[Ffmpeg-devel-irc] ffmpeg.log.20150106
burek
burek021 at gmail.com
Wed Jan 7 02:05:01 CET 2015
[01:47] <pagios> how can i merge two mp4 files?
[02:10] <c_14> define "merge"
[02:10] <c_14> Do you mean concatenate?
[02:11] <xreal> Is MP4 a streaming format? When writing to MP4, I can't open the file the same time in Windows. When writing to MKV, it works without a problem.
[02:14] <pagios> c_14: yes concat
[02:14] <pagios> c_14: i have 2 files that i would like to combine into
[02:14] <pagios> 1
[02:15] <c_14> https://trac.ffmpeg.org/wiki/Concatenate
[02:16] <c_14> xreal: not sure what you mean with streaming format, but you can only play an MP4 once the entire file has been written
[02:16] <c_14> The link was for pagios
[02:16] <xreal> c_14: With "streaming format" I mean: read while writing to it. I thought, MP4 is capable of it.
[02:16] <c_14> nope
[02:16] <xreal> c_14: So MKV is a better way to use?
[02:17] <c_14> yep
[02:17] <c_14> Matroska is better at everything, except for compatibility.
[02:19] <pagios> c_14: is aac installed by default with ffmpeg?
[02:19] <pagios> the file is h264/aac
[02:19] <xreal> c_14: Yeah, that's my main problem... my DVD player can't play MKV, but my Raspberry can :)
[02:20] <c_14> pagios: yep, ffmpeg has an internal encoder/decoder
[02:20] <pagios> oh
[02:20] <pagios> libav does not
[02:20] <muken> xreal: mp4 "can" be streamable
[02:21] <c_14> muken: are you talking about fragmentation?
[02:21] <muken> c_14: yes
[02:21] <xreal> muken: gimme more details :D
[02:21] <pagios> you can do progressive download with mp4 only
[02:21] <pagios> unless you have a smart player
[02:21] Action: c_14 has never really looked into fragmentation, but does know that many clients don't support it
[02:23] <muken> mp4 can be streamable without fragmentation if optimized for progressive download, i.e. moov atom precedes mdat atom
[02:23] <xreal> frag_duration works, thanks!
[02:23] <muken> but cannot be __live__ streamable without fragmentation
[02:24] <xreal> let me play with -movflags +faststart
[02:25] <c_14> xreal: that won't help for live streaming/playing while it's still encoding
[02:25] <c_14> And I'm not sure what that option does when fragmentation is enabled.
[02:25] <xreal> c_14: let me try :)
[02:25] <pagios> why not HLS?
[02:25] <c_14> pagios: I think he just wants to start watching a video while he's encoding it.
[02:25] <c_14> And then wants the file to be around afterwards as well.
[02:26] <xreal> pagios: c_14 is right.
[02:26] <xreal> Perhaps I can use -f tee and pipe it to ffplay ?
[02:26] <pagios> you can encode to hls at the same time and watch the fragments
[02:27] <xreal> pagios: does HLS need a running webserver?
[02:27] <pagios> yes
[02:27] <xreal> pagios: that's overkill for my use.
[02:27] <pagios> or youcan do an avplay playlist.m3u8
[02:27] <pagios> if locally stored
[02:27] <c_14> pagios: libav does have an aac encoder/decoder
[02:27] <c_14> xreal: you could just use the tee pseudo muxer, yes
[02:28] <c_14> either with a pipe or a udp stream or something
[02:31] <xreal> HLS also works, but I don't like the small .ts files which get created :D
[02:48] <pagios> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x2461ac0] Invalid SampleDelta -18810 in STTS, at 22616 st:1
[02:48] <pagios> [mp4 @ 0x254dfc0] Non-monotonous DTS in output stream 0:1; previous: 61529079, current: 61505344; changing to 61529080. This may result in incorrect timestamps in the output file.
[02:49] <pagios> any idea about these errros
[02:49] <c_14> That looks like a warning, not an error.
[02:50] <pagios> ignorable?
[02:50] <c_14> If the encoding process doesn't die and the output is playable, ignore it.
[02:50] <muken> input timestamp is invalid. libavformat interpolates the timestamp
[02:51] <c_14> ^ if you wanted a more technical explanation
[02:52] <pagios> timestamp invalid can be due to slow uplinks right
[03:16] <xreal> msg NickServ identify fewih32ihfid
[03:16] <c_14> xreal: you might want to change your NickServ password
[03:17] <xreal> c_14: I already did, this was just a joke :)
[03:17] <xreal> When downmixing AC3 to stereo, the effects are always too loud. I hate this :(
[03:17] <c_14> AC3 is not a channel layout
[03:17] <xreal> I know...
[03:18] <c_14> Did you mean, downmixing 5.1 (or whatever) to stereo?
[03:18] <xreal> yeah, sure. AC3 5.1 sound from DVD
[03:19] <c_14> What command are you using?
[03:19] <xreal> c_14: 1 sec
[03:19] <xreal> -strict experimental -c:a aac -b:a 240k -ac 2
[03:21] <c_14> Find the channels that have the effects on them, then use the pan filter to downmix while making sure those channels are quieter
[03:21] <c_14> or use the compand filter to apply drc
[03:21] <xreal> c_14: that's not easy at all :(
[03:23] <c_14> I believe effects are normally on FL,FR and BL,BR so pan='stereo|FL<0.5*FL+FC+0.4*BL|FR<0.5*FR+FC+0.4*BR' might do it/help
[03:25] <xreal> c_14: okay, let me try this. DRC works like this? -drc_scale between >= 0 ?
[03:26] <c_14> I'd probably start around 0.5 and increase/decrease as needed
[03:27] <xreal> ok
[03:45] <chan1> Fjorgynn: I mean what is the ffmpeg command to see the video and audio codec type of a file? (without converting it)
[03:45] <c_14> chan1: if memory serves, his command was correct. you can also use ffprobe
[03:46] <xreal> Can I use "q:v" with x264? Is it an alias of qp ?
[04:01] <c_14> eeeeeeeeh, -qscale is ignored, -crf is recommended.
[04:01] <c_14> -q:v being an alias for -qscale
[04:53] <xreal> Can I use "q:v" with x264? Is it an alias of qp ?
[04:53] <xreal> and what about crf? :D
[04:57] <c_14> eeeeeeeeh, -qscale is ignored, -crf is recommended.
[04:57] <c_14> -q:v being an alias for -qscale
[04:57] <c_14> https://trac.ffmpeg.org/wiki/Encode/H.264
[05:02] <chan1> my Samsung TV supports avi file containing h.264 and ac3.
[05:02] <chan1> I converted my Sony Camcoder output mts file containing h.264 and ac3
[05:02] <chan1> into avi file containing h.264 and ac3
[05:02] <chan1> using command 'ffmpeg -i file1.MTS -c copy file1.avi (output codec is just copy)
[05:02] <chan1> and the conversion is fast as expected but my TV cannot play it.
[05:02] <chan1> I don't know why. Using ffprobe, I see some differences in the file codec format
[05:02] <chan1> Video: h264 (High) (HDMV / 0x564D4448), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
[05:02] <chan1> => Video: h264 (High) (H264 / 0x34363248), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 14783 kb/s, 59.94 fps, 59.94 tbr, 59.94 tbn, 59.94 tbc
[05:02] <chan1> Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s
[05:02] <chan1> => Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, 5.1(side), fltp, 448 kb/s
[05:02] <chan1> As you see, some values changed (in parentheses) and maybe that's the reason.
[05:02] <chan1> I'm not a video/audio/ffmpeg expert..
[05:02] <chan1> Can anybody suggest any additional option to preserve as many parameters as possible?
[05:02] <chan1> Thanks in advance and sorry for the long question.
[05:02] <xreal> c_14: okay, qscale for xvid/divx and crf for x264/x265 ?
[05:03] <c_14> xreal: yep
[05:06] <c_14> chan1: does your tv support the high profile?
[05:06] <c_14> might only support main (or if it's terrible baseline)
[05:06] <chan1> c_14: yes I'm sure.
[05:06] <c_14> Do you have a video that you know works?
[05:07] <c_14> Can you pastebin the ffprobe from that video?
[05:08] <chan1> sorry I'm at work and I have go to a meeting in a minuite. I'll get back to you later (hope you don't miss it :))
[05:10] <xreal> c_14: Is there an easy way to figure out, if source is interlaced?
[05:11] <c_14> use the idet filter
[05:13] <xreal> c_14: good idea, thanks
[05:39] <xreal> c_14: When opening my DVB-S stream in ffplay, the resolution seems to be smaller. The same input on ffmpeg looks great.
[05:40] <xreal> even when piping from ffmpeg to ffplay, it's looking ugly.
[05:44] <c_14> What about other players?
[06:49] <xreal> c_14: you mean piping ?
[06:50] <c_14> Try playing the stream/ffmpeg output with another player.
[06:50] <xreal> hmm, need to try how to do this :)
[06:57] <xreal> c_14: looks nice in VLC
[06:57] <c_14> Maybe it's a bug in ffplay, what version are you running?
[06:58] <xreal> c_14: ffplay version N-68826-g504267f
[06:58] <xreal> let me try mplayer
[07:02] <chan1> c_14: sorry to interrupt, but this is the ffprobe info of a file that can be played at my TV.
[07:02] <chan1> ffprobe version N-68826-g504267f Copyright (c) 2007-2014 the FFmpeg developers
[07:02] <chan1> built on Jan 2 2015 22:18:10 with gcc 4.9.2 (GCC)
[07:02] <chan1> configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --ena ble-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --e nable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme
[07:02] <chan1> --enable-lib gsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencor e-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enabl e-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-l ibtheora --enable-libtwolame
[07:02] <chan1> --enable-libvidstab --enable-libvo-aacenc --enable- libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-l ibwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --ena ble-lzma --enable-decklink --enable-zlib
[07:02] <chan1> libavutil 54. 16.100 / 54. 16.100
[07:02] <chan1> libavcodec 56. 19.100 / 56. 19.100
[07:02] <chan1> libavformat 56. 16.102 / 56. 16.102
[07:02] <chan1> libavdevice 56. 3.100 / 56. 3.100
[07:02] <chan1> libavfilter 5. 6.100 / 5. 6.100
[07:02] <chan1> libswscale 3. 1.101 / 3. 1.101
[07:02] <chan1> libswresample 1. 1.100 / 1. 1.100
[07:02] <chan1> libpostproc 53. 3.100 / 53. 3.100
[07:03] <xreal> c_14: mplayer looks gr8
[07:04] <c_14> chan1: first of all, please use a pastebin service for anything more than 2 lines, second that's only stderr. Where's stdout?
[07:06] <c_14> xreal: might be an ffplay bug then
[07:06] <xreal> c_14: okay, let me try different versions.
[07:21] <xreal> c_14: how do I pipe audio? rawvideo does video only :)
[07:22] <c_14> just pick a format that supports video + audio, mpegts or matroska or something
[07:23] <xreal> doesn't work for pipe, I think ?
[07:23] <c_14> It does.
[07:24] <xreal> c_14: I hate you :)
[07:24] <xreal> why do you know anything?
[07:25] <c_14> Practice makes perfect.
[07:26] <xreal> c_14: it looks like ffplay playbacks the content in half the resolution
[07:26] <xreal> and scales it up then
[07:27] <c_14> If you can get a sample, submit a bug report.
[07:29] <xreal> okay, I'll do it tomorrow
[07:38] <xreal> c_14: when using HLS, the small files automatically get created. Can I create them on demand when accessing the playlist?
[07:38] <xreal> c_14: or is the HLS function about generating those files?
[07:47] <chan1> c_14: here's the original and two converted files' ffprobe info. http://pastebin.com/yRYBV9pY
[07:47] <chan1> The TV can play only the .mkv file converted using -acode ac3_fixed and cannot play .avi file that was converted with -c copy. .MTS is the original.
[07:48] <chan1> c_14:hope you could guess the problem..
[09:06] <Elirips> Is it normal that every ffmpeg instance uses exactly 528m virtual size or should I worry about ffmpeg loosing memory?
[09:07] <Elirips> I'm calling ffmpeg like this'ffmpeg -i $URL -an -c:v libx264 -b:v 128K -x264opts keyint=$X264_KEYINT -r $FPS -f flv $NGINX 2> $LOGFILE &'
[10:14] <Cas> Hi, I try to send my Decklink input to v4l2loopback and my audiocard but the two slowly drift away from each other.
[10:14] <Cas> Any tips how I can help this?
[10:15] <Cas> my command is: `ffmpeg -f decklink -i 'Decklink SDI' -f v4l2 /dev/video0 -f alsa -ar 4800 -ac 2 -acodec pcm_s16le hw:0`
[10:16] <sfan5> I have no idea whether this will help, but try -async 1
[10:17] <Cas> sfan5: thanks for the suggestion, tried that by didn't help
[10:40] <rhagu> hi, when I install ffmpeg using this guide: http://ask.xmodulo.com/compile-ffmpeg-ubuntu-debian.html will there be support for h264 and vc-1 mkvs?
[10:44] <Mavrik> encoding or decoding?
[10:51] <zap0> i have some webm, i'd like to make a small(thumbnail) 3-frame GIF.. how?
[10:58] <rhagu> Mavrik I will use this command: ffmpeg -v warning -i "$X" -map 0:v -map 0:a -threads 3 -f null -
[11:04] <Mavrik> rhagu, that doesn't answer my question.
[11:04] <Mavrik> zap0, ffmpeg -i blah.webm -vframes 3 test.gif ?
[11:05] <zap0> thanks, shall try in a moment.
[11:06] <rhagu> Mavrik I think this commands mean decoding and encoding to /dev/null, not sure whether ffmpeg really encodes if it goes to null
[11:07] <Mavrik> You still didn't answer the question if you want to decode or encode those formats.
[11:07] <Mavrik> And that command line also doesn't tell anything abou it.
[11:07] <Mavrik> (it's also hideous, what are you even trying to do with that? O.o)
[11:08] <rhagu> I want to decode and encode both h264 and vc-1
[11:21] <Fjorgynn> Good morgnin
[11:21] <Fjorgynn> morning
[11:42] <feliwir> sorry that i need to ask this here, but does someone know if there's a program with that i can download ultraviolet videos on my pc?
[11:46] <Fjorgynn> :)
[12:41] <c_14> chan1: I'm sorry, I have no idea. Your TV is probably looking for something very specific, but I don't know what.
[12:42] <Mavrik> using AAC and Main video profile is probably a better idea
[13:02] <eyal-ff> How can I know in real-time what is my bitrate? in live streaming
[13:04] <Fjorgynn> Welcome
[13:11] <Mavrik> eyal-ff, em, it shows up in the output line.
[13:15] <eyal-ff> I know that, I was wondering if there is a way to know it from ffprobe or the output line is the only way?
[13:23] <c_14> You can ffprobe the stream and it will give you the average bitrate for the samples it collected
[13:38] <eyal-ff> Thanks, I'll try
[14:20] <Popara> Guys i have noticed huge differences in FFmpeg 2.5.x even in Simple Streams whereas in the FFmpeg 2.4.x everything was perfect
[14:21] <Popara> it is hard for me to explain what my current problems are with details but in a few words some streamings are being closed without a reason in 2.5.x, some others say some errors and some other just lags
[14:21] <Popara> in v2.4.x the exactly the same streams are working very well
[14:21] <Popara> and i tried that in the same time, it can't be wrong
[14:21] <Popara> what so you suggest me to do
[14:22] <Popara> i stream via HLS
[14:39] <ubitux> Popara: can you try to git bisect?
[14:46] <Popara> @ubitux i grabbed the latest static binary today
[14:46] <Popara> and same happened
[14:49] <Popara> I would really like to report the bugs, but in this case what i will say, that it just lags? Nobody will reply. But i'm sure that this happens
[14:49] <ubitux> that's why you can try to git bisect yourself
[14:49] <ubitux> to point out the commit that caused this
[14:51] <Popara> aaa git bisect now i get it
[14:57] <Popara> i have completely no idea how to use it :(
[15:19] <c_14> Popara: http://git-scm.com/book/en/v2/Git-Tools-Debugging-with-Git#Binary-Search http://git-scm.com/docs/git-bisect
[15:51] <Cas> How do I know if I use the XCB screengrabber? I did compile without --x11grab but its a little unclear what is grabbing my display now
[16:59] <shevy> is there a short-variant of ffmpeg -i that just tells me which audio codec is in use? e. g. whether I have AAC or .mp3 in a .mp4 file?
[17:01] <Ultima> im not sure with ffmpeg but I know mediainfo will tell you
[17:02] <Mavrik> shevy, ffprobe
[17:02] <Mavrik> also accepts formatting options so you can get output in JSON and other formats for machine consumption
[17:02] <shevy> ah
[17:04] <schiho> Is there a list what options are set when using -preset ultrafast -tune zerolatency ?
[17:05] <SirDarius> shevy: with mediainfo, use something like `mediainfo --Inform="Audio;%Format%" file.mp4` this is as short as you can get
[17:05] <Mavrik> schiho, you'll probably have to look at x264 source :/
[17:07] <shevy> hmm mediainfo is an external command?
[17:07] <Mavrik> it's a separate project not related to ffmpeg :)
[17:09] <schiho> Marvik: Oh i see, i am just searching for a way to support more "seekability" of videos, pushing more intraframes... i can set the gopsize to 1 but i think the option "keyint" is the right thing but i don't know how to type this to the ffmpeg command
[17:10] <Mavrik> uhh
[17:10] <Mavrik> you do know that if you set keyint to 1 you'll get video with only IDR frames?
[17:10] <Mavrik> meaning it's going to be huge / ugly?
[17:11] <iive> shevy: ffprobe ?
[17:11] <shevy> iive yeah, I think I'll stick to ffmpeg-related commands
[17:12] <schiho> Mavrik: yes you are right but marvik when i want to enable seeking i thought it is essential to have as many i frames as possible?
[17:12] <schiho> Mavrik: What happens, when i click on a P Frame in the timeline, it has to go back and recalculate till where the Frame was clicked?
[17:12] <Mavrik> yes.
[17:13] <schiho> Is this something players do fast?
[17:13] <Mavrik> But the whole POINT of video codecs is the P and B frames.
[17:13] <Mavrik> If not you can just send list of JPEGs to people.
[17:13] <Mavrik> You'll get in range of 100x worse compression.
[17:13] <Mavrik> I frames are 30-150x as big as P and B frames.
[17:13] <schiho> ah you say, sending jpegs will have better compression?
[17:13] <schiho> so is one I frame bigger than a compressed jpg image?
[17:14] <Mavrik> ...
[17:14] <Mavrik> No.
[17:14] <Mavrik> It's about the same.
[17:14] <schiho> yes, and that is, instead of sending jpg's i can send i frames, i am in a local network
[17:14] <schiho> so bandwith is not a big problem
[17:16] <schiho> so changing gop size made the file only 2 times bigger...thats okey
[17:16] <schiho> -g set to 1
[17:17] <Mavrik> well.
[17:17] <Mavrik> remember to also set min-keyint
[17:17] <Mavrik> or gop size will get overriden
[17:18] <schiho> frame I:2035 it seems to be not overriden
[17:23] <iive> keyframes are like jpeg pictures. they don't need any other info to be decoded, but they are big. very big.
[17:23] <iive> that's why you want as little as possible of them. for rips, it is usually 1 keyframe per 10 seconds (or 250 frames)
[17:23] <schiho> :) what does very big mean in terms of MB ? So a 1920 x 1080 image with jpg compression is about 240kb in average
[17:24] <schiho> for real-time streaming it would be bad to have such a big interval? or am i overseeing an advantage?
[17:24] <iive> [libx264 @ 0x346dc00] frame I:760 Avg QP:19.34 size: 47999
[17:24] <iive> [libx264 @ 0x346dc00] frame P:19845 Avg QP:22.26 size: 9884
[17:24] <iive> [libx264 @ 0x346dc00] frame B:43028 Avg QP:25.66 size: 1558
[17:25] <iive> that's SD encode.
[17:26] <iive> so, yes, if you want to stream, you need to have frequent keyframes, or people would have to wait for the next keyframe to get normal image.
[17:26] <iive> usually 1 per second.
[17:28] <schiho> Okey, there are to problems with streaming 1) Latency 2) Wait for next clean image the latter one isn't interesting for me
[17:28] <iive> h264 provides another trick... partial keyframes. aka you use intra macroblocks to refresh just a portion of the image. this way you distribute the keyframe over a number of P frames. You'd have to discard n frames before you can show image...
[17:35] <BtbN> How is it possible that the same input video(a concat list) with -c copy produces 44.1kHz audio when muxed to flv, but 4800kHz when muxed to mp4? The original .ts files all show 48kHz.
[17:36] <BtbN> The flv output video has weird audio issues, with it jumping and cracking a bit all the time, so i think the 44.1kHz rate is just wrong
[17:37] <JEEB> flv the container limits the rate to that, but it just means "look at the stream's rate instead" with AAC
[17:37] <JEEB> if you look at the flv container's specs
[17:37] <JEEB> and flash seems to do exactly that, too
[17:38] <BtbN> the youtube transcoder doesn't aparently.
[17:38] <shevy> quick question - I can convert from .aac to .mp3 just fine? quality loss is no problem; command I use: 'ffmpeg -i test.aac test.mp3', so far it seems to work
[17:38] <JEEB> yes, many implementations didn't read the spec too well
[17:38] <JEEB> so they instead just encode in 44.1kHz
[17:38] <BtbN> http://youtu.be/HMA-m3-k6h4?t=2h56m40s that's what it sounds like
[17:39] <BtbN> it's not off by much, but doesn't sound completely right
[17:39] <JEEB> anyways, ffmpeg should show it as the proper rate if it can access the stream itself
[17:40] <JEEB> if it doesn't and/or something else is blindly trusting the container ratio (which is wrong), then of course there can be fuck-ups
[17:40] <JEEB> but well, I have no idea what the fuck you've done and which of all the moving parts has fucked up :P
[17:40] <schiho> @live so what is best for low-latency-streaming i want to have as fast as possible the image which is at the receivers side
[17:40] <JEEB> I'm just saying that if you are only looking at the container audio data, then that will tell you 44.1kHz
[17:40] <JEEB> because that's what the container metadata will say with AAC in flv
[17:41] <BtbN> JEEB, recording a twitch livestream. So i have a massive ammount of 4 seconds .ts files, and combine them with the -f concat input.
[17:41] <BtbN> For some reason that needs "-bsf:a aac_adtstoasc"
[17:41] <BtbN> But works fine otherwise
[17:41] <JEEB> (because the spec says "write this value, the demuxer/decoder combo will then ignore it and use the actual AAC stream"
[17:41] <BtbN> ffmpeg -f concat -i "${inp}" -c copy -bsf:a aac_adtstoasc -movflags +faststart -y "${outp}"
[17:41] <BtbN> is the commandline i use
[17:41] <BtbN> (without the movflags for flv)
[17:42] <JEEB> is the bitstream filter required or does it even work correctly for FLV output?
[17:42] <BtbN> it's required, ffmpeg aborts without it
[17:42] <JEEB> ok
[17:42] <BtbN> and tells me to use it
[17:42] <JEEB> if you decode the output with mpv or something else sane does the result work?
[17:43] <BtbN> i'll have to download one part for that
[17:43] <BtbN> https://bpaste.net/show/1c883ed97490
[17:43] <JEEB> well you created the files
[17:43] <BtbN> on my sever
[17:43] <JEEB> you should have them on hand then :P
[17:43] <BtbN> *server
[17:43] <JEEB> well, then grab one of them and check
[17:44] <BtbN> yeah, i'm on it right now. Just takes a moment, as they are 10GB each
[17:44] <JEEB> with something relatively sane like mpv, and if you are willing enough, with flash player
[17:44] <JEEB> because the latter is the "reference" for FLV
[17:44] <BtbN> so ffprobe reporting it as 44.1 is ok, or a bug?
[17:44] <JEEB> which part of ffprobe
[17:44] <BtbN> see my paste
[17:44] <JEEB> container only or does it actually decode
[17:45] <JEEB> because if it touches the stream then it should get the actual rate
[17:45] <JEEB> if container only, then 44.1kHz should be gotten (and ignored by the player/whatever)
[17:45] <JEEB> I have no idea how much ffprobe does in that case
[17:46] <JEEB> I will guess that it doesn't actually decode, or there is a bug
[17:46] <JEEB> also have you tried with a newer ffmpeg?
[17:46] <JEEB> 2.2 series is pretty old
[17:46] <BtbN> Not yet, 2.2.11 is already the "unstable" version on gentoo
[17:46] <BtbN> stable is 1.2.6
[17:47] <JEEB> yes, because they haven't updated everything under gentoo to be compatible with the newer APIs
[17:47] <shevy> huh
[17:47] <JEEB> that said, pretty much everything a normal user uses is updated
[17:47] <shevy> I have 2.5.1 here
[17:47] <JEEB> so updating to a newer ffmpeg shoud be relatively problemless
[17:47] <BtbN> 2.5 is available, but hardmasked
[17:47] <shevy> I thought gentoo was source based
[17:47] <JEEB> yes, they're all hardmasked
[17:47] <JEEB> shevy, basically they're not updating their "stable" shit until _everything_ in their repos gets updated
[17:48] <JEEB> there's some really minor things or whatever there, which are taking really long
[17:48] <JEEB> so even debian has now been quicker at updating libav/ffmpeg than gentoo :P
[17:48] <shevy> yeah. arch too https://www.archlinux.org/packages/extra/i686/ffmpeg/
[17:48] <BtbN> debian even has better support, as it can install libav and ffmpeg ad the same time
[17:48] <BtbN> on gentoo it's mutualy exclusive
[17:48] <shevy> I guess what someone else said is true; like when the gentoo wiki died
[17:49] <BtbN> the gentoo wiki died?
[17:49] <shevy> yeah
[17:49] <JEEB> the last time I poked the libav person who was maintaining libav/ffmpeg in gentoo
[17:49] <BtbN> looks very alive to me
[17:49] <JEEB> and asked what the fuck is taking time with he just yelled at me and ragequit
[17:49] <JEEB> because as always the libav/ffmpeg maintainers get all the shit
[17:49] <JEEB> even though it's the projects that use those projects' libraries that are not being updated
[17:49] <shevy> BtbN http://www.gentoo-wiki.info/ "Gentoo-Wiki.com has been lost and it will be rebuilt from scratch."
[17:50] <BtbN> shevy, that's an inofficial wiki, and it's been down forever
[17:50] <BtbN> http://wiki.gentoo.org/ is the Gentoo wiki
[17:50] <JEEB> anyways, you can always build a simple ffmpeg build in your homedir :P
[17:50] <JEEB> for testing
[17:50] <JEEB> and trying ffprobe/remuxing with it
[17:50] <BtbN> I'm not aware of any software on that machine which links against ffmpeg
[17:50] <BtbN> i just use the cli tools
[17:50] <BtbN> so i'll just bump it
[17:50] <JEEB> k
[18:02] <shevy> Unknown encoder 'libx264' - question. seems I forgot to compile that support in, will recompile later. Is there an alternative to a .mp4 codec? I am not sure how to find out which codecs will work and which not with my current ffmpeg
[18:03] <c_14> mpeg4 should work
[18:03] <sfan5> shevy: ffmpeg -codecs tells you which codecs are compiled in; mpeg4 is an alternative for .mp4 (it has a lower quality though)
[18:04] <shevy> aha
[18:05] <shevy> well the quality is fine, I only wanted to add an image to a mp3 file and call that a "video"
[18:08] <sfan5> if you want to add artwork to an mp3 file use mjpeg
[18:08] <sfan5> IIRC it only works with mjpeg
[18:20] <kepstin-laptop> on the other hand if you want to make a "video" file that just shows a static image to e.g. upload to youtube, you're on the right track.
[18:22] <shevy> hmm
[18:22] <shevy> the .mp4 is much larger than the .mp3
[18:23] <shevy> can that be? if you add like only one small .png to a .mp3 file. I have the feeling I must have done something wrong... from 4 MB to 12 MB
[18:24] <BtbN> did you re-encode the audio?
[18:26] <shevy> hmmm
[18:26] <BtbN> Paste your full command (and output) please
[18:58] <kruug> Hello, I'm converting all of my video media files to MP4, and I see a lot of them have the YUV420p codec (is it a codec?) being used. My question is this: does this mean that it's not HD? As in, 420p vs 720p vs 1080p. Using WinFF, I'd like to keep the files HD if they originally were HD, but how can I tell?
[18:59] <ezekiel> kruug: that's a color space designation
[19:00] <ezekiel> https://en.wikipedia.org/wiki/YUV
[19:00] <ezekiel> it's not necessarily linked to "HD" which is a quality designation
[19:00] <kruug> How can I use WinFF without losing definition then? I don't want to have to manually find the "definition" for each individual file
[19:17] <nasojlsu> ffmpeg -i filename
[19:18] <nasojlsu> will give you specs
[19:18] <nasojlsu> or you could use mediainfo
[20:25] <swillits> I have a movie file with one video and one audio track in it. What I need to see is the start times of the individual tracks. I suspect that the audio starts before the video and need to confirm it. I'm new to ffmpeg surely somewhere in its 8 million capabilities it can show me this.
[20:39] <ezekiel> swillits: my suggestion may not be the simplest answer, but I have done what you're talking about
[20:39] <ezekiel> I used filters to show the "Presentation Time Stamps" (PTS)
[20:39] <swillits> I just figured out that ffprobe -show_streams did the trick. I looked at the start_pts/time of each stream
[20:40] <swillits> I had no idea streams could have negative start times.
[20:40] <ezekiel> nice
[23:43] <jookiyaya> does ffmpeg support h263 encoding?
[23:44] <JEEB> I think yes
[23:44] <jookiyaya> which one
[23:45] <JEEB> ?
[23:45] <JEEB> I think there's just one ITU-T H.263 :P
[23:45] <jookiyaya> which h263 encoder
[23:45] <JEEB> I would guess lavc has its own
[23:45] <JEEB> libavcodec/h263enc or something along those lines?
[23:45] <jookiyaya> i see
[23:46] <JEEB> hmm or actually
[23:46] <JEEB> I see a h263dec
[23:47] <JEEB> but not h263enc
[23:47] <JEEB> anyways, it would be much simpler for you to just run `ffmpeg -codecs:v |grep "h263"`
[23:48] <JEEB> ah yes
[23:48] <JEEB> it's in mpegvideo_enc
[23:48] <JEEB> of course
[23:48] <JEEB> the horribly big monster :P
[00:00] --- Wed Jan 7 2015
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