[Ffmpeg-devel-irc] ffmpeg.log.20120628

[burek]

[00:48] <cbreak> http://paste.the-color-black.net/277890 :(
[00:48] <cbreak> this is on OS X
[00:48] <cbreak> so no ELF
[00:51] <lolfrenz> hey
[00:51] <lolfrenz> I have a .flv file that I'd like to convert to mp3 with libmp3lame. I'm doing ffmpeg -i file.flv -qscale 8 -ainput libmp3lame -ab 192k file.mp3
[00:51] <lolfrenz> but while the flv sounds nice, the mp3 doesn't (especially bass on higher volume)
[00:53] <lolfrenz> in fact, even -acodec copy which creates an aac makes horrible sound
[00:55] <llogan> use a pastebin site to show your ffmpeg command and the complete console output
[00:57] <ubitux> cbreak: i think that's now fixed
[00:58] <ubitux> or maybe it will be in the next merge
[00:58] <lolfrenz> one moment
[01:04] <lolfrenz> llogan, http://pastie.org/4162658
[01:04] <lolfrenz> the .flv sounds good but the .mp3 sounds horrible
[01:06] <llogan> lolfrenz: replace "-sameq -ab 192k" with "-aq 4"
[01:07] <lolfrenz> one min
[01:08] <lolfrenz> btw, where is this documented? ffmpeg says "codec-specific" for -aq
[01:08] <lolfrenz> so I don't even know what 4 means
[01:09] <llogan> lolfrenz: yeah, the value varies depending on the encoder. in this case, consider it the same as "lame -V".
[01:10] <lolfrenz> so why 4 and not 0?
[01:10] <llogan> http://wiki.hydrogenaudio.org/index.php?title=LAME#VBR_.28variable_bitrate.29_settings
[01:10] <lolfrenz> oh, nice
[01:10] <llogan> also scroll up and read the stuff under "Recommended encoder settings"
[01:11] <llogan> 4 is default
[01:11] <llogan> for lame, IIRC
[01:11] <llogan> 4 should be close to perceptual transparency
[01:13] <llogan> so to directly answer your question, most people probably can't tell a difference between 0 and 4.
[01:13] <lolfrenz> yeah, I've read that on the wiki
[01:13] <lolfrenz> just tried with 4, still sounds horrible
[01:14] <lolfrenz> it's an youtube-downloaded .flv, downloaded using youtube-dl
[01:14] <lolfrenz> you can probably reproduce with ./youtube-dl http://www.youtube.com/watch?v=9w1PYLJEpxk&feature=related and then ffmpeg -i file.flv -acodec libmp3lame -aq 4 file.mp3
[01:15] <lolfrenz> hmm, maybe it's some volume stuff
[01:15] <lolfrenz> if I lower the volume a bit it sounds pretty much like the video
[01:15] <lolfrenz> brb, sorry
[01:19] <llogan> maybe you need this: http://www.poormojo.org/pmjadaily/archives/009864.php
[01:20] <cbreak> ubitux: yep, that problem got fixed :)
[01:20] <cbreak> but a next one appeared: http://paste.the-color-black.net/277937
[01:32] <cbreak> http://paste.the-color-black.net/277940
[01:35] <Jan-> has anyone ever p/invoked avformat/avcodec/etc from inside a C# application?
[01:35] <Jan-> I guess it's *possible*
[01:35] <Jan-> but has it ever been done
[01:46] <Nickname123> what is the recommended aac encoder for ffmpeg?
[01:50] <llogan> Nickname123: libfaac, libvo_aacenc, or aac (descending order of "quality" / bitrate). or libaacplus for lower bitrates.
[01:51] <llogan> Jan-: did you just read Phil's post on ffmpeg-user mailing list?
[01:51] <Nickname123> thanks llogan.  i thought I read libfaac wasn't recommended
[01:51] <llogan> why not?
[01:52] <llogan> lolfrenz: when you come back try the command with -vn. perhaps the png "video" in the mp3 is screwing up stuff in your player.
[01:52] <Nickname123> i haven't looked at using it in a year or so I don't really remember what the reasons were. just wanted to double check what was recommended now
[01:54] <llogan> lolfrenz: also the input and output sound the same to me (with and without -vn), but i'm in a noisy room.
[01:55] <llogan> you could also pipe from ffmpeg to an external encoder such as faac or neroaacenc
[01:56] <Nickname123> I am going to give libfaac a try.  My hearing isn't great so I defer to others opinions about sound quality
[01:58] <Nickname123> What would be the bitrate cutoff for "lower bitrates" that you would reocmmend libaacplus?
[02:05] <llogan> Nickname123: i'm not sure. <128k?
[02:05] <Nickname123> Okay thanks
[02:06] <llogan> with libfaac use -aq 100 if you just care about quality
[02:07] <llogan> (see "faac -q")
[02:08] <Nickname123> nice
[03:58] <PurpleSkinnyJean> hi. if i'm cropping, deinterlacing and scaling, does it matter which order i put this in "crop=in_w-4:in_h-6:4:3,yadif=1:-1,scale=1280:720"? also, does it matter where i put this whole thing in the syntax?
[04:01] <PurpleSkinnyJean> anyone here use ffmpeg?
[04:05] <PurpleSkinnyJean> is anyone even here?
[04:05] <brocatz> what
[04:05] <brocatz> are you new to irc
[04:05] <PurpleSkinnyJean> yes. this is my first time
[04:06] <PurpleSkinnyJean> popping my cherry
[04:06] <brocatz> okay, basically just wait around
[04:06] <brocatz> people are all in different timezones
[04:06] <brocatz> ask your question again in a couple of hours
[04:06] <PurpleSkinnyJean> how about you?
[04:06] <brocatz> if you see poeple chatting, send them a funny cat picture
[04:06] <brocatz> then ask your question
[04:06] <PurpleSkinnyJean> where do i get those?
[04:06] <brocatz> people don't usually answer unless they know
[04:07] <brocatz> i don't know
[04:07] <PurpleSkinnyJean> can you answer my ?
[04:07] <PurpleSkinnyJean> i will send you a nude pic of me if you do
[04:08] <brocatz> that sounds risky
[04:08] <PurpleSkinnyJean> not at all
[04:10] <PurpleSkinnyJean> still there bro bro?
[04:10] <brocatz> yep
[04:10] <brocatz> i've helped as much as i can
[04:10] <PurpleSkinnyJean> is everyone in here new to ffmpeg like us?
[04:12] <PurpleSkinnyJean> burek: you! are you online?
[04:29] <PurpleSkinnyJean> hi. if i'm cropping, deinterlacing and scaling, does it matter which order i put this in "crop=in_w-4:in_h-6:4:3,yadif=1:-1,scale=1280:720"? also, does it matter where i put this whole thing in the syntax?
[04:59] <garme> hi, folks.
[04:59] <garme> I'm trying to convert some wmv files to avi.
[04:59] <garme> I'm using ubuntu 12.04 64 bits.
[05:00] <garme> I did: ffmpeg -i input.wmv output.avi... but i got the following error: 'error opening filters'
[05:00] <garme> What's happening, guys?
[05:05] <diegoviola> garme: what ffmpeg version?
[05:10] <garme> diegoviola, 0.8.3
[05:20] <PurpleSkinnyJean> hi. if i'm cropping, deinterlacing and scaling, does it matter which order i put this in "crop=in_w-4:in_h-6:4:3,yadif=1:-1,scale=1280:720"? also, does it matter where i put this whole thing in the syntax?
[06:01] <relaxed> PurpleSkinnyJean: deinterlace, crop, then scale
[06:36] <rm-rf> burek: my command is as follows: ffmpeg -i rtsp://192.168.145.200:554/video.mp4 -vcodec copy -ar 44100 -ab 128k -t 300 /tmp/ffmpeg.mp4
[06:36] <rm-rf> burek: the error i get is: [mp4 @ 0x87fb800] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 245 >= 245
[06:37] <rm-rf> (not trying to cross-post, just posting in the relevant channel after accidentally posting in #videolan)
[07:16] <Viking667> hey ho. Can ffmpeg keep up with recording two video streams in addition to two audio streams?
[07:16] <Viking667> I have it working well enough to one video stream on a hard disk and another two audio streams to a separate disk.
[07:16] <Viking667> ... I was interested in including a webcamera feed in addition.
[08:58] <iUnix> Ffmpeg how can i test the H.265?
[08:58] <iUnix> currently HEVC draft 7
[09:44] <hi117> iUnix: code it yourself? i would be surprised if theres an encoder for it yet
[09:55] <burek> rm-rf, can you please use pastebin.com, to show your command line and its output?
[09:57] <JEEB> hi117, the lolreference lolimplementation?
[09:57] <JEEB> http://hevc.kw.bbc.co.uk/git/w/jctvc-hm.git
[10:01] <iUnix> JEEB, is it the working module according to the Draft 7?
[10:07] <gnarface> hey, does ffmpeg support VOB files with multiple camera angles yet?
[10:20] <iUnix> http://www.vcodex.com/h265.html  - where is those Feb 2012 committee draft?
[10:20] <iUnix> July 2012 - Draft international standard
[10:20] <iUnix> upcoming
[11:31] <astroler> How to decode the smooth streaming without silverlight, have any good suggestions.
[11:50] <burek> what is the "smooth streaming"
[11:53] <astroler> burek: Smooth Streaming, an IIS Media Services extension, enables adaptive streaming of media to Silverlight and other clients over HTTP.
[11:53] <burek> what is so special about it?
[11:57] <astroler> seamless switching bit rate.
[11:58] <burek> ok
[12:59] <voltagex> hey, how do I verify/report a GPL violation?
[13:01] <voltagex> http://www.4videosoft.com windows version contains ffmpeg dlls and no mention at all
[16:07] <THEBLACKKING> how can i compile ffmpeg with sdl support
[16:07] <THEBLACKKING> ?
[16:07] <THEBLACKKING> i compiled sdl
[16:07] <cbsrobot> then --enable-libsdl (if im not wrong)
[16:08] <THEBLACKKING> and configured configured ffmpeg with:
[16:08] <THEBLACKKING> ./configure --target-os=mingw32 --cross-prefix=i586-mingw32msvc- --arch=x86 --pkg-config=pkg-config
[16:08] <THEBLACKKING> sorry, *cross-compile
[16:09] <THEBLACKKING> Unknown option "--enable-libsdl".
[16:09] <THEBLACKKING> See ./configure --help for available options.
[16:09] <THEBLACKKING> i add --enable-outdev=sdl too
[16:10] <THEBLACKKING> but the configure output for sdl support still "no"
[16:13] <THEBLACKKING> what should i do ??
[16:32] <zambo> is it possibl to append audio to the audio track of a video using ffmpeg?
[16:42] <cbsrobot> zambo: join audio and video together ?
[16:43] <cbsrobot> see http://ffmpeg.org/ffmpeg.html#Advanced-options
[16:45] <THEBLACKKING> <THEBLACKKING>	what should i do ??
[17:03] <shahriman0>  if sws_scale sees that input and output parameters are identical will it just pass through the data or does the user have to check that himself?
[17:10] <zambo> re: the map command: it looks like this is only for mapping multiple streams in one files, no?  What I want to do is append multiple audio files to one audio stream in a video file
[17:12] <shahriman0> zambo: multiple audio file to one audio stream? you mean mix the multiple input to one stream and mux it in a video?
[17:13] <virtus_> hello, I was wondering if someone could help me out? I'm trying to using the lutrgb filter to split an input FLV into two output MP4s, one for RGB channels, one for ALPHA.. any ideas on best settings?
[17:17] <zambo> shahriman0: Yes exactly
[17:17] <shahriman0> there's one amix filter afaik
[17:18] <zambo> I'm also trying to concatenate the m4a files separately first to make one audio track, but not having much success with that.
[17:19] <shahriman0> I am not sure if ffmpeg can concatanate
[17:19] <shahriman0> oh it can
[17:20] <shahriman0> wait there is a concat protocol
[17:20] <shahriman0> zambo: try to use that, I have used it with video before
[17:20] <shahriman0> should work for audio too
[17:21] <zambo> I highly doubt it will work on Android
[17:21] <zambo> but I will try, thank you
[17:21] <shahriman0> you doubt without trying?
[17:21] <shahriman0> any reason for your doubt?
[17:21] <zambo> yeah, if it uses command line shell cat
[17:22] <shahriman0> it does not use command line shell cat
[17:22] <zambo> I will take a look, thanks again
[17:34] <virtus_> suppose at first I only want the RGB channels, what is wrong with this comment:
[17:34] <virtus_> ffmpeg -i input.flv -vf "lutrgb=r=val:g=val:b=val:a=0" -vcodec libx264 -vpre slow -vpre ipod640 -b 200k -r 50 output_rgb.mp4
[18:16] <iam8up> i'm trying to get one frame out of an rtsp, so 1) is it possible to do this with ffmpeg and 2) is this the right idea/what's wrong? ffmpeg -i rtsp://admin:passwd@10.10.10.192:554/live/ch00_0 -an clip.jpg and here's my problem/log: http://pastebin.com/XhdTbd8a
[20:06] <dericed> could this be true? "you WILL get a better and more accurate result in up-conversion from SD to HD if the original analog video was quantized at 10 bit and you keep the 10 bit data" [as opposed to scaling with 8 bit]
[20:12] <cbreak> analog?
[20:12] <cbreak> like pal?
[20:12] <cbreak> or ntsc?
[20:12] <cbreak> those are terrible
[20:16] <shahriman0> sales people will say anything
[20:26] <zap0> what comes off cameras is analog.
[21:04] <zambo> When I try to use the concat protocol with as a parameters for -i, I get a file/directory not found error.  I know the files are there, is it an ffmpeg build/config issue?
[21:37] <jesk> why does ffserver stops responding to any http request as soon as those few multicast options are configured?
[22:20] <function1> is there some way i can tell ffmpeg to use the libavformat implementation of rtmp rather than librtmp? man page seems abiguous... there is a generic listing for rtmp and then one specifically for librtmp, either simply uses rtmp://<server>...
[22:26] <saste> function1: if you compiled with --enable-librtmp, ffmpeg always favors librtmp implementation
[22:26] <saste> if you want to use native implementation don't use --enable-librtmp
[22:26] <saste> please suggest how to change docs in case it is not clear
[22:38] <function1> actually... it is clear, i read it too quickly :)
[22:38] <nunofgs> can anyone help me? I'm using libavcodec/libx264 to encode frames in h264 and sending them through RTMP. When I try to playback the rtmp stream I get errors like: [h264 @ 0x7fef50828400] Missing reference picture and decode_slice_header error and concealing 396 DC, 396 AC, 396 MV errors
[22:39] <nunofgs> but in flash media server I can preview the stream just fine. it's just when I try to use ffmpeg/libav to grab the stream that that happens
[22:41] <nunofgs> my encoding must be wrong somehow. I suspect it could be the PTS. does anyone know if I also have to set the DTS and Duration in this scenario?
[23:49] <Freakshow> nunofgs: did you post a pastebin?
[00:00] --- Fri Jun 29 2012