[Ffmpeg-devel-irc] ffmpeg.log.20160801

Tue Aug 2 03:05:01 EEST 2016

[02:39:21 CEST] <wallbroken> 32bit ffmpeg vs 64bit ffmpeg, i encoded the same video but i have different output, why?
[02:58:37 CEST] <gro> Hallo @all
[03:00:50 CEST] <gro> could somebody help me with some coding problems?
[03:07:31 CEST] <gro> does someone know, how i can mute audio programmatically? i tried to set the channels with av_dict_set(&opts, "channels", 0, 0) but it has no effect
[03:20:52 CEST] <DHE> as in toggle mute while running, or just generate silence for use in an audio track?
[03:26:32 CEST] <gro> as an option to stream the video without audio
[03:28:05 CEST] <gro> i'm streaming my gopro to youtube and would like have the option to mute the audio
[03:49:48 CEST] <gro> @DHE some ideas?
[04:01:24 CEST] <DHE> why not just discard it? most formats will accept a video file containing no audio stream at all
[04:03:14 CEST] <gro> ok so only use the video stream
[04:04:18 CEST] <gro> thanks, i will try this later.
[04:05:53 CEST] <gro> is there a list with all av_dict options i can use? i tried to debug ffmpeg with some options like -an but i cant figure out how to find them
[04:14:03 CEST] <DHE> they're not all ffmpeg commandline options. you'll find a ton of accepted parameters under libavcodec/options_table.h and similar for libavformat. the ffmpeg-specific ones are mostly under ffmpeg_opt.c
[04:14:16 CEST] <DHE> and that's not even counting the codec-specific and format-specific options
[04:15:12 CEST] <DHE> it's why the docs say the AVDictionary you pass in gets stripped of all passed options and anything not processed is left behind. helps ffmpeg take one set of parameters and just deal with them
[04:17:06 CEST] <gro> sounds reasonable :)
[04:21:25 CEST] <gro> Thank you for your help!
[04:42:29 CEST] <ferdna> anyone doing hls streaming?
[04:42:43 CEST] <ferdna> i get this message in ffserver:
[04:42:51 CEST] <ferdna> Sun Jul 31 20:39:32 2016 [hls @ 0x1173210]failed to rename file .tmp to
[05:00:32 CEST] <DHE> let me guess, you're uploading via webdav
[05:13:17 CEST] <ferdna> DHE, is that to me?
[05:13:32 CEST] <DHE> yes
[05:13:43 CEST] <ferdna> no i am not...
[05:13:50 CEST] <ferdna> i am running ffserver
[05:15:35 CEST] <DHE> wait, just ".tmp" as a filename? no other name? sounds like it failed to rename to the empty string...
[05:15:52 CEST] <DHE> and I'd bet ffserver doesn't handle HLS very well. you're better off using ffmpeg and apache instead
[05:16:12 CEST] <ferdna> DHE, same problem as here:
[05:16:12 CEST] <ferdna> http://ffmpeg.gusari.org/viewtopic.php?f=12&t=2611
[06:48:56 CEST] <ferdna> DHE
[06:48:58 CEST] <ferdna> that was it
[06:49:02 CEST] <ferdna> thank you very much
[09:11:27 CEST] <Fyr> guys, concatenation of video and audio separately gives desync. can FFMPEG convert into LPCM?
[09:38:09 CEST] <jiggunjer> Fyr: input files same codecs?
[09:38:16 CEST] <Fyr> sure
[09:42:21 CEST] <Fyr> what container FFMPEG can make use of to mux ALAC?
[09:45:19 CEST] <relaxed> Fyr: m4a
[09:45:31 CEST] <Fyr> m4a == mp4
[09:45:59 CEST] <relaxed> yes
[09:46:40 CEST] <relaxed> matroska too
[09:46:41 CEST] <Fyr> why on Earth FFMPEG refuses to mux ALAC into mp4 and does that when muxing into M4A?
[09:47:13 CEST] <relaxed> well, m4a is similar to mp4
[09:47:23 CEST] <relaxed> it's apple's audio container
[09:49:59 CEST] <relaxed> Fyr: it will work with ... -f ipod output.mp4
[09:50:08 CEST] <Fyr> oh, thanks
[09:50:13 CEST] <Fyr> I hate FFMPEG.
[09:52:12 CEST] <JEEB> m4a most probably has different identifiers in the container, and likely "-f ipod" is just going to do that instead of finding it out by the extension
[09:52:15 CEST] <JEEB> :P
[10:24:21 CEST] <Fyr> does FFMPEG support variable framerate with MP4?
[11:16:56 CEST] <carado> hi ! mpv tv:// can read from my webcam fine (although it prints some errors, http://paste.awesom.eu/84lp ) but ffmpeg prints Invalid data found when processing input and quits immediately ( http://paste.awesom.eu/W4Ck ). what can I do ?
[13:39:07 CEST] <Fyr> JEEB, do you happen to know why concatenation of four files gives crap in the joints and desync while concatenating one-by-one gives normal video/audio?
[13:41:08 CEST] <JEEB> nfi
[13:41:16 CEST] <JEEB> esp. since you're using the stuff that works without re-encoding
[13:41:29 CEST] <JEEB> never used that stuff and I just know none of them are perfect
[13:48:25 CEST] <senyai> Hi! Could you guys check what's wrong with this file? https://arseniy.net/demo.avi - I'm using latest ffmpeg and avcodec_send_frame/avcodec_receive_packet interface, and I can't play it with vlc.
[13:49:50 CEST] <Fyr> so, I'm discovering the completely new territory where no developer ever set foot on it.
[13:50:30 CEST] <Fyr> senyai, it cannot be played with PotPlayer, Light Alloy. ffprobe gives almost nothing.
[13:54:38 CEST] <JEEB> Fyr: no - some developer made the things for some specific use case and now people are of course thinking it's a generic solution which it isn't ;)
[13:54:53 CEST] <Fyr> ='(
[13:55:20 CEST] <JEEB> because "more features is better even if they're not perfect"
[13:55:32 CEST] <JEEB> (well, things are never perfect but you know what I mean)
[13:55:58 CEST] <Fyr> a tool must do one thing and do that perfectly.
[15:03:03 CEST] <shincodex> why
[15:03:07 CEST] <shincodex> is there two timeouts
[15:03:10 CEST] <shincodex> rtsp stimeout timeout
[15:03:17 CEST] <shincodex> http timeout
[15:03:29 CEST] <shincodex> pass in timeout to rtsp avinput dies and says option doesnt exist
[15:04:00 CEST] <shincodex> timeout should imply a generic word for timeout on anything
[15:04:06 CEST] <shincodex> socket, curl call or wtf ever
[15:04:27 CEST] <shincodex> microseconds is a bit extreme
[15:04:35 CEST] <spooooon> rtsp is very particular and silly, it could be the socket timeout, and the GET/SET_PARAMETER timeout
[15:04:49 CEST] <shincodex> everything is silly
[15:04:53 CEST] Action: shincodex suicides
[15:05:05 CEST] <spooooon> I'm not 100% sure as I don't know the code, but I do know rtsp very particular about things it shouldn't be
[15:05:05 CEST] <shincodex> best part of waking up is ffmpeg code in your cup
[15:05:41 CEST] <shincodex> im thankful at least
[15:05:47 CEST] <shincodex> I have a visual studio debugger attached to it
[15:05:57 CEST] <Fyr> xD
[15:05:57 CEST] <spooooon> I would guess stimeout is the socket timeout, and timeout is the GET_PARAMETER timeout(which rtsp rfc says SET_PARAMETER should be preferred btw!)
[15:06:09 CEST] <shincodex> cause i would plug pull my network and connect to a rtsp camera and it be like infinite loop
[15:06:19 CEST] <shincodex> and im like i bet some dude set a infinite timeout ina poll somewhere
[15:06:21 CEST] <shincodex> YEP! THEY DID
[15:06:43 CEST] <spooooon> oh god you haven't read the rfcs as well
[15:07:18 CEST] <shincodex> someday xrdp needs to support copy paste from windows to linux
[15:07:30 CEST] <shincodex> so i can paste this protocol link
[15:07:34 CEST] <shincodex> of timeout and stimeout
[15:59:34 CEST] <hero_biz> hi guys
[15:59:39 CEST] <Fyr> hi
[16:00:00 CEST] <Fyr> guys, how do native speaker call people, whose hands grow from their asses?
[16:00:06 CEST] <hero_biz> guys, I have downmixed a 5.1 audio to streo with this command:
[16:00:22 CEST] <hero_biz> ffmpeg -i audio.mkv -vn -ac 2 -c:a libfdk_aac -b:a 48k audio2.aac
[16:00:53 CEST] <hero_biz> but when I was checking I noticed 5.1 audio is 25min and 11 sec while stero audio is only 25 min.
[16:00:58 CEST] <hero_biz> any idea why this happens?
[16:01:53 CEST] <Fyr> hero_biz, if i were you, I would convert them into wav and compared in Audacity.
[16:03:13 CEST] <hero_biz> when length of 2 audio streams differ, they practically should differ,isn't it?
[16:03:18 CEST] <hero_biz> O.O
[16:25:32 CEST] <shincodex> i plug pull and pass in timeout and ffmpeg gives error -138 and av_strerror doesnt return junk
[16:25:38 CEST] <shincodex> ETIMEDOUT doesnt = -138 either
[19:29:16 CEST] <shincodex> is there a av_ignore_options_youdontgiveashitabout_instead_of_failing_avopeninput function?
[19:30:15 CEST] <JEEB> that's bad design, so no
[19:30:37 CEST] <JEEB> you set something so clearly you wanted something
[19:30:59 CEST] <JEEB> it's like autoconf based configure scripts ignoring unknown parameters
[19:31:02 CEST] <JEEB> which is !?!
[19:31:16 CEST] <shincodex> AV_GUESS_FORMAT
[19:31:20 CEST] <shincodex> !!!!!!111
[19:31:42 CEST] <shincodex> holycrap thats rare googling and its like boom there it is!
[19:32:06 CEST] <shincodex> actually i had to revolve my eyeballs around the example code the google link posted
[19:32:10 CEST] <JEEB> I did avio stuff before and when calling libavformat I didn't have to specify that
[19:32:19 CEST] <shincodex> Well heres the trick
[19:32:20 CEST] <JEEB> it would probe what the thing could be by default
[19:32:39 CEST] <shincodex> avformat_open_input
[19:32:59 CEST] <shincodex> if i use timeout when its RTSP it goes fail no such dictionary option
[19:33:09 CEST] <shincodex> and im like that seems wrong cause docs say there is
[19:33:12 CEST] <shincodex> whatev... ditch it
[19:33:16 CEST] <shincodex> stimeout works...
[19:33:40 CEST] <shincodex> http is like ? whats stimeout? whatever ill load anyways
[19:33:48 CEST] <shincodex> avformat_open_input success!
[19:35:58 CEST] <shincodex> not that one
[19:36:02 CEST] <shincodex> guess_stream_format
[19:36:04 CEST] <shincodex> seems to be it
[19:49:59 CEST] <shincodex> oh no... its like filestreams only
[19:50:02 CEST] <shincodex> oh no
[20:11:36 CEST] <shincodex> its rtsp
[20:11:39 CEST] <shincodex> it does suck
[20:11:46 CEST] <shincodex> so does this options thing.
[20:11:52 CEST] <shincodex> if timeout > 0 oh start listening
[20:12:31 CEST] <shincodex> else timeout -1 oh nevermind use stimeout for socket poll timeout where as regular timeout on http is for socket timeout
[20:32:17 CEST] <wallbroken> 32bit ffmpeg vs 64bit ffmpeg, i encoded the same video but i have different output, why?
[20:33:25 CEST] <Fyr> underpants gnomes?
[20:46:49 CEST] <jkqxz> wallbroken:  What codecs are you using?
[21:04:01 CEST] <wallbroken> jkqxz
[21:04:19 CEST] <wallbroken> ffmpeg.exe -i "Audio.mp3" -i "video.avi" -map 0:a -map 1:v -af atempo=2395.767375/2498.120625 -c:v copy 1.avi
[21:07:16 CEST] <jkqxz> So video is copied but you haven't specified anythng for audio so it will use some random default (aac, maybe?).  At least specify the audio codec you want on the output.
[21:09:07 CEST] <wallbroken> jkqxz, the audio output  by default is mp3 abr
[21:09:11 CEST] <jkqxz> After that, I would guess floating point in the audio stuff.  Is that 32-bit vs. 64-bit x86?  If so, the 32-bit is probably using the x87 FPU while the 64-bit will SSE.  They don't have the same precision and won't get the same answers.
[21:09:49 CEST] <wallbroken> ok, and is better to do it with 32 or 64 bit?
[21:10:44 CEST] <jkqxz> The answers will be different, but it's unlikely that one will be in any sense better than other.
[21:12:08 CEST] <jkqxz> In general you are better off using 64-bit, but that is because it tends to be faster rather than anything to do with the quality.
[21:33:51 CEST] <wallbroken> jkqxz, so, quality will be identical?
[21:48:09 CEST] <jkqxz> wallbroken:  Yes.
[22:10:41 CEST] <wallbroken> jkqxz, but you said "64 bit get more precision"
[22:14:12 CEST] <jkqxz> No, 32-bit has more precision.  It uses the legacy x87 FPU which keeps things in 80-bit internally, while SSE gives you the normal 64-bit precision.
[22:15:25 CEST] <jkqxz> Anyway, the precision in either case is generally much more than is relevant to any calculation done on audio.  It just affects the low bits and makes the results differ irrelevantly.
[22:21:32 CEST] <thebombzen_> not really an FFmpeg question, but do you guys see any real downside nowadays to compiling libx264 as 10-bit?
[22:22:23 CEST] <thebombzen_> I've been compiling it as 10-bit so it'll default to High10, and so far no decoder has complained to me, but I also haven't tried to convert a video for a dumbphone in a long time.
[22:22:42 CEST] <thebombzen_> Is there any practical reason for NOT doing 10-bit H.264?
[22:26:42 CEST] <ritsuka> 10bit is not supported by a lot of hardware decoders
[22:26:51 CEST] <jkqxz> Only that it won't work with most devices and proprietary decoders (including hardware).  If you're happy to always decode in software with lavc then it's awesome.  Otherwise, not so much.
[00:00:00 CEST] --- Tue Aug  2 2016