[Ffmpeg-devel-irc] ffmpeg.log.20150522

Sat May 23 02:05:01 CEST 2015

[02:39:59 CEST] <t4nk309> hi
[04:09:24 CEST] <Zeranoe> Has anyone been able to compile libmfx with FFmpeg?
[09:11:27 CEST] <Blefuska> hello, I'm having issues while using ffmpeg over rtp (2,6)
[09:12:04 CEST] <Blefuska> is this a propper command:ffmpeg -i <file> -c copy -f rtp_mpegts rtp;//<ip>:port
[09:12:05 CEST] <Blefuska> ?
[09:12:34 CEST] <Blefuska> and can I get ffmpeg to outpu how much of a payload (in bytes) was sent to the rtp recipient?
[12:00:13 CEST] <acidflash> Hi, does this channel offer support for ffserver aswell as ffmpeg?
[12:06:16 CEST] <iive> acidflash: in theory - yes.
[12:11:36 CEST] <Anoia> hi alll
[12:12:12 CEST] <Anoia> I see avcodec_register_all does a check to see if it's already initialised, but no synchronisation
[12:12:50 CEST] <DeadSix27> is there a way to tell ffmpeg to wait for a cameras autofocus and brightness control?
[12:12:53 CEST] <Anoia> if I have two threads call that at the same time, will registering them again cause issues?
[12:13:15 CEST] <DeadSix27> in fact i dont even know if it automatically does that in any case
[12:14:10 CEST] <BtbN> Don't call it in a thread
[12:14:15 CEST] <BtbN> call it once at application startup
[12:18:56 CEST] <Anoia> this is a DLL that won;t have a specific app startup
[12:19:17 CEST] <BtbN> DLLs have DllMain
[12:19:19 CEST] <Anoia> while instantiating both classes that use it at the same time is very unlikley, I just wanted to make sure
[15:10:36 CEST] <animax> hello all together. short question. http://www.pasteall.org/58590. disregarded further settings do I need to use code in line 1 or is it sufficient to take line 3 (without '-f image2')?
[16:04:15 CEST] <animax> second second trial. hello all together. short question. http://www.pasteall.org/58590. disregarded further settings do I need to use code in line 1 or is it sufficient to take line 3 (without '-f image2')?
[16:15:02 CEST] <BtbN> Well, does the second one work?
[16:17:06 CEST] <animax> BtbN: ?
[16:17:35 CEST] <animax> connection was interrupted
[16:17:37 CEST] <BtbN> If it doesn't work, you need the format option.
[16:17:45 CEST] <animax> the POST is the second trial ...
[16:17:56 CEST] <BtbN> You have two commands there
[16:18:01 CEST] <BtbN> Does the second one work?
[16:18:02 CEST] <animax> yes
[16:18:09 CEST] <animax> mom
[16:18:16 CEST] <BtbN> If not, you have your answer ;)
[16:18:33 CEST] <BtbN> I'd assume it'll just try to open that exact filename
[16:18:45 CEST] <animax> the second one does work
[16:19:05 CEST] <BtbN> Interesting
[16:19:17 CEST] <animax> for what does '-f image2' stand?
[16:19:23 CEST] <BtbN> The input format
[16:19:34 CEST] <animax> and when do I need this command?
[16:19:44 CEST] <BtbN> When the input format isn't obvious from the input file
[16:20:58 CEST] <animax> I've  often seend command lines with this part. so if I take i.e. 'image%04d.png'
[16:21:13 CEST] <animax> then I don't need to use '-f format'?
[16:21:50 CEST] <animax> because the format is given by the png-extension?
[16:21:53 CEST] <BtbN> Depends, i guessed it would just try to open a file with that name, but it seems like the default format understands that.
[16:22:56 CEST] <animax> mh. ok.
[16:27:53 CEST] <animax> an other question: is it possible to encode image sequences in a way that single frames get individual durations in the encoded output file like it is possible with GIF in photoshop i.e.? and if so, is there a gui for ffmpeg or x264 which can mange that?
[16:28:22 CEST] <animax> .. manage that ...
[16:38:48 CEST] <t4nk897> hi all, I'm processing AAC-HE files, boxing them in m4a files. Btw I get file to became AAC.
[16:40:22 CEST] <t4nk897> original file is: format: Advanced Audio Codec; Format version: Version 2; Format profile: HE-AAC / LC.
[16:42:36 CEST] <t4nk897> ffmpeg build a file: Format: Advanced Audio Codec; Codec ID: 40
[16:44:44 CEST] <t4nk897> MP4Box builds a file: Format: Advanced Audio Codec; Format profile: HE-AAC / LC; Dodec ID: 67
[16:45:41 CEST] <t4nk897> my car plays MP4Box generated files... it says ffmpeg created files have some errors
[16:46:58 CEST] <t4nk897> MP4Box -add foo.aac foo.m4a -new -v
[16:48:11 CEST] <t4nk897> ffmpeg -i foo.aac -vn -acodec copy -map 0 -movflags faststart foo.m4a
[17:46:53 CEST] <shevy> can ffmpeg convert i. e. a .wav file into an .opus file?
[17:50:05 CEST] <joer1> Got a quick question if anyone can help. I have an rtp stream - audio and video are being sent. I want ffmpeg to merge the two streams into one and save it to disc. Right now I can only seem to grab one at once. Any tips on how to go about doing this?
[17:50:55 CEST] <joer1> Video is h264, audio is Opus so needs transcoding to save both to a target mp4 file
[17:52:19 CEST] <shevy> ah there is a configure option --libopus
[18:02:28 CEST] <joer1> shevy: thanks for that, seems the error was just a missing lib
[18:02:35 CEST] <shevy> huh
[18:02:42 CEST] <shevy> I have had my own problem
[18:02:48 CEST] <shevy> tell me that I managed to solve your problem too :-)
[18:02:59 CEST] <shevy> that's mysterious synergism
[18:03:29 CEST] <shevy> all I am trying is to find out whether I can generate .opus file ... I did not even know about that format until quite recently; so far I understood that I require to have libopus
[18:10:05 CEST] <gr1sha> I have a frame that I want to convert to RGB, manipulate and then convert to YUV
[18:10:14 CEST] <gr1sha> but when I try to convert it to YUV I get [swscaler @ 0x7f966c05e440] bad src image pointers
[18:10:23 CEST] <gr1sha> how can I trace the cause of the error?
[18:27:18 CEST] <nuxusr> I just updated from a really old version of ffmpeg to the latest. I'm seeing this weird thing where users who's mobile connection is on the Telus cellular in Canada can no longer stream music.  Its only on cellular though, if they use wifi they can stream the music again.  The streams don't work in the browser either for them but the weird thing is that they work when using the old ffmpeg builds.  here is an example stream htt
[18:27:19 CEST] <nuxusr> p://stream.musiconeradio.com/musicone/aac/48k
[18:28:40 CEST] <shevy> file test.opus -> test.opus: Ogg data
[18:28:41 CEST] <shevy> hmm
[18:31:20 CEST] <ac_slater_> hey guys. Using libavformat, etc ... I get PTS values from a video stream in intervals of 3000, starting at 126000. Is it possible to transform this value into something I can compare against av_gettime_relative() ? ie - I want to do rate emulating
[18:31:25 CEST] <ac_slater_> emulation *
[18:33:53 CEST] <jrun> hello, does Tegra K1 do NVENC?
[18:34:28 CEST] <jrun> oh before that actually, does ffmpeg support x265 enconding within NVENC?
[18:36:15 CEST] <jrun> i'm looking for a dedicated encoder for home automation (doorbell/intecomm)/security video streaming over gigabit LAN but hear different opinions about whether it's better to do it on a asic or cpu
[18:37:05 CEST] <jrun> apparently previous attemps by gpu makers (CUDA-based for example) have failed compared to sofware encoding so i'm wondering what's the latest state of affairs
[18:50:22 CEST] <gr1sha> I've managed to convert the video to RGB now I want to change R G or B values, what's the correct way to do it? When I try to access data[i] I get segmentation faluts
[19:05:09 CEST] <ac_slater_> jrun: I have an ARM cortex A10 that can encode 1080p video at 30hz using ffmpeg
[19:05:23 CEST] <ac_slater_> ie - no asic or FPGA or GPGPU required... only neon
[19:05:54 CEST] <ac_slater_> jrun: but in ffmpeg2.5, I tried to vdpau based h264 encoder and it did use my gpu
[19:05:59 CEST] <ac_slater_> not sure if it's still there
[19:30:26 CEST] <joer1> Is anyone familiar with the errors "pts has no value" and "application provied duration... is out of range for mov/mp4 format" when trying to combine an audio and video stream?
[19:44:09 CEST] <rjp421> using the latest git static build on centos6, how can i optimize this cmd http://pastebin.com/vN7PP0DM so it still creates livestream1 2 and 3 in the same order, but duplicates the source stream without wasting resources out to both ustream and live/livestream?
[19:58:47 CEST] <shevy>       ENCODER         : Lavc56.26.100 libopus
[19:58:51 CEST] <shevy> does this mean it is opus?
[19:58:56 CEST] <shevy> because when I use "file" I get this
[19:59:06 CEST] <shevy> file output.opus
[19:59:06 CEST] <shevy> output.opus: Ogg data
[19:59:12 CEST] <shevy> and that confuses me. is file wrong?
[19:59:22 CEST] <shevy> The commandline I was using was this here:
[19:59:25 CEST] <shevy>   ffmpeg -i 2.wav -ar 48000 -ac 2 -acodec libopus -ab 256k output.opus
[20:01:56 CEST] <cilly> hi
[20:03:21 CEST] <cilly> I am having problems with -map_metadata 0 option and mp4 as output. The metadata 'location' aka geo location isn't copied. It looks like this bug: https://trac.ffmpeg.org/ticket/4337
[20:03:28 CEST] <cilly> Is there a patch or solution?
[20:03:55 CEST] <cilly> btw. while using output .mkv, the location is applied, but mkv doesn't play on OS X without additional tools.
[20:04:07 CEST] <Nitori> shevy, opus is a codec, ogg is a container. try to: ffprobe output.opus and take a look at the streams
[20:04:44 CEST] <cilly> oki
[20:05:23 CEST] <shevy> yeah ffprobe gives me that ENCODER line about libopus; I am confused about the unix/linux "file" command, it seems to be working ok for e. g. .mp3 files
[20:05:26 CEST] <shevy> "Audio file with ID3 version 2.4.0, unsynchronized frames, contains: MPEG ADTS, layer III, v1, 192 kbps, 44.1 kHz, JntStereo"
[20:05:39 CEST] <shevy> is opus too recent for file to understand it?
[20:05:50 CEST] <Nitori> cause ".opus" extension is actually ogg AFAIK
[20:05:54 CEST] <shevy> oh
[20:06:05 CEST] <shevy> ok then that would make sense
[20:06:05 CEST] <Nitori> opus is from the same people doing ogg vorbis
[20:06:15 CEST] <shevy> thanks Nitori
[20:17:59 CEST] <cilly> well, currently while converting video files with ffmpeg which contain location metadata seems not to work for: mp4, mkv, m4v.
[20:20:17 CEST] <cilly> I see:
[20:20:18 CEST] <cilly> http://its.ffmpeg.org/ticket/4209
[20:51:47 CEST] <rjp421> when streaming a pi cam with raspvid -fps 15 piped to ffmpeg -c:v copy -f flv 'rtmpuri' shows fps=15 during encode but the metadata object's framerate property is always 25
[20:52:18 CEST] <rjp421> using the latest git compiled last night on both pi model B+ and pi2
[22:45:38 CEST] <jasonwhite> Hi guys I have a question and I hope some guru here could help me out :D I have file .m4a download from youtube labeled as M4A (Audio Only) I cannot play it with my music player, Audacious but can play with VLC. I run ffmpeg to see, it turns out to be an Audio(AAC) stream contained inside a video container(Mp4) if my knowledge is right. Can I "split" this audio  stream out to AAC audio instead of using libfdk_aac to "convert" ?
[22:46:27 CEST] <Mavrik> yes you can remux that audio
[22:46:32 CEST] <Mavrik> to what do you want?
[22:47:06 CEST] <jasonwhite> Mavrik : I just want to get that audio stream to native AAC audio file
[22:47:09 CEST] <Mavrik> Also are you sure you're not just missing AAC plugin for audacious?
[22:47:16 CEST] <Mavrik> jasonwhite, what is a native AAC audio file?
[22:47:37 CEST] <Mavrik> jasonwhite, by standard MPEG-4 AAC audio should be inside a MPEG-4 container (usually marked mp4 or m4a)
[22:47:47 CEST] <Mavrik> so there's nothing wrong with that - YouTube uses "native AAC"
[22:47:50 CEST] <Mavrik> that's why I'm asking.
[22:48:51 CEST] <jasonwhite> Mavrik : I still can play other m4a files as well. I will give you the output of ffprobe
[22:50:36 CEST] <jasonwhite> Mavrik : It said that the file is mp4a stream. Humm.. Not sure what's wrong here considering what you said is right.
[22:50:54 CEST] <Mavrik> jasonwhite, try just remuxing it
[22:51:04 CEST] <Mavrik> ffmpeg -i <file>.m4a -codec copy temp.m4a
[22:51:08 CEST] <Mavrik> and see if it's ok then
[22:52:03 CEST] <jasonwhite> Mavrik : Sure. I will.
[22:56:47 CEST] <jasonwhite> Mavrik : It plays ! It did compare ffprobe of original file and remuxed file, turns out : original file : major_brand : dash and remuxed file : major_brand : M4A.
[22:57:08 CEST] <Mavrik> heh, wierd :)
[22:57:12 CEST] <jasonwhite> Mavrik : Is it the problem ?
[22:57:25 CEST] <Mavrik> could be yes, if audacious does strict checking for that marker
[22:57:30 CEST] <jasonwhite> *source of problem that Audacious/music player cannot play
[22:57:43 CEST] <jasonwhite> Mavrik : Yeah you are right
[22:59:41 CEST] <jasonwhite> Mavrik : Thank you. Saved me a lot of headache...
[00:00:00 CEST] --- Sat May 23 2015