[Ffmpeg-devel-irc] ffmpeg.log.20170316

[burek]

[00:31:44 CET] <shincodex> $TMPDIR does not exist on -msvc build
[00:32:05 CET] <shincodex> so this happens
[00:32:06 CET] <shincodex> 1>  ./configure: line 4971: mktemp: command not found 1>  ./configure: line 4972: mktemp: command not found 1>  mkdir: cannot create directory `': No such file or directory 1>  ln: target `' is not a directory: No such file or directory 1>  rm: cannot lstat `': No such file or directory 1>  rm: cannot lstat `': No such file or directory
[00:32:21 CET] <shincodex> might be a way to redirect that to windows temp dirs
[00:32:30 CET] <shincodex> since no /tmp or whatever
[00:37:34 CET] <shincodex> 1>  CC	libavformat/rtpdec_ac3.o 1>cl : Command line warning D9025: overriding '/W0' with '/W4'
[00:37:38 CET] <shincodex> you scoundrels!
[00:37:47 CET] <shincodex> i need that to stop
[06:30:14 CET] <matkatmusic> Howdy.  slightly off-topic.  has anyone taken a look at https://github.com/cisco/openh264 or http://git.videolan.org/?p=x264.git;a=summary
[06:30:27 CET] <matkatmusic> they're open-source H.264 codecs
[07:21:40 CET] <DHE> yeah. both are supported by ffmpeg as external libraries
[09:25:39 CET] <matkatmusic> Does anyone know where in the API we can pass in jpegs or bitmaps and specify which frames and for how many frames they should be written into an mp4?
[09:32:03 CET] <matkatmusic> i know you can pass in sequential images and it'll make a vid depending on parameters passed in.
[09:32:46 CET] <matkatmusic> i guess i'm wondering how specific you can get.  I didn't see anything in the CLI docs about passing in specific frame numbers per file
[09:38:55 CET] <matkatmusic> I saw this: For creating a video from many images: ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi
[09:42:15 CET] <matkatmusic> i did see -itsoffset :
[09:42:20 CET] <matkatmusic> The offset is added to the timestamps of the input files. Specifying a positive offset means that the corresponding streams are delayed by the time duration specified in offset.
[09:44:49 CET] <matkatmusic> Is it possible to even declare specific frame numbers that the input should be placed at?  It seems like you can only specify a timestamp in HH:mm:ss:m format
[10:12:14 CET] <d-fens_> hi, i need to generate 64 images with the counter printed on it, like "1" written on 01.jpg ... "64" written on 64.jpg, can this be automated in ffmpeg?
[10:13:02 CET] <matkatmusic> d-fens_: https://ffmpeg.org/ffmpeg.html#toc-Video-and-Audio-file-format-conversion
[10:13:28 CET] <matkatmusic> section 6.3
[10:14:39 CET] <d-fens_> matkatmusic: what filter woul accept a variabe to print the numbers on it?
[10:15:09 CET] <matkatmusic> ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi is the example line
[10:15:43 CET] <matkatmusic> %03d means an integer 3 characters long.
[10:15:53 CET] <matkatmusic> so, 010, 004, 102, etc..
[10:16:01 CET] <matkatmusic> if you need two 0's, then use %02d
[10:16:12 CET] <matkatmusic> er, two characters in length
[10:16:46 CET] <matkatmusic> oh, you want to write it to the image itself
[10:16:51 CET] <ikevin> i think he need the number inside the image, not in the name
[10:16:52 CET] <matkatmusic> i have no idea how to do that
[10:18:13 CET] <d-fens_> yep i start with nothing (or a white jpg) and need X images with the number written on the output images
[10:19:17 CET] <d-fens_> maybe imagemagik is the better tool for that
[10:24:42 CET] <durandal_170> drawtext
[10:24:46 CET] <durandal_170> filter
[10:39:34 CET] <nyuszika7h> why does MediaInfo say my encoded file is variable frame rate? I've checked and the duration is the exact same and subtitles fit, so it seems like it's wrong
[10:40:04 CET] <nyuszika7h> ffmpeg -i 'C.mkv' -map 0:v:0 -vf 'crop=720:432:0:72,scale=1024:432' -aspect 2.35/1 -c:v libx264 -preset veryslow -tune film -crf 17 -map 0:a:0 -c:a libfdk_aac -vbr 4 -ac 2 -metadata:s:a:2 'title=Stereo' 'Come diventare grandi nonostante i genitori (encode 2).mkv' -y
[10:40:08 CET] <nyuszika7h> oops
[10:40:08 CET] <ZeroWalker> well, vfr should have the same duration, that's the point right?
[10:40:22 CET] <nyuszika7h> accidentally cut the file name but you get the idea
[10:40:25 CET] <nyuszika7h> this is what I used
[10:41:04 CET] <ZeroWalker> and the input file is cfr?
[10:41:06 CET] <nyuszika7h> yes
[10:41:18 CET] <nyuszika7h> 25 fps (MPEG-2 from PAL DVD in .mkv)
[10:41:33 CET] <ZeroWalker> then it's probably just not stating it's cfr
[10:41:54 CET] <ZeroWalker> think you can force it to do some with some x264 command
[10:42:02 CET] <ZeroWalker> think it's like -cfr or something
[10:48:49 CET] <d-fens_> got it: this did the trick basically   -loop 1 -i input.jpg   -vf "drawtext=fontfile=arial.ttf: text=%{n+1}: x=(w-tw)/2: y=h-(2*lh): fontcolor=white: box=
[10:48:50 CET] <d-fens_> 1: boxcolor=0x00000099" -t 1 out%d.jpg
[10:53:09 CET] <d-fens_> actually text=%{eif\:n+1\:d}
[11:02:50 CET] <Nacht> Goodmorning everyone. Does anyone know if HLS support AES on HEVC ? I can make an MP4 in HEVC and it plays perfectly, then make a non-AES HLS from it, works perfectly as well. But once I add AES I can't seem to play it. However, using the same settings with a 264 source does work.
[11:03:18 CET] <Nacht> Simply using: ffmpeg -i ../265_mp4/movie.mp4 -c copy -hls_time 6 -hls_key_info_file key_info  -hls_playlist_type vod -hls_segment_filename "segment_%d.ts" index.m3u8
[11:03:20 CET] <JEEB> the encryption should be the same with AVC and HEVC in HLS
[11:03:20 CET] <Nacht> Nothing fancy
[11:03:52 CET] <JEEB> so if HEVC doesn't work then either you have a player problem or there's some sort of weird bug with the HLS muxer
[11:04:02 CET] <Nacht> I'm still not sure if it might be the player, but I've tried VLC, Windows Edge and MX Player, neither like it
[11:13:42 CET] <ZeroWalker> how does set a lower frame size for opus?
[11:16:48 CET] <JEEB> Nacht: uhh, as far as I know Edge doesn't support HEVC
[11:16:58 CET] <JEEB> VLC can just be too old, try the nightlies for 3.0
[11:17:10 CET] <JEEB> also a recent build of mpv
[11:17:30 CET] <JEEB> mpv utilizes libavformat+libavcodec in the back
[11:17:40 CET] <JEEB> so if it fails then libavformat+libavcodec themselves fail at playing it :P
[11:19:31 CET] <Nacht> From what I've read, Edge supports HEVC along with Hardware decoding
[11:20:01 CET] <Nacht> Ill try those you suggested
[11:23:09 CET] <Nacht> Thanks JEEB, VLC 3.0 Nightly did the job. It's playing now.
[11:25:18 CET] <ZeroWalker> --framesize n      Maximum frame size in milliseconds, this one is available in opus encoder, but i don't know how to set this in ffmpeg (through code)
[11:53:09 CET] <furqa> helllo, how can i use trim filter in this command
[11:53:10 CET] <furqa> .\ffmpeg -i intro.mp4 -i 1.MOV -i outro.mp4 -filter_complex "[0:v:0] [1:v:0] [2:v:0] concat=n=3:v=1 [v]" -map "[v]"  output.mp4
[12:56:44 CET] <memeka> when transcoding a file, the result has wrong fps (like 12000 fps) ... i tried using -r parameter but it's ignored .... any help?
[12:57:15 CET] <dt9> Hello, I've a question regarding to decoding with use coded_width/coded_height size. Is it possible to do that? For example encoded movie has 1920x1088 (coded size) and 1920x1080 as output size. I need to get to last 8 lines of data
[13:19:07 CET] <Nacht> memeka: are you transcoding ?
[13:19:37 CET] <Nacht> Or to be more exact: What's the command you're using
[13:33:59 CET] <memeka> Nacht: yes
[13:34:46 CET] <memeka> Nacht: it works well with ffmpeg 3.0, but not with 3.2
[13:38:31 CET] <matkatmusic> !topic
[13:38:37 CET] <matkatmusic> hmmm
[13:38:45 CET] <matkatmusic> i thought that would give me the channel topic and the link to the log..
[13:49:06 CET] <Nacht> Topic is 'Welcome to the FFmpeg USER support channel | Development channel: #ffmpeg-devel | Bug reports: http://bit.ly/cqvkhs | Wiki: https://trac.ffmpeg.org/ | FFmpeg 3.2.4 is released | FFmpeg forum: http://ffmpeg.gusari.org/ | This channel is publicly logged'
[13:50:13 CET] <Nacht> memeka: does -vf fps=30 work ?
[13:53:03 CET] <memeka> Nacht: no :(
[13:54:50 CET] <memeka> Nacht: I am using custom encoder (patched ffmpeg), this is my command: "ffmpeg -vcodec h264 -i ./sintel_trailer-720p.mp4 -vcodec h264_v4l2m2m encoded.mp4" -- it worked with the same patches in ffmpeg 3.0, not anymore on 3.2 -- I get 12000 fps in resulting output, when i play the file it's like on fast-forward (the frames are there, encoded, but the fps is
[13:54:50 CET] <memeka> wrong) ....
[13:56:34 CET] <memeka> Nacht: the weird bit is that while encoding, it sees the right fps: http://pastebin.com/XxwQqhZt
[13:57:00 CET] <memeka> Stream #0:0(und): Video: h264 (h264_v4l2m2m) ([33][0][0][0] / 0x0021), yuv420p, 1280x720, q=2-31, 200 kb/s, 24 fps, 12288 tbn, 24 tbc (default)
[13:57:47 CET] <memeka> but after the encoding is done: Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720, 10939868 kb/s, 12288 fps, 12288 tbr, 12288 tbn, 24576 tbc (default)
[13:59:41 CET] <memeka> Nacht: if i write to .mkv => it's 1000 fps ... so i think the issue is at muxing
[14:01:56 CET] <furq> sounds like the encoder is using the timebase as the framerate
[14:02:43 CET] <furq> do you have a reason for using 3.2
[14:03:02 CET] <furq> if these are unofficial patches then all i can really suggest is downgrading until they make it upstream
[14:03:19 CET] <memeka> furq: it's required by other packages on my os :(
[14:03:35 CET] <furq> i don't see why that matters if you're building it yourself
[14:03:50 CET] <furq> just have a separate local copy
[14:04:33 CET] <memeka> furq: looks like i'll do that :(
[14:05:04 CET] <memeka> what's even more strange
[14:05:17 CET] <memeka> is that when outputting to mkv
[14:06:14 CET] <furq> mkv's timebase has to be expressible in nanoseconds
[14:06:21 CET] <memeka> mediainfo shows 1000fps (same thing to mp4 shows 12000fps); but ffmpeg -i encoded.mkv shows the correct 24 fps (ffpemg -i encoded.mp4 shows 12000fps like mediainfo)
[14:06:33 CET] <memeka> but mkv also plays like fast forward ...
[14:06:44 CET] <furq> fun
[14:07:00 CET] <memeka> so i am assuming the issue is at muxing ...
[14:07:01 CET] <furq> maybe different timestamps in the header and the stream itself
[14:07:19 CET] <furq> it sounds like an encoder issue
[14:07:33 CET] <memeka> i do get this:
[14:07:34 CET] <memeka> [mp4 @ 0x7f6aeb10] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
[14:07:41 CET] <furq> you can probably demux the h264 stream and remux it with the right framerate
[14:07:42 CET] <memeka> [mp4 @ 0x7f6aeb10] Encoder did not produce proper pts, making some up
[14:07:51 CET] <furq> yeah that might be an issue
[14:08:05 CET] <memeka> the same output i get on 3.0 tho`, and output is correct on 3.0 ....
[14:08:08 CET] <furq> like i say, you can rewrite the stream timestamps, but it's two extra steps
[14:08:20 CET] <furq> it sounds easier to just downgrade to 3.0
[14:08:43 CET] <memeka> furq: thanks .... not sure when V4L2 M2M will make it official into ffmpeg :(
[14:08:52 CET] <furq> i've never even heard of it before
[14:08:54 CET] <memeka> looks like i;ll downgrade :(
[14:09:04 CET] <furq> i'm guessing it's something similar to h264_omx on rpi
[14:09:38 CET] <memeka> yeah probably, just that more VPUs are looking to use V4L2 now ...
[14:09:56 CET] <memeka> so hopefully it will push official development :)
[14:10:04 CET] <memeka> thanks
[14:21:39 CET] <piem> howdy! how should I cross-compile a package against ffmpeg to windows? builds from zeranoe do not seem to include the pkg-config .pc files
[14:26:53 CET] <matkatmusic> found the logs..  http://lists.ffmpeg.org/pipermail/ffmpeg-devel-irc/2017-March/004171.html
[14:51:51 CET] <temp> /usr/local/lib/gcc/x86_64-pc-linux-gnu/7.0.1/../../../../x86_64-pc-linux-gnu/bin/ld: /usr/local/lib/gcc/x86_64-pc-linux-gnu/7.0.1/../../../../lib64/libz.a(libz_a-inffast.o): relocation R_X86_64_32S against `.rodata.str1.1' can not be used when making a shared object; recompile with -fPIC
[14:51:51 CET] <temp> /usr/local/lib/gcc/x86_64-pc-linux-gnu/7.0.1/../../../../x86_64-pc-linux-gnu/bin/ld: final link failed: Nonrepresentable section on output
[14:51:52 CET] <temp> collect2: error: ld returned 1 exit status
[14:52:47 CET] <temp> what's reason for this error
[15:42:37 CET] <Rathann> temp: your static libc wasn't compiled with -fPIC, so you cannot use it for linking
[15:42:52 CET] <Rathann> for linking shared objects that is
[16:08:05 CET] <feliwir> how is huffman implemented in libavcodec? I keep seeing some VLC struct, but nowhere is an explanation what exactly that VLC struct is supposed to represent
[16:08:14 CET] <feliwir> https://github.com/FFmpeg/FFmpeg/blob/967feea5ebb744dce97ab327d33502b43fca0c7f/libavcodec/vp6.c#L265
[16:53:06 CET] <mcjack> Hi everybody, I try writing a mp4 file with the libs, and I get "track 1: codec frame size not set" as debug output. What does this relate to, and what did I do wrong?
[16:53:20 CET] <mcjack> here is the full av_dump: http://pastebin.com/6PkJAZ1B
[16:54:03 CET] <mcjack> and here is the source: https://github.com/filmstro/filmstro_ffmpeg/blob/master/modules/filmstro_ffmpeg/filmstro_ffmpeg_FFmpegVideoWriter.cpp#L152
[16:54:31 CET] <mcjack> it is a wrapper to use ffmpeg in an audio centric framework named JUCE&
[17:10:07 CET] <mcjack> or more precisely, which codec, the audio codec, video codec or container codec?
[17:46:26 CET] <matkatmusic> mcjack: i'm sure you've stared at this a lot, but are you forgetting to set one of these properties? https://ffmpeg.org/doxygen/trunk/structAVCodec.html
[17:49:03 CET] <mcjack> Hi matkatmusic, thanks& as I understand it, the AVCodec is the encoder, that you get using avcodec_find_encoder (codec_id); you are not supposed to alter that, instead you set your settings in the AVCodecContext
[17:49:34 CET] <mcjack> I did set many, but I cannot claim to know, if I did all needed ones&
[17:50:04 CET] <mcjack> The thing is, the debug doesn't tell me, which codec might be missing a setting
[17:50:18 CET] <matkatmusic> https://ffmpeg.org/doxygen/trunk/structAVCodecContext.html#aec57f0d859a6df8b479cd93ca3a44a33 says it sets the audio frame size
[17:50:43 CET] <matkatmusic> "Number of samples per channel in an audio frame."
[17:51:03 CET] <mcjack> I'll try that&
[17:51:56 CET] <mcjack> the thing with that is, the docs you linked say, when encoding avcodec_open2() will set that&
[17:52:14 CET] <matkatmusic> maybe it's a public parameter, and you can set it?
[17:53:44 CET] <mcjack> I definitely can, because there is no enforcement from the compiler (like encapsulation in C++), but the question is, will the code then do what I expect it to do?
[17:53:49 CET] <matkatmusic> https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/avcodec.h#L2497
[17:54:58 CET] <matkatmusic> seems public, since it's inside 'typedef struct AVCodecContext { }'
[17:55:16 CET] <matkatmusic> it says on line 1707 of that file:
[17:55:22 CET] <matkatmusic> You can use AVOptions (av_opt* / av_set/get*()) to access these fields from user applications.
[17:55:32 CET] <mcjack> My understanding is, that when opening the codec with avcodec_open2() the lib will chose a reasonable framesize and set it&
[17:56:02 CET] <mcjack> I don't even know, if it is the audio codec, that is complaining
[17:56:22 CET] <mcjack> but you are right, probably
[17:56:37 CET] <mcjack> as the docs say it relates to samples
[17:58:11 CET] <mcjack> the other thing is, none of the examples set this, e.g. http://ffmpeg.org/doxygen/trunk/muxing_8c-example.html#a29
[17:58:36 CET] <matkatmusic> https://ffmpeg.org/doxygen/trunk/group__opt__set__funcs.html#details
[17:59:49 CET] <matkatmusic> I'd see if I could find that error message in the source code for ffmpeg
[18:00:05 CET] <mcjack> same thought, grep is running
[18:00:52 CET] <matkatmusic> it's a shame searching for it on github doesn't find exact strings
[18:02:35 CET] <matkatmusic> https://github.com/FFmpeg/FFmpeg/search?utf8=%E2%9C%93&q=%22size+not+set%22+in%3Afile+&type=Code
[18:03:39 CET] <matkatmusic> https://github.com/FFmpeg/FFmpeg/blob/b05d8e7184f2f61c1e6261b79ef20abd48b539cd/libavformat/mpegtsenc.c#L969
[18:04:27 CET] <matkatmusic> so maybe you need to call av_set_audio_frame_duration2 ?
[18:04:31 CET] <llogan> easy to find using "git grep" if you have a copy of the git repository
[18:04:57 CET] <matkatmusic> you can search for quoted strings if you add 'in:file' after your string in quotes
[18:05:13 CET] <matkatmusic> aka "frame size not set" in:file
[18:08:12 CET] <piem> any idea of a project using ffmpeg and cross-compiling for windows? i'm missing the pkg-config.pc files in the binaries from ffmpeg.zeranoe.com
[18:08:39 CET] <matkatmusic> mcjack: see those links
[18:09:39 CET] <mcjack> yes, but I think it's: https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/movenc.c#L5778
[18:10:14 CET] <matkatmusic> ah
[18:10:38 CET] <matkatmusic> yes, you didn't set your bits per sample in your code betwen line 219 and 225
[18:13:39 CET] <mcjack> didn't change anything :-(
[18:14:43 CET] <matkatmusic> what about...  https://ffmpeg.org/doxygen/trunk/structAVCodecContext.html#aec57f0d859a6df8b479cd93ca3a44a33
[18:14:54 CET] <matkatmusic> "May be 0 when the codec has AV_CODEC_CAP_VARIABLE_FRAME_SIZE set, then the frame size is not restricted."
[18:18:18 CET] <matkatmusic> mcjack: what about setting it to that?
[18:18:43 CET] <mcjack> to 0 or to an assumed value?
[18:18:56 CET] <matkatmusic> to that AV_CODEC thing
[18:19:30 CET] <matkatmusic> from that link you posted, it seems to want the audio format in Variable Bit Rate, and you're supplying a static bit rate?
[18:20:08 CET] <matkatmusic> (link to your code)
[18:20:40 CET] <mcjack> uhm, one thing is the input, that is a static value, because I'm feeding it raw floats. The encoder will do whatever is best for the selected Codec, as I understand it&
[18:21:56 CET] <matkatmusic> hmm, well I'd add some breakpoints to line 5776 of movenc.c, and then probe st->codecpar and see what properties are available to you, and what they're set to.
[18:22:29 CET] <matkatmusic> if (!st->codecpar->frame_size && !av_get_bits_per_sample(st->codecpar->codec_id)) it wants those two things set to something other than 0
[18:22:33 CET] <matkatmusic> afaik
[18:22:49 CET] <mcjack> right
[18:23:10 CET] <mcjack> no, it want's at least one of the two different from 0
[18:23:55 CET] <matkatmusic> you sure about that?
[18:24:30 CET] <mcjack> argn, now I'm confused. slowly: the error will appear, if frame_size == 0 and bits per samples == 0
[18:24:47 CET] <matkatmusic> you could rewrite it: if( st->codecpar->framesize == 0 && av_get_bits_per_sample(st->codecpar->codec_id) == 0 )
[18:24:50 CET] <mcjack> so it won't if either of it is set different from 0
[18:25:08 CET] <matkatmusic> 'either' would be using ||
[18:25:14 CET] <matkatmusic> it's saying if both of them are 0
[18:25:23 CET] <matkatmusic> then, assume compressed audio
[18:25:28 CET] <mcjack> yes, but I am reversing the outcome, so de morgans rules apply
[18:26:34 CET] <mcjack> if both apply (&&) then the error is shown vs. if one is not (||) the error won't be shown...
[18:26:59 CET] <matkatmusic> ok.  Well, what does your stream->codecpar look like after line 225 in your code?  is frame_size or codec_id 0?
[18:28:05 CET] <matkatmusic> maybe add a breakpoint on 227 and inspect stream->codecpar?
[18:28:21 CET] <mcjack> it's not !codec_id but !av_get_bits_per_sample(st->codecpar->codec_id)
[18:29:28 CET] <matkatmusic> hmmm, so maybe you're supplying a codec_id that is set up for VBR?
[18:30:34 CET] <matkatmusic> FFmpegVideoWriter::setAudioCodec() what kind of codec are you supplying?
[18:31:12 CET] <mcjack> audio is aac
[18:31:30 CET] <mcjack> I basicvally copy a video replacing the audio samples
[18:31:37 CET] <matkatmusic> aac is VBR
[18:32:22 CET] <matkatmusic> what if you choose a different audio codec.  like, one for non-VBR.   I don't know much about the different codecs, so ignore me if my statement is ill-informed
[18:33:04 CET] <mcjack> well, that's a sane test& I could do that&
[18:33:32 CET] <matkatmusic> i guess pick the audio codec that matches what JUCE's AudioSource spits out?
[18:33:40 CET] <matkatmusic> where are the codec types listed?
[18:35:30 CET] <mcjack> https://ffmpeg.org/doxygen/trunk/group__lavc__core.html#gaadca229ad2c20e060a14fec08a5cc7ce
[18:35:40 CET] <mcjack> quite a few to test
[18:36:21 CET] <matkatmusic> right right
[18:36:40 CET] <matkatmusic> I'm gonna guess that ID_PCM_S16LE is Signed 16bit Little Endian
[18:38:32 CET] <matkatmusic> mcjack: can we see how you're using your FFmpegVideoWriter class?
[18:39:34 CET] <mcjack> sure, it's all in the demo: https://github.com/filmstro/filmstro_ffmpeg/blob/master/examples/VideoPlayer/Source/OSDComponent.h#L59
[18:40:06 CET] <mcjack> simply remove the #if 0 and corresponding #endif and a save button will appear
[18:40:43 CET] <mcjack> then you load a video and click save, it asks for a filename and starts writing&
[18:41:04 CET] <mcjack> but the files were not ok, unfortunately, nothing to be heard etc...
[18:41:20 CET] <mcjack> so many rivers to cross
[18:41:45 CET] <matkatmusic> I hear ya.  hopefully i'm helping a little bit...
[18:43:15 CET] <alexpigment> mcjack, are you trying to write PCM audio to an MP4 file?
[18:43:38 CET] <matkatmusic> mcjack: just looking at line 190 in that file, trying to figure out what type of audio format your AudioSourceChannelInfo is in
[18:43:54 CET] <matkatmusic> file:///Users/Shevy/Dropbox/CharlesShared/JUCE-4_2_4/doxygen/doc/juce__AudioSampleBuffer_8h.html#a97a2916214efbd76c6b870f9c48efba0
[18:43:57 CET] <matkatmusic> er
[18:44:14 CET] <matkatmusic> it says it uses a A multi-channel buffer of 32-bit floating point audio samples.
[18:44:29 CET] <matkatmusic> so, you probably want the codec AV_CODEC_ID_PCM_F32LE
[18:45:25 CET] <alexpigment> just out of curiosity, is this problem that you're feeding it 32-bit audio and you're getting an error when trying to encode AAC?
[18:45:51 CET] <mcjack> yes, is this not supported?
[18:46:16 CET] <alexpigment> if so, why not try something like -c:a aac -sample_fmt s16 -ar 44100 -b:a 160000
[18:46:35 CET] <alexpigment> i don't know for sure that it's not supported, but i have never seen 32-bit aac ever
[18:46:49 CET] <alexpigment> i would be surprised if it *did* support it
[18:46:50 CET] <matkatmusic> alexpigment: he's writing an API to interface ffmpeg with the JUCE audio framework.
[18:47:07 CET] <mcjack> thanks for the suggestion. However I use the library, not the commandline&
[18:47:20 CET] <alexpigment> yeah i get that, it just seems like the error is that you need to convert to 16-bit
[18:47:24 CET] <mcjack> but it's a good idea to double check the supported formats
[18:47:56 CET] <matkatmusic> mcjack: does juce have a means to go from raw samples to aac data?
[18:48:06 CET] <IntruderSRB> anyone have any idea if MBAFF stream can be encrypted using MPEG-CENC in order to be DRM protected?
[18:48:16 CET] <matkatmusic> I'm looking in the API, and haven't seen anything about it.  found some forum posts where Jules said he didn't plan on implementing it
[18:48:41 CET] <IntruderSRB> my encryptor has no issue with progressive videos DRM protection (using CENC scheme) but when I provide MBAFF cenc protected - all players throw decode error
[18:49:01 CET] <mcjack> there is an AudioFormatWriter which can produce AAC files, but that's here off topic, and it will not work for writing in a video file
[18:59:06 CET] <alexpigment> IntruderSRB: are you trying to stream interlaced content?
[19:01:03 CET] <alexpigment> i just haven't seen an interlaced stream ever, and i don't know how web players handle it. having said that, I trust that you get no decode errors when there's no DRM, so who knows what's going on
[19:03:40 CET] <IntruderSRB> exactly ...
[19:03:54 CET] <IntruderSRB> works perfectly fine in browser (native player or bitmovin/dash-if)
[19:04:04 CET] <IntruderSRB> until I turn on encryption ('cenc')
[19:04:15 CET] <IntruderSRB> than all hell break loose and I get decode_error instantly
[19:05:21 CET] <IntruderSRB> I guess it can be deinterlaced before encryption but the CPU cost would probably be high ...
[19:21:03 CET] <mcjack> alexpigment and matkatmusic: thanks for trying with me, I have to leave for today&
[19:38:32 CET] <alexpigment> IntruderSRB: yeah deinterlacing is something i used to do a lot, but I leave everything as MBAFF these days because a) it takes a lot less time and b) it keeps blu-ray compatibility.
[19:38:59 CET] <alexpigment> good luck on figuring that one out - i honestly have no idea why encryption would break because of mbaff
[20:05:41 CET] <JEEB> encryption shouldn't unless it somehow handles stuff inside samples in a special way
[20:05:52 CET] <JEEB> generally just encrypting contents of samples would make sense'ish
[20:15:06 CET] <Mista-D> is there a global video filter command without ,split[1][2][3] ?
[20:16:52 CET] <thebombzen> Mista-D: what are you trying to do
[20:16:59 CET] <thebombzen> split[1][2][3] won't even work
[20:17:35 CET] <Mista-D> encode to between 5 and 15 ouputs in one command while deinterlacing source.
[20:18:18 CET] <thebombzen> ah. afaik using the same filterchain with several outputs is hard, but something you CAN do is output something lossless and pipe it to the encoding command
[20:18:57 CET] <thebombzen> like ffmpeg -i input -vf filters -c ffv1 -f nut - | ffmpeg -f nut - output1.mkv output2.mkv output3.mkv
[20:18:59 CET] <thebombzen> etc.
[20:22:16 CET] <thebombzen> I think you can do it it one command too though with: ffmpeg -i input -lavfi '[0:v] yadif [v]' -map '[v]' out1.mkv -map '[v]' out2.mkv -map '[v]' out3.mkv
[20:22:20 CET] <thebombzen> but this might not work
[20:22:24 CET] <thebombzen> you could try i tthough
[20:22:47 CET] <matkatmusic> does anyone here use the library instead of the CLI to do their conversion via an app of their own?
[20:22:51 CET] <thebombzen> Mista-D: I'm not quite sure if -lavfi behaves globally or per-output, but I'm sure the docs would know
[20:23:20 CET] <thebombzen> matkatmusic: someone definitely does (not me) so I suggest you ask the question you're actually trying to ask
[20:23:26 CET] <Mista-D> thebombzen: will try that, thanks
[20:24:42 CET] <thebombzen> matkatmusic: because then someone who does use libav* could potentially answer your question, which nobody knows yet
[20:42:12 CET] <matkatmusic> i'm just trying to continue where mcjack left off with getting his API to work with ffmpeg
[20:44:33 CET] <matkatmusic> interacting with the decoder/encoder via code as opposed to the CLI tool
[20:44:56 CET] <JEEB> see the stuff under docs/examples
[20:45:20 CET] <JEEB> and then there's stuff like Handbrake, VLC and mpv that utilize libav*
[20:45:27 CET] <johoja> Hi, - I'm trying to compile ffmpeg 3.2.4 - with --enable-librtmp - on ubuntu 14.04 , and i have librtmp-dev installed, but i get an error ERROR: librtmp not found using pkg-config
[20:45:41 CET] <JEEB> see config.log
[20:45:47 CET] <JEEB> that should tell you what exactly failed at the end of it
[20:45:58 CET] <JEEB> also do note that recent FFmpeg doesn't require librtmp for rtmp
[20:46:03 CET] <furq> is there any reason to use librtmp now
[20:46:03 CET] <JEEB> it has its own implementation
[20:46:09 CET] <johoja> this is what I'm getting :
[20:46:10 CET] <johoja> END /tmp/ffconf.HIc9WuDZ.c gcc -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -I/root/ffmpeg_build/include -std=c99 -fomit-frame-pointer -pthread -I/usr/include/freetype2 -I/usr/include/fribidi -I/usr/include/freetype2 -I/usr/include/opus -I/usr/include/p11-kit-1 -R/usr/lib/x86_64-linux-gnu -c -o /tmp/ffconf.ygSadPNx.o /tmp/ffconf.HIc9WuDZ.c gcc: error: unrecognized command line
[20:46:25 CET] <JEEB> johoja: do not post the content here, just pastebin your config.log and link it here
[20:46:31 CET] <johoja> one second - sorr
[20:46:42 CET] <JEEB> furq: no real reason to use librtmp unless the lavf one has a boog
[20:46:48 CET] <johoja> http://paste.ubuntu.com/24190653/
[20:47:01 CET] <johoja> yeah - lavf is not working for a stream, i want to see if its lavf, or if it will work in librtm
[20:47:02 CET] <furq> i assume there's some justification for not getting rid of --enable-librtmp
[20:47:03 CET] <johoja> yeah - lavf is not working for a stream, i want to see if its lavf, or if it will work in librtmp
[20:47:15 CET] <johoja> http://paste.ubuntu.com/24190653/
[20:47:22 CET] <johoja> Sorry for the double post there.
[20:47:24 CET] <JEEB> > unrecognized command line option '-R'
[20:47:30 CET] <JEEB> check if that's in the pkg-config file
[20:47:38 CET] <johoja> Where would that be ?
[20:47:39 CET] <JEEB> or what the flying F is going on
[20:49:22 CET] <JEEB> johoja: you should be able to check with `pkg-config --cflags --libs librtmp`
[20:49:32 CET] <JEEB> if it comes out of there then the pkg-config file is just busted
[20:50:00 CET] <johoja> I don't see any -R in the output from pkg-config
[20:51:06 CET] <JEEB> ok, then I officially have nfi where that shit's coming
[20:51:23 CET] <johoja> :( - maybe the configure script ?
[20:51:24 CET] <JEEB> because there's no -R in the configure file
[20:51:55 CET] <JEEB> and something specifically is setting it as it even has a parameter
[20:51:56 CET] <JEEB> -R/usr/lib/x86_64-linux-gnu
[20:52:23 CET] <johoja> ya its weird.
[23:55:26 CET] <IntruderSRB> hey guys, I'm trying to extract h264 nalunits into h264 file (bytearray) from fragment fmp4
[23:55:33 CET] <IntruderSRB> is there a way to pass "header" mp4 as an argument
[23:55:39 CET] <IntruderSRB> ffmpeg -i fileSequence1.mp4 -an -vcodec h264 fileSequence1.h264
[23:55:48 CET] <IntruderSRB> this is what I managed to find out atm
[23:55:58 CET] <IntruderSRB> got
[23:56:01 CET] <IntruderSRB> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f8dfe000400] could not find corresponding trex
[23:56:01 CET] <IntruderSRB> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f8dfe000400] error reading header
[23:56:13 CET] <IntruderSRB> which is expected as the header alone is in fileSequence0.mp4
[23:56:56 CET] <JEEB> I'd just concatenate them and pass to ffmpeg through stdin
[23:57:19 CET] <IntruderSRB> worth a try ...
[23:57:27 CET] <JEEB> cat initSegment.mp4 actualSegment1.mp4 | ffmpeg -i -c copy out.264
[23:57:48 CET] <JEEB> that will copy the NAL units, -c:v h264 would be transcoding
[23:58:18 CET] <iive> JEEB: you can concat .mp4? thats quite a news for me.
[23:58:26 CET] <JEEB> fragmented isobmff yes
[23:58:38 CET] <IntruderSRB> it worked...
[23:58:47 CET] <IntruderSRB> sometimes the simplest solutions are the best :D
[23:59:00 CET] <iive> :D
[23:59:10 CET] <JEEB> the mov demuxer probably wouldn't like your usualy non-fragmented isobmff like that though
[23:59:21 CET] <JEEB> but E_HAVE_NOT_TRIED
[00:00:00 CET] --- Fri Mar 17 2017