[Ffmpeg-devel-irc] ffmpeg.log.20160818

[burek]

[00:33:09 CEST] <CFS-MP3> [ffmpeg/video] h264: co located POCs unavailable
[00:33:12 CEST] <CFS-MP3> What does that mean?
[00:34:13 CEST] <DHE> stream corruption. I've seen that if the first frame isn't a keyframe
[04:01:28 CEST] <i336> Hi. How can I get monitor the output of a filter graph using -sdl with ffmpeg?
[04:01:31 CEST] <i336> I'm using: ffmpeg -f v4l2 -input_format yuyv422 -framerate 30 -video_size 640x480 -i /dev/video0 -i /dev/video1 -filter_complex "[0:v]setpts=PTS-STARTPTS, transpose=1, pad=iw*2:ih[bg]; [1:v]setpts=PTS-STARTPTS, transpose=2[fg]; [bg][fg]overlay=w" -vcodec libx264 -tune zerolatency  -preset slow -f h264 -b:v 1000k udp://127.0.0.1:9999
[04:02:05 CEST] <i336> I'd like to figure out where to cram -sdl in there so that I can monitor the output of the filter graph - -sdl reports that it can't monitor both input sources.
[04:03:30 CEST] <i336> The reason I want to do _that_ is because udp:// introduces about 15 seconds of latency, for some reason, even over localhost - if I can minimize that to maybe a second that would be a workable solution too
[04:04:09 CEST] <i336> Oh, and "ffmpeg -b:v 10000k - | ffplay -" is really slow to update as well.
[06:45:05 CEST] <c_14> i336: -f sdl blah
[08:25:48 CEST] <kdehl> kepstin: I can't find any pointer to the start of the image.
[08:26:34 CEST] <kdehl> I'm using DecodeFrameNoDelay(), which is recommended according to the comments in the header file.
[08:27:38 CEST] <kdehl> The only information I get is iBufferStatus, uiInBsTimeStamp, uiOutYuvTimeStamp, width, height, format and stride.
[09:40:23 CEST] <viric> I'm finally testing shotcut
[10:08:17 CEST] <Mateon1> Hi, I have a quick question. How do I convert an audio file to raw PCM? `ffmpeg -i file.ext -c:a u16le file.u16.raw` doesn't work, it says it's unable to find suitable output format for file.u16.raw, and then says it's an invalid argument
[10:10:55 CEST] <Mateon1> log: http://pastebin.com/vjqJuW3k
[10:11:32 CEST] <Mateon1> Actually, here I use -acodec, but -c:a gives the same result
[11:43:07 CEST] <flux> mateon1, ffmpeg -y -i input.wav -f s16le -ar 48k ${base}.pcm has worked for me
[11:43:38 CEST] <Mateon1> flux: Ah, so did I miss a few options? What does -y and -ar do?
[11:43:53 CEST] <flux> oh, well -y is force overwrite, but -f is the more important one
[11:44:16 CEST] <flux> it forces the output file format
[11:44:23 CEST] <Mateon1> Ah, so -f, not -c
[11:44:50 CEST] <flux> I think -f is like mpeg4, avi when -c is like h264 or mp3
[11:45:16 CEST] <Mateon1> flux: Thanks, that worked perfectly, by the way
[11:47:58 CEST] <Mateon1> Also, I just tried playing the source wav file in ffplay, and the Fourier transform visualization built in is better than in my actual audio analysis toolset... Where can I see documentation on that, or the sourcecode?
[11:49:07 CEST] <viric> Mateon1: remember that it is lgpl :)
[11:54:27 CEST] <durandal_1707> Matron: you haven't tried showspectrum filter, it is that on steroids
[11:56:13 CEST] <durandal_1707> Mateon1: see above
[11:56:29 CEST] <Mateon1> Okay, I'll try that
[12:00:03 CEST] <Mateon1> Hm, I'm not quite sure how to pass the audio through a filter
[12:00:27 CEST] <Mateon1> ffplay file -af showspectrum fails, while -vf doesn't do anything
[12:01:31 CEST] <Mateon1> http://pastebin.com/8k38TRjK
[12:01:56 CEST] <viric> ffplay -vf showspectrum file.mp3    looks good
[12:02:25 CEST] <viric> ah, the same spectrum as without the -vf
[12:08:44 CEST] <Mateon1> I found the source code, and I'm both scared and amazed
[12:09:52 CEST] <Mateon1> I'm amazed that all the source code for ffplay is just 135k of C code, I'm scared it's all in a single file, with not the best looking code I've seen.
[12:10:22 CEST] <durandal_1707> viric: Dont use ffplay, use ffmpeg
[12:11:12 CEST] <durandal_1707> Mateon1: see showspectrum code
[12:11:49 CEST] <durandal_1707> It's 1k lines
[12:15:08 CEST] <viric> The manual has a oneline that shows how to use showspectrum in ffplay
[12:15:11 CEST] <viric> (man ffmpeg-filters)
[12:15:38 CEST] <durandal_17> https://www.youtube.com/watch?v=DJwX0KfGmmo
[12:17:31 CEST] <viric> ffplay -f lavfi 'amovie=jovial.mp3, asplit [a][out1]; [a] showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt [out0]'
[12:29:44 CEST] <Mateon1> Wow, now I wonder if I can hook up a microphone to this thing
[14:03:59 CEST] <miller15> Hey everyone, I'm trying to get thumbnails from a video every 5 seconds. I have tried ffmpeg -i input.mp4 -vf fps=1/5 out%d.jpg, but it seems to be grabbing the frames in this order. 0s, 2s, 7s, 12s, 17s
[14:04:26 CEST] <miller15> Has anyone encountered this before?
[14:06:13 CEST] <miller15> Also using -ss 5 gives frames from the following times, 5s, 7s, 12s, 17s
[14:08:53 CEST] <BtbN> try a diffrent rounding method
[14:09:07 CEST] <BtbN> like, zero or inf
[14:09:38 CEST] <BtbN> Or just plain down. The default near might behave weird for a that low framerate.
[14:14:33 CEST] <miller15> forgive me, but different rounding method?
[15:11:29 CEST] <miller15> I tried fps=0.20 but still the same output occurring. hmm
[15:14:12 CEST] <durandal_1707> rounding
[15:14:48 CEST] <DHE> miller15: https://ffmpeg.org/ffmpeg-filters.html#fps-1 <-- read
[15:18:19 CEST] <miller15> ah thank you very much, I think that's got it. I used :round=zero
[15:20:10 CEST] <miller15> going to test with a few videos
[15:24:56 CEST] <maziar> what does it mean "frame=28636 fps=257 q=28.0 size=N/A time=00:19:05.96 bitrate=N/A"
[15:28:11 CEST] <durandal_17> that you are not using latest ffmpeg
[15:29:59 CEST] <maziar> durandal_17 you talk with me ?
[15:33:12 CEST] <durandal_17> yes
[18:47:27 CEST] <jpack> Hello, I use FFMPEG to mix two audio files that are in .m4a format. The audio files are rarely over 2 minutes long, but the command takes a bit long for the user. I was curious if anyone sees anything that could optimize for speed with this. http://pastebin.com/FfpNTKP8 Thank you
[18:57:45 CEST] <durandal_17> jpack: why volume=1? that does nothing
[18:58:15 CEST] <durandal_17> jpack: pan may be slow, try mixing with swresample
[18:58:25 CEST] <jpack> I normally have a float variable there but just placed it to 1 to avoid confusion in it.
[18:59:04 CEST] <durandal_17> jpack: what "bit long" means?
[19:02:19 CEST] <jpack> I can't seem to find the swresample documentation durandal_17
[19:06:01 CEST] <viric> durandal_1707: I think you suggested shotcut to me; I thought I tried it before, but I did not. I tried it today... it's quite bad at handling 1h long clips
[19:06:10 CEST] <viric> durandal_1707: any change of zoom in the timeline takes ages
[19:10:00 CEST] <jpack> I can't seem to find the swresample documentation durandal_17
[19:20:29 CEST] <durandal117> jpack: how long it takes? how long you want it should take?
[19:41:04 CEST] <ChocolateArmpits> Something might be up with read speed, write speed
[22:45:09 CEST] <tab1293_> what are the accepted formats when specifying bitrates? ive only seen -b:a 256k
[22:45:21 CEST] <tab1293_> are there other suffixes that are valid?
[22:46:10 CEST] <parsec> tab1293_, I've never needed anything else
[22:47:22 CEST] <furq> you can use m or g
[22:47:27 CEST] <furq> not that you'd need to
[22:47:37 CEST] <furq> https://ffmpeg.org/ffmpeg.html#Options
[22:47:41 CEST] <tab1293_> and a number by itself is just bits?
[22:48:02 CEST] <furq> you can also use B for bytes, and KB/MB/etc
[22:48:06 CEST] <furq> yes
[22:48:34 CEST] <tab1293_> cool thanks!
[00:00:00 CEST] --- Fri Aug 19 2016