[Ffmpeg-devel-irc] ffmpeg.log.20130612

[burek]

[02:43] <SKiTZO> how can i force rgb24  colorspace when output video codec is ffv1?
[03:10] <relaxed> SKiTZO: with -pix_fmt
[03:11] <relaxed> SKiTZO: look at `ffmpeg -h encoder=ffv1` for supported pixel formats
[03:14] <llogan> or the format filter if you're already filtering (i'm late to the conversation)
[03:20] <relaxed> SKiTZO: ffvhuff suports rgb24
[04:22] <mudkipz> hello, if I just switched to pulse and I'm having sync issues. If I call ffmpeg with options like this -f pulse [other options] -i default are the other options just informative or will they actually resample the audio?
[04:23] <mudkipz> the reason is that I think that pulseaudio is set to 44.1khz by default but my ffmpeg seems to be detecting it as 48khz
[04:23] <mudkipz> This is version 1.2.1 on archlinux, by the wa.
[04:24] <relaxed> try ffmpeg -ar 48000 -f pulse -i default ...
[04:24] <mudkipz> I'm debating between trying to change the default pulseaudio sampling rate to 48khz or somehow mess with ffmpeg's
[04:24] <mudkipz> okay
[04:28] <mudkipz> by the way, here's my command before I came in here.
[04:28] <mudkipz> http://pastebin.com/6d5rvhKR
[04:28] <mudkipz> some of those are variables in a bash script, but they shouldn't make any difference.
[04:28] <mudkipz> just that I'm outputting to flv and resampling to 44.1khz since flv requires that.
[05:16] <Bourbon> what's up with videolan's git repo recently - is it a github corporate repo or something?
[05:16] <Bourbon> just curious - I'm not too indignant that my access to free software is being impeded
[08:06] <mudkipz> I ended up doing a number of things. I was checking the pulse stream with ffprobe and it seemed to pull a sample rate of 48khz no matter what I set pulse to in the daemon.conf file. Eventually I just set pulse to 48khz so now ffmpeg resamples it to 44.1khz (which I need to output).
[08:07] <mudkipz> This made my audio stay in sync much longer but it now goes out of sync by a couple seconds every hour or so.
[08:08] <mudkipz> I'm pushing video (from x11grab) and audio (from pulse) out to an rtmp streaming server. The video stays synced but the audio lags behind.
[08:09] <mudkipz> I normally just capture alsa for streaming but I recently converted my system to pulse. The reason is that I installed a new videocard and use the hdmi for output. Using pulse for audio is overall less problematic than alsa.
[08:10] <mudkipz> Is there something I could have misconfigured with my new pulseaudio setup? I mean looking around the internet I'm not seeing a huge amount of users complaining about audio out of sync issues.
[08:11] <mudkipz> The main thing is that the audio isn't out of sync the whole time, it goes out of sync over time.
[08:13] <mudkipz> am I supposed to be using "-f pulse -i default" or "-f alsa -i pulse"
[08:13] <mudkipz> ?
[11:09] <luc4> Hi! I'm using ffmpeg to mux h264 data into an mp4 container. The result seems to be played by most players but it seems that android fails for some files to generate a thumb. Do you have any suggestion on how to check whether my muxer is not muxing entirely correctly?
[11:11] <khali> luc4: I have seen improperly generated thumbnails on my HTC Desire C phone (Android 4.0 ICS) as well
[11:11] <khali> luc4: sometimes even for files generated the very same way, some were thumbnailed OK and some were not
[11:11] <luc4> khali: the thumb is not generated at all it seems. But the player plays the mp4 perfectly.
[11:12] <khali> luc4: I noticed that the problem was more likely to happen when the width wasn't a multiple of 16
[11:12] <luc4> khali: but it seems that gstreamer also is failing to play the file, so I was suspecting something is wrong...
[11:12] <khali> luc4: and I also had trouble playing some files when the initial time difference between audio video was more than one frame
[11:13] <khali> my conclusion was that whatever player is used by default on Android 4.0 is crappy
[11:14] <luc4> khali: and the fact that gstreamer can't play it? Isn't that strange?
[11:14] <khali> luc4: it certainly is
[11:15] <khali> luc4: but it would help if it would tell you why
[11:15] <khali> guessing is just too difficult
[11:15] <khali> with the Android player you can only guess, but with gstreamer it should be able to get the exact reason for failure
[11:16] <luc4> khali: do you have any advice on how to get some logs from gstreamer?
[11:17] <khali> luc4: start the played from the command line with the relevant verbosity / debugging options?
[11:17] <khali> luc4: what player are you using?
[11:19] <luc4> khali: I tried totem.
[11:19] <luc4> khali: maybe there is something better which uses gstreamer?
[11:23] <luc4> khali: the totem video thumbnailer seems to give proper logs, but the error is not very clear.
[11:32] <khali> luc4: I only know of totem and me-tv using gstreamer, but I'm sure there are others
[11:35] <luc4> khali: in the logs of gstreamer I see this: h264parse gsth264parse.c:1822:gst_h264_parse_set_caps:<h264parse0> avcC size 11 < 8 and h264parse gsth264parse.c:1837:gst_h264_parse_set_caps:<h264parse0> refused caps video/x-h264, stream-format=(string)avc, alignment=(string)au, level=(string)4, profile=(string)baseline, codec_data=(buffer)01428028ffe10000010000, width=(int)640, height=(int)480, framerate=(fraction)1000/57, pixel-aspect-ratio=(fraction)1/1. Maybe
[11:35] <luc4> something is wrong in my extradata?
[11:39] <khali> luc4: this goes beyond my skills, sorry
[11:39] <luc4> khali: no problem, thanks for your help!
[11:46] <luc4> Anyone who knows what byte order should be given to SPS length when placed in extradata?
[13:56] <Kolibry> hi everyone, i'm looking for help, i'm just about to give up on a thing that i think is not that complicated, but really is painful for a noob like me :/
[13:56] <Kolibry> i just want to put 2 videos in one
[13:56] <Kolibry> with concat command
[13:56] <Kolibry> the problem is : the videos don't have the same settings
[13:57] <ubitux> then use the concat filter
[13:57] <Kolibry> so concat succeed, but it's laggy on the second half on my new video
[13:57] <Kolibry> ffmpeg -y -i "mur.mp4" -c copy -bsf:v h264_mp4toannexb -f mpegts "mur.ts"
[13:57] <Kolibry> ffmpeg -y -i concat:"mur2.ts|mur3.ts" -absf aac_adtstoasc -c copy -f mov "destination.mp4"
[13:57] <Kolibry> here are the commands i used
[13:58] <ubitux> ah, well then just change the frame rate
[13:58] <ubitux> with -r
[13:58] <Kolibry> where do i put the -r ?
[13:59] <ubitux> as output option in the first one i would guess
[14:01] <Kolibry> [NULL @ 0x9456a80] Requested output format '-r' is not a suitable output format
[14:01] <Kolibry> mpegts: Invalid argument
[14:01] <Kolibry> i must specify a number ?
[14:02] <khali> Kolibry: you probably mixed two options on the command line, to get the error above
[14:02] <khali> (-r and -f)
[14:03] <Kolibry> i've just put the -r in the first command i gave you
[14:03] <khali> Kolibry: and yes you must specify a frame rate after -r
[14:03] <Kolibry> yes r and f
[14:03] <Kolibry> i remove -f then
[14:03] <khali> no, keep -f
[14:04] <khali> -r 25 -f mpegts
[14:04] <khali> (for example)
[14:04] <Kolibry> i'll try this one
[14:04] <Kolibry> ok great
[14:04] <Kolibry> so i do the same for the second video, and then concat command
[14:07] <Kolibry> still doesn't work :(
[14:08] <Kolibry> the second part of the new video created is still horribly lagging
[14:08] <khali> Kolibry: I am not familiar with the concat command
[14:08] <khali> Kolibry: but I suspect you must ensure that all settings are the same for all input files
[14:09] <Kolibry> maybe i can try to convert each file with same settings
[14:10] <Kolibry> but i'm not familiar with ffmpeg at all :/
[14:11] <Kolibry> ffmpeg -i source_video.avi -b 300 -s 640x360 -vcodec xvid -ab 32 -ar 24000 -acodec aac final_video.mp4
[14:11] <Kolibry> can this command may work ?
[14:11] <Kolibry> or am i missing something here ?
[14:24] <Kolibry> [url=http://www.zimagez.com/zimage/capturedcran-12062013-142430.php][img]http://www.zimagez.com/miniature/capturedcran-12062013-142430.php[/img][/url]
[14:25] <Kolibry> here are the settings of my 2 videos, can someone give me a command to convert the second one like the first one ?
[14:34] <Kolibry> anyone ?
[14:42] <khali> Kolibry: -b 300 is definitely wrong, you meant -b 300k
[14:42] <khali> likewise -ab 32k, not -ab 32
[14:43] <khali> AFAIK xvid needs -vcodec libxvid
[14:52] <Kolibry> Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height
[14:52] <Kolibry> 
[14:52] <Kolibry> 
[14:52] <Kolibry> still doesn't work
[14:52] <Kolibry> amazingly complicated :)
[14:53] <Kolibry> ok
[14:53] <relaxed> oh, -b:a 300k (you meant -b:v 300k)
[14:53] <Kolibry> i don't really know what i mean...i'm pretty blind here :)
[14:54] <relaxed> use -b:a 32k instead of -ab 32k          -b:a = audio bitrate | -b:v = video bitrate
[14:54] <Kolibry> oh, works with v
[14:55] <relaxed> are you targeting some device?
[14:56] <Kolibry> no, just my computer
[15:01] <Kolibry> http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20concatenate%20%28join,%20merge%29%20media%20files
[15:01] <Kolibry> on this page they seem to say it's possible to concatenate without convert each file
[15:02] <Kolibry> in this part : Concatenation of files with different codecs
[15:02] <Kolibry> am i wrong ?
[15:03] <relaxed> <encoding options>
[15:03] <Kolibry> that's the part i was not capable of handling
[15:04] <relaxed> meaning you can't stream copy, so ffmpeg must encode each stream to a similar codec for the output
[15:05] <Kolibry> oh ok :(
[19:09] <jackbaus> is there way to automatically drop rtmp packets in ffmpeg
[19:19] <AISICAO1> no way, i think
[19:23] <schtinky> I'm using "ffmpeg -skip_frame nokey -i (piped mpegts) imagefile%9d.jpg" to grab all the key/iframes from a broadcast mpegts stream
[19:23] <schtinky> right now I've also included -r 1 to grab just one frame per second
[19:23] <schtinky> but if I remove -r 1, I should only get 2 frames per second because broadcast mpegts streams only have 2 keyframes per second
[19:24] <schtinky> instead, I'm getting many, many frames, way more than 2 per second. Anything strike you as odd about that?
[20:33] <anonymou_> I'm using ffmpeg to produce multiple outputs from an input mp3. One of the outputs is the artwork encoded in the mp3 (which may or may not exist). When there is no mp3 artwork, ffmpeg outputs "Output file #0 does not contain any stream" and doesn't write out any of the other outputs. Is there a way to force ffmpeg to write out the other outputs even when there is no video stream for the mp3 artwork?
[21:59] <bunniefoofoo> when encoding to mpeg format, do I need to set bufsize? I thought I only need bufsize if I have a max bitrate (and/or min bitrate) The reason I ask is that for a 480p encode, it seems to work, but for 720p (abr 20mb) i get "packet too large" and "buffer underflow" errors
[22:05] <braincracker> Barbie and Ken http://www.youtube.com/watch?v=yKtmvYt2k3A
[22:05] <braincracker> <;
[22:07] <shroomy> Anyone here have experience with ffmpeg android port ?
[22:08] <shroomy> Not sure if its a configure bug or if I'm too stupid to use make
[22:37] <elkng> there are all sorts of videos on youtube, even if it high definition like that one http://www.youtube.com/watch?v=29uRvifupqQ it has 1920x1080 resolution and 4336 kb/s bitrate, but there could be bitrate of 3000 kb/s or even 2000 kb/s why those bitrates can differ for same resolution, isnt 2000 kb/s enough for 1920x1080 resolution ? or it could differ if you watch it on 17" motinor or on 30" monitor ? when its better to have high bitrate ?
[23:00] <llogan> elkng: it depends on the complexity of the input. a complex, hard to encode scene will use more bits to achieve a specific quality than a blank, white wall.
[00:00] --- Thu Jun 13 2013