[Ffmpeg-devel-irc] ffmpeg.log.20150921

[burek]

[00:04:48 CEST] <BullHorn> hey BtbN is the build released today including the fix from yesterday?
[00:06:42 CEST] <BullHorn> looks like it is
[00:06:45 CEST] <BullHorn> :)
[00:09:24 CEST] <Rudde> Wait
[00:09:29 CEST] <Rudde> BtbN is here too? :o
[00:11:09 CEST] <BullHorn> yes https://www.youtube.com/watch?v=TAGbrM-MMRk
[00:36:29 CEST] <squeegily> ffmpeg -f alsa -i pulse -y test.wav gives (snd_config_hooks_call) Cannot open shared library libasound_module_conf_pulse.so
[00:36:31 CEST] <squeegily> even though /usr/lib/{i386,x86_64}-linux-gnu/alsa-lib/libasound_module_conf_pulse.so exist
[00:36:33 CEST] <squeegily> Why is this?
[00:40:49 CEST] <c_14> try setting your LD_LIBRARY_PATH to include that dir
[00:41:55 CEST] <squeegily> That didn't work
[00:41:57 CEST] <squeegily> Nor did LD_PRELOAD=/usr/lib/x86_64-linux-gnu/alsa-lib/libasound_module_conf_pulse.so ffmpeg -f alsa -i pulse -y test.wav
[00:42:18 CEST] <c_14> Have you tried -f pulse -i default ?
[00:43:34 CEST] <squeegily> I don't have access to "real" pulse support
[00:43:51 CEST] <squeegily> The only build I have access to only has alsa support
[00:44:05 CEST] <squeegily> But arecord -D pulse works perfectly
[00:44:36 CEST] <squeegily> So I just need to figure out why ffmpeg cannot open libasound_module_conf_pulse.so
[00:51:02 CEST] <c_14> You could always build a copy.
[00:51:04 CEST] <c_14> It's not that hard.
[00:51:56 CEST] <waressearcher2> can I upload 1024x768 video on youtube or should I convert it to 1280x720 ? any disadvantages of aploading nonstandart resolution ?
[01:13:03 CEST] <joeschmo123> you can upload anything to youtube, i think
[02:03:38 CEST] <DHE> waressearcher2: youtube will resize it down to 720p. given the resolution difference is so minor it probably doesn't matter.
[02:03:51 CEST] <waressearcher2> minor ?
[02:04:04 CEST] <waressearcher2> 1024x768 and 1280x720
[02:04:20 CEST] <waressearcher2> if you increase 1024 to 1280 you need to increase 768 to 960
[02:04:28 CEST] <waressearcher2> and 960 is much bigger than 720
[02:04:39 CEST] <waressearcher2> so will it strech the video then ?
[02:30:35 CEST] <fred1807> I need to convert some videos to "raw h264"  to play in raspberry pis ' hardware acelerated payer (video.c program)... For that I am following the given instructions  " ffmpeg -i movie.mp4 converted.h264... It is working for .mp4 videos... But for some .mov videos, the playback is totally screwed in the raspberry... So I wonder, Is there anything more I can add to this line, to force the output format to this so called "raw h264" , so I can keep this 1
[02:30:36 CEST] <fred1807> "universal" conversion command ? Thank you.
[02:35:56 CEST] <waressearcher2> I want to concatenate 2 files, I use "ffmpeg -f concat -i list.txt -c copy -y -f avi out.avi" but I want it started concatenation of first file from 20 second position, is there a way to tell ffmpeg to seek 20 second in the first concatenated file ?
[02:47:32 CEST] <wolfman2000> Evening. Out of curiousity, does anyone know of a mirror of ffmpeg that uses cmake for generation and building instead of autoconf/automake/autotools/autosomething?
[10:42:35 CEST] <luc4> Hello! I'm "porting" my code from ffmpeg libs 2.7.2 to ffmpeg libs 2.8 and it seems not to work with some video files. Unfortunately even by debugging I'm not able to find what changed from 2.7.2 to 2.8. Anyone who can advise on what I could to determine what changed?
[10:45:30 CEST] <JEEB> is it something that can be tested with the ffmpeg command line application
[10:46:25 CEST] <luc4> JEEB: how would you test it?
[10:47:39 CEST] <luc4> JEEB: I have to say I dumped data from each encoded frame ffmpeg returns. No difference between 2.7.2 and 2.8. Same exact bytes. Also same pts and dts.
[12:14:46 CEST] <squ> how to extract audio (.wav) from .mkv
[12:19:00 CEST] <DHE> ffmpeg -i input.mkv -map 0:a output.wav
[12:26:45 CEST] <Nindustries> Hi, so I was playing around with ffmpeg and wanted to convert a mov into a mkv. I used Big Buck Bunny and used this command;  ffmpeg -i big_buck_bunny_1080p_h264.mov -vcodec ffv1 -acodec pcm_s16le bunny.mkv  bunny.mkv is a 8.8GB file now, while the original was around 900MB. :O
[12:28:28 CEST] <relaxed> welcome to lossless
[12:39:16 CEST] <Nindustries> heh
[12:39:31 CEST] <Nindustries> so it decompressed the file?
[12:39:46 CEST] <relaxed> flac and libx264 lossless would shave some off
[12:40:25 CEST] <Nindustries> I actually just wanted to switch containers
[12:40:45 CEST] <relaxed> ffmpeg -i input -c copy output
[12:41:31 CEST] <Nindustries> yeah
[12:41:34 CEST] <Nindustries> Just found that one out
[12:41:35 CEST] <Nindustries> Thanks
[12:48:57 CEST] <chintak> Hi all
[12:49:31 CEST] <chintak> I want to non-linearly speed the video
[12:50:12 CEST] <chintak> ffmpeg command line interface let's me modify the speed in a linear manner
[12:50:22 CEST] <chintak> Hence, I started learning the C API
[12:50:55 CEST] <chintak> I'm following the decoding_encoding.c example here: https://www.ffmpeg.org/doxygen/2.1/doc_2examples_2decoding_encoding_8c-example.html
[12:53:48 CEST] <chintak> The example implements a video decoder for a stream based codec. In my application I want to read an mp4, which is a frame based codec. So how should I modify the for loop inside the video_decode_example() (pasted here: http://pastebin.com/aq4xwx1q) for reading a frame
[12:53:55 CEST] <chintak> any other examples?
[12:54:12 CEST] <chintak> where and how should I specify the width and height mentioned?
[13:38:42 CEST] <Nindustries> Hi, where can I find a list of all supported video containers? (mkv, mp4, ..)
[13:56:48 CEST] <cbsrobot_> Nindustries: ffmpeg -formats
[14:11:49 CEST] <Nindustries> cbsrobot_, hmm, is there a way to only show video formats?
[14:11:53 CEST] <Nindustries> and not audio
[14:13:01 CEST] <JEEB> yeah, I'm not sure such grouping exists
[14:13:23 CEST] <JEEB> generally muxers just limit the codecs by codec ids when you try to stuff 'em in
[14:14:59 CEST] <Nindustries> I'm trying to limit ffmpeg input by checking an extension against a list
[14:19:25 CEST] <JEEB> rather use ffprobe and check if it has video?
[14:19:34 CEST] <JEEB> it has json output which you can parse
[14:26:54 CEST] <Nindustries> thanks JEEB
[16:21:04 CEST] <blue112> Hi here.
[16:21:19 CEST] <blue112> Is it the good place to ask about libav usage in CPP ?
[16:22:13 CEST] <fritsch> no
[16:22:31 CEST] <blue112> On which channel should I ask, then ?
[16:22:35 CEST] <fritsch> you could try #intel-gfx but that will most likely not be that helpful, too
[16:22:49 CEST] <fritsch> depending on what you are doing, check mpv or kodi code
[16:22:55 CEST] <c_14> hmm?
[16:23:02 CEST] <c_14> Assuming he means the libav* libraries, he can ask here.
[16:23:13 CEST] <fritsch> ouh :-)
[16:23:14 CEST] <c_14> Though he might get better answers on the libav-users ffmpeg mailing list.
[16:23:18 CEST] <blue112> Yeah, libav libraries, like the avcodec_decode_video2 and such
[16:23:24 CEST] <fritsch> blue112: sorry i confused it with libva
[16:23:30 CEST] <blue112> Ok, no problem.
[16:23:38 CEST] <fritsch> blue112: yeah, welcome here :-) and right channel
[16:23:49 CEST] <blue112> Thanks :)
[16:24:31 CEST] <blue112> My problem is I get crappy audio, full of cracks. I've decicated a thread to decoding audio and another to playing audio, but still, it's crappy
[16:24:47 CEST] <fritsch> did you try to dump the data?
[16:24:48 CEST] <blue112> I'm decoding in real time a 1080P video, with YUV to RGB conversion.
[16:25:01 CEST] <fritsch> the audio data and play that afterwards? is this fine?
[16:25:02 CEST] <blue112> fritsch, I'm not sure in which format I could dump it.
[16:25:22 CEST] <fritsch> some raw pcm out
[16:25:48 CEST] <blue112> I'm trying thart
[16:26:03 CEST] <fritsch> to see if you decode the error into your samples
[16:26:24 CEST] <Mavrik> blue112, cracking audio usually means you have holes in samples, wrong samplerate, wrong data interpretation or something simillary
[16:26:33 CEST] <Mavrik> you're messing something up on the decoded buffers
[16:27:05 CEST] <blue112> I'm trying to dump them
[16:30:24 CEST] <blue112> fritsch, if I dump the audio data, after I've reencoded it with avresample_convert, the audio is ok, without cracks.
[16:43:52 CEST] <blue112> Mavrik, any idea of where I could look?
[16:47:02 CEST] <ewew> @Mavrik How is output from device ?
[17:02:07 CEST] <fling> Is not it possible to convert from ratdvd?
[17:11:53 CEST] <luc4> Hello! I'm "porting" my code from ffmpeg libs 2.7.2 to ffmpeg libs 2.8 and it seems not to work with some video files. Unfortunately even by debugging I'm not able to find what changed from 2.7.2 to 2.8. Anyone who can advise on what I could to determine what changed?
[17:19:36 CEST] <maksbotan> hi
[17:19:50 CEST] <maksbotan> found this hilarious error with -f hls -hls_flags delete_segments
[17:19:57 CEST] <blue112> Hi maksbotan
[17:20:11 CEST] <maksbotan> http://i.imgur.com/tX2PLUP.png
[17:20:16 CEST] <maksbotan> see on screenshot ^
[17:20:20 CEST] <maksbotan> should i report it? :D
[17:26:40 CEST] <DHE> maksbotan: I've seen that as well. yeah looks like something's getting corrupted
[17:26:58 CEST] <maksbotan> DHE: thanks for your info
[17:27:05 CEST] <maksbotan> i'll try to file bug report when i have time
[17:27:18 CEST] <DHE> I had HLS working fine a few months back, so that's something introduced recently
[17:27:36 CEST] <maksbotan> DHE: do you use HLS frequently?
[17:27:43 CEST] <DHE> lately yeah
[17:27:44 CEST] <maksbotan> i have a couple of newbie's qeustions
[17:27:48 CEST] <maksbotan> questions*
[17:27:52 CEST] <DHE> sure
[17:28:13 CEST] <maksbotan> i've noticed that ffmpeg writes playlist file only after first fragment (.ts) file is written
[17:28:18 CEST] <maksbotan> leading to a delay on start
[17:28:35 CEST] <maksbotan> browser can't start playing until first fragment of video is not written
[17:28:39 CEST] <DHE> the .m3u8 needs to know the duration of each segment, so it needs to finish a .ts before it can write an .m3u8 referencing it
[17:28:48 CEST] <maksbotan> DHE: ah so...
[17:28:53 CEST] <maksbotan> what does this duration depend on?
[17:29:09 CEST] <maksbotan> i've thought it is set by -hls_time, no?
[17:29:22 CEST] <DHE> yes, but the keyframe interval in the codec will dictate the actual duration
[17:30:05 CEST] <maksbotan> okay
[17:30:18 CEST] <maksbotan> i'm just looking for a way to start HLS streams on demand
[17:30:28 CEST] <maksbotan> seems like i have to live with this delay
[17:31:15 CEST] <DHE> also players usually start a few items back from the end of the stream. so if you join a live stream already in progress you're likely going to be 10-20 seconds behind "live"
[17:31:28 CEST] <DHE> this is part of why twitch runs so far behind real-time
[17:31:54 CEST] <maksbotan> i guess this is not a big issue for my case
[17:31:59 CEST] <maksbotan> but thanks
[17:36:14 CEST] <sedulous> How good is the internal AAC encoder in VBR mode nowadays?
[17:37:04 CEST] <JEEB> sedulous: good enough that soon'ish the experimental flag should be removed
[17:37:27 CEST] <JEEB> after which it will probably get more testing by some really funky people to dissect possible issues
[17:38:26 CEST] <DHE> is there a way to make ffmpeg (cmdline tool) either exit or reinitialize if there are too many errors? I've had ffmpeg running on a live TV channel (multicast IPTV) and the audio at some point just turned into this loud buzzing. I'd like it to recover more gracefully
[17:38:59 CEST] <sedulous> JEEB: great, thanks
[17:39:15 CEST] <sedulous> i will use it. the result sounds fine to me subjectively
[17:41:22 CEST] <sedulous> -c:a aac -q:a 6   is reasonable?
[17:41:27 CEST] <JEEB> sedulous: I just listened to a talk by the guy improving it massively lately, cool guy and lots of cool stuff being worked on
[17:42:01 CEST] <luc4> JEEB: for the problem I explained to you before, do you think I could try to send an email to the mailing list and see if someone has any idea?
[17:53:12 CEST] <waressearcher2> I want to concatenate 2 files, I use "ffmpeg -f concat -i list.txt -c copy -y -f avi out.avi" but I want it started concatenation of first file from 20 second position, is there a way to tell ffmpeg to seek 20 second in the first concatenated file ?
[19:09:31 CEST] <oreWizard> coded_frame
[19:09:38 CEST] <oreWizard> is deprecated
[19:09:41 CEST] <oreWizard> can someone help me?
[19:25:53 CEST] <BtbN> your statement is correct, it's deprecated.
[19:50:20 CEST] <oreWizard> what should i do instead
[19:50:39 CEST] <BtbN> Instead of what?
[19:51:01 CEST] <oreWizard> of using coded_frame
[00:00:00 CEST] --- Tue Sep 22 2015