.git 03Stefano Sabatini 07master:33f10fa65706: ffserver: fix spacing typoin error message
[19:21] <cone-86> ffmpeg.git 03Stefano Sabatini 07master:9985710a5a92: ffserver: add stream Metadata option
[19:21] <cone-86> ffmpeg.git 03Stefano Sabatini 07master:1d6666a6b828: ffserver: extend report_config_error() facilities, and use it for warnings
[22:04] <cone-86> ffmpeg.git 03Reinhard Tartler 07master:3bc2e89c76e8: Bump libavutil major version to account for the LLS API/ABI changes.
[22:04] <cone-86> ffmpeg.git 03Michael Niedermayer 07master:792845e436dd: Merge commit '3bc2e89c76e88ae6f1fd5287e0b11abcfc3c601c'
[23:26] <cone-86> ffmpeg.git 03Reinhard Tartler 07master:6d3ea1957f68: Bump major of libavfilter for reference counted buffer API changes.
[23:26] <cone-86> ffmpeg.git 03Michael Niedermayer 07master:a788949255d5: Merge commit '6d3ea1957f681b3bf9c752e6d21a501cc8d4180d'
[00:00] --- Tue Dec 17 2013
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[00:08] <relaxed> farsheed: you'll have to compile ffmpeg yourself for that support.
[00:08] <relaxed> https://trac.ffmpeg.org/wiki/UbuntuCompilationGuide
[04:13] <hailwood> Hi guys, I am executing a command and the output directory is chmodded to 777, and idea why when running the command I would be getting a permission denied error?
[04:16] <hailwood> here is the actual command: http://pastebin.com/Ae6BqXpW
[04:19] <relaxed> first of all, what version of ffmpeg is this?
[04:41] <Compn> run it as root :P
[04:59] <liam-kelly> I am having some issues setting up ffserver, with ffmpeg as the source, http://pastebin.com/sD3Gnvq1. I have also been following the tutorial here https://trac.ffmpeg.org/wiki/Streaming%20media%20with%20ffserver
[04:59] <liam-kelly> What am I missing?
[05:30] <braincracker> hi
[05:30] <braincracker> i should compile something, because my psu exhales cold water
[10:42] <XwZ> hi, I try to open a camera with libav, it's working with ffmpeg -i /dev/video0 but I don't know how to open it with avformat_open_input, if I open it directly, I get a segfault and if I set as input file "video=Camera name" I get the error -20
[10:44] <XwZ> then how can I open it ?
[10:47] <relaxed> XwZ: https://trac.ffmpeg.org/wiki/How%20to%20capture%20a%20webcam%20input
[10:49] <XwZ> relaxed: thank you I want to use it in my software, with avformat_open_input not with the ffmpeg command, I already did.
[10:50] <relaxed> oh, can't help you there
[10:51] <relaxed> see how ffmpeg does it
[10:51] <XwZ> yes it was my idea I will look at it
[10:52] <XwZ> but find the doc is better than reverse engeneering for me
[11:40] <benbro1> I'm trying to create a video from images with
[11:40] <benbro1> ./ffmpeg -r 25 -i frames/frame%d.png -c:v libx264 -r 25 -pix_fmt yuv420p out.mp4
[11:40] <benbro1> getting an error: Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
[11:44] <relaxed> add -f image2 before the input
[11:46] <benbro1> ./ffmpeg -r 25 -f image2 -i frames/frame%d.png -c:v libx264 -r 25 -pix_fmt yuv420p out.mp4 ?
[11:47] <relaxed> yes
[11:47] <relaxed> how are the images named?
[11:47] <benbro1> frame1.png, frame2.png.... frame10.png...
[11:47] <benbro1> better to use frame0001.png ?
[11:48] <relaxed> frame%d.png should work
[11:48] <relaxed> or, ./ffmpeg -r 25 -f image2 -pattern_type glob -i frames/*png ...
[11:50] <benbro1> ok. tahnks
[11:50] <benbro1> thanks
[11:58] <benbro1> relaxed: the image width was 1069. when I use 1070 I don't see the error
[11:59] <relaxed> by padding it?
[11:59] <benbro1> yes
[11:59] <benbro1> no, sorry
[12:00] <benbro1> I was taking a screenshot so I just take it with 1 more pixel
[12:00] <relaxed> that works too
[12:00] <benbro1> what's wrong with 1069?
[12:00] <relaxed> it's not /2
[12:01] <benbro1> ok
[12:01] <benbro1> in avidemux I'm getting "Index is not up to date"
[12:01] <benbro1> I was trying with and without "-movflags faststart"
[12:01] <relaxed> command?
[12:02] <benbro1> ./ffmpeg -r 25 -i frames/frame%d.png -c:v libx264 -r 25 -pix_fmt yuv420p -movflags faststart out.mp4
[12:02] <benbro1> I can paste the full output if needed
[12:02] <relaxed> is it? -movflags +faststart
[12:03] <benbro1> trying
[12:04] <benbro1> same index issue
[19:32] <ufa> hello :)
[19:33] <ufa> I want to dump audio from a mkv chaptered file into multiple mp3 files
[19:33] <ufa> Is it possible?
[21:22] <Zeranoe> I have a device that plays avi files, mjpeg and adpcm_ima_wav. I'm trying to match the sample avi the device came with, but everything I encode with ffmpeg results in a format error. Is there something like baseline for mjpeg? I don't know how to make it more compatible.
[22:20] <DeadSix27> what option would prevent video feels from having "giant seek positions" ?
[22:20] <DeadSix27> files*
[22:20] <DeadSix27> i often have files that can only seek between 5secs.. instead of every sek or less
[00:00] --- Tue Dec 17 2013
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