[Ffmpeg-devel-irc] ffmpeg.log.20180119

[burek]

[00:11:16 CET] <b0bby__> kepstin: You still there?
[00:50:54 CET] <b0bby__> How does this filter work [0:0] [0:1] [1:0] [1:1] [2:0] [2:1] concat=n=3:v=1:a=1 [v] [a]
[00:51:05 CET] <b0bby__> ?
[01:12:33 CET] <b0bby__> hello
[01:12:40 CET] <b0bby__> is anyone online?
[01:13:57 CET] <dakar> i am
[01:15:54 CET] <b0bby__> dakar: cool
[01:16:37 CET] <b0bby__> dakar: do you know how to use the concat filter on three videos(one with no audio stream)?
[01:16:50 CET] <dakar> i dont know jack tbh, i'm here to ask some very basic question
[01:19:08 CET] <DHE> b0bby__: you'll either need to make a video with no audio or built a fake audio stream for the ones without audio
[01:19:47 CET] <b0bby__> DHE: cool to know that there is a way. How would I go about this
[01:20:13 CET] <b0bby__> here is what I have so far:
[01:20:15 CET] <b0bby__> ffmpeg -i 1.webm -i 2.webm -i 3.webm -c:v libx264 -c:a aac -filter_complex '[0:0] [0:1] [1:0] [1:1] [2:0] [2:1] concat=n=3:v=1:a=1 [v] [a]' -map '[v]' -map '[a]' out.mp4
[01:21:21 CET] <b0bby__> this script is meant for automation so ffmpeg will need to detect which videos don't have audio
[01:22:32 CET] <b0bby__> you still there DHE
[01:22:35 CET] <b0bby__> ?
[01:23:05 CET] <DHE> sorry, I don't think that can be done in pure ffmpeg. you'd need some app to build a commandline. ffmpeg and/or ffprobe can help
[01:25:27 CET] <b0bby__> DHE: ok
[01:25:44 CET] <b0bby__> DHE: lets say I know which video does not have audio
[01:26:02 CET] <b0bby__> DHE: in my above command how would I add the audio
[01:28:19 CET] <DHE> you'll need a file containing a short burst of silence (like a 0.01 second WAV file) to insert as the audio where it's missing
[01:29:06 CET] <b0bby__> DHE: go on
[01:31:45 CET] <DHE> that's it for explanation. put that into the filter where you need it
[01:32:38 CET] <b0bby__> Got it
[01:33:08 CET] <b0bby__> DHE: trying ffmpeg -i 1.webm -i 2.webm -i 3.webm -c:v libx264 -c:a aac -filter_complex '[0:0] [0:1] [1:0] [1:1] [2:0] [2:1] concat=n=3:v=1:a=1 [v] [a]' -map '[v]' -map '[a]' out.mp4
[01:33:19 CET] <b0bby__> with 3 videos with audio streams
[01:33:49 CET] <b0bby__> nvm
[01:34:03 CET] <b0bby__> forgot to apply my resize filters
[01:35:57 CET] <b0bby__> DHE: sorry to keep bothering you but I'm having some more trouble
[01:36:37 CET] <b0bby__> I need to combine a filter the resizes the videos like : scale=w=1280:h=720:force_original_aspect_ratio=1,pad=1280:720:(ow-iw)/2:(oh-ih)/2
[01:36:43 CET] <b0bby__> to my original command
[01:36:49 CET] <b0bby__> how would I do that
[01:39:23 CET] <mix123> Hi, after building, can i delete the ffmpeg_sources folder, or is that needed?
[01:41:23 CET] <DHE> mix123: if you have your binaries installed and they work, you can delete the sources. the risk is if you want to add a feature later it'll be a bit more of a hassle
[01:42:48 CET] <mix123> DHE: ok, thanks
[01:49:25 CET] <b0bby__> DHE: what do semicolons mean vs commas
[01:51:24 CET] <DHE> b0bby__: semicolons can be used for arbitrary connections with labels. commas are good for making a simple chain where labels are not necessary
[01:51:44 CET] <b0bby__> DHE: got it
[01:54:17 CET] <b0bby__> DHE: can you check my command https://pastebin.com/rmXrqbkb
[01:54:28 CET] <b0bby__> I don't know whats wrong with it
[01:54:40 CET] <b0bby__> it gives No such filter: ' concat=n=3:v=1:a=1 [v] [a]'
[01:55:56 CET] <b0bby__> found the dangling quote but now I get Stream specifier ':1' in filtergraph description [0:1]scale=w=1280:h=720:force_original_aspect_ratio=1,pad=1280:720:(ow-iw)/2:(oh-ih)/2;[1:1]scale=w=1280:h=720:force_original_aspect_ratio=1,pad=1280:720:(ow-iw)/2:(oh-ih)/2;[2:1]scale=w=1280:h=720:force_original_aspect_ratio=1,pad=1280:720:(ow-iw)/2:(oh-ih)/2;[0:0] [0:1] [1:0] [1:1] [2:0] [2:1] concat=n=3:v=1:a=1 [v] [a] matches no streams.
[01:58:28 CET] <b0bby__> nvm
[01:59:58 CET] <b0bby__> DHE: what does SAR mean?
[02:14:04 CET] <DHE> b0bby__: sample aspect ratio. the radio of a pixel. 1:1 means square pixels
[02:14:41 CET] <b0bby__> DHE: thanks, fixed it by setting the dar
[02:14:55 CET] <b0bby__> DHE: got the script working
[02:15:06 CET] <b0bby__> DHE: thanks for all the help :)
[03:52:46 CET] <siix> hey folks - using ffmpeg 3.4 on arch - trying to get rid of black bars around a video i used cropdetect.   ffmpeg -s 90 -i filename.mp4 -vframes 10 -vf cropdetect -f null -    it gave the error "Option video_size not found".  help please :)
[03:54:56 CET] <furq> you probably meant -ss 90
[03:56:05 CET] <siix> yeah sorry -ss
[07:02:42 CET] <M6HZ> Hello, I encounter an issue since several days when playing http audio streams with ffmpeg based programs. After many hours of playing (more than 24 hours), the audio begins to jolt for few hours before letting place to silence ("rdft" flat with ffplay).
[07:03:41 CET] <M6HZ> This happens at least with mpv, ffplay, mplayer
[07:06:44 CET] <M6HZ> I don't know if it's a problem related to ISP, which would be weird because the stream are carried over TCP, or if it's a policy from the streamers, but this problem happens for at least three different Media servers owned by different organizations.
[07:07:25 CET] <M6HZ> Or maybe is there something to dig into about ffmpeg ?
[11:11:25 CET] <diverdude> Hi, is it possible to share a frame from a video read by ffmpeg with a linear algebra library like eigen or any other data container?
[11:48:32 CET] <JEEB> diverdude: if you are using the API you would get the decoded data from the AVFrames returned by the decoding API and you could do whatever you wanted with it
[13:20:14 CET] <mifritscher1> hi
[13:20:59 CET] <mifritscher1> intel quick sync is not working for me on Windows 64 bit on Skylake:  "40 [h264_qsv @ 00000000219bbde0] Initialized an internal MFX session using hardware accelerated implementation", and then "Could not open codec: Internal bug, should not have happened"
[13:21:51 CET] <mifritscher1> both for mpeg2_qsv and h264_qsv
[13:25:28 CET] <JEEB> the QSV stuff is notoriously unstable it seems
[13:25:48 CET] <JEEB> which generally has been the intel stuff as oppsed to the FFmpeg stuff, although you should always test with the latest
[13:32:05 CET] <mifritscher1> it is the ffmpeg 3.4.1 + most current driver I get for my dell laptop (middle of 2017)
[13:33:28 CET] <JEEB> try a build based on the current git master's HEAD, but I do not recall if there were changes in that part of things since 3.4.1
[13:34:00 CET] <JEEB> but in unfortunately many cases issues with QSV end up with "well, tough luck. the driver's doing something derpy"
[13:35:47 CET] <mifritscher1> ok, I'll try it
[14:55:10 CET] <Fyr> guys, when muxing an audio track from an MKV to an MP4, FFMPEG adds:
[14:55:11 CET] <Fyr> Stream #0:1(eng): Data: bin_data (text / 0x74786574)    Metadata:       handler_name    : SubtitleHandler
[14:55:36 CET] <Fyr> MediaInfo cannot retrieve any information from the MP4 file.
[14:55:43 CET] <therage3> what command did you use to mux the mkv to mp4?
[14:56:08 CET] <Fyr> ffmpeg -i "in.mkv" -map 0:2 -c copy -vn -sn out.mp4
[14:57:35 CET] <Fyr> when muxing, FFMPEG shows one stream, the output MP4 file contains two streams, 0:0 - the audio and 0:1 - the unknown shit.
[14:57:48 CET] <therage3> what c_14 said
[14:59:40 CET] <Fyr> chcp 65001
[14:59:56 CET] <Fyr> sorry, wrong window.
[15:04:35 CET] <Fyr> c_14,
[15:04:35 CET] <Fyr> https://pastebin.com/4kLXeXk8
[15:04:35 CET] <Fyr> https://pastebin.com/Qa1DhwVd
[15:07:11 CET] <ritsuka> Fyr: it's the chapters track
[15:07:29 CET] <c_14> I'd assume so too
[15:07:37 CET] <Fyr> 0:1 (eng) bin_data?
[15:07:38 CET] <c_14> you can see if -map_chapters -1 gets rid of it
[15:08:40 CET] <Fyr> ffprobe says: "Unsupported codec id."
[15:09:16 CET] <Fyr> yes, -map_chapters got rid of it.
[15:09:18 CET] <Fyr> thanks
[15:09:42 CET] <Fyr> though, it's strange that FFPROBE can't handle it.
[15:18:23 CET] <CaimAstraea> Hello :)  If anyone can chime in not sure if this is possible :D  I'm using ffmpeg on linux to capture screen from a chrome window. Like so ffmpeg -f x11grab -s 1024x768 -r 30 -i $DISPLAY -f alsa -ac 2 -i pulse -acodec libmp3lame -ab 128k -vcodec libx264 myvideo.mpg
[15:19:12 CET] <CaimAstraea> Was wondering if it's possible to launch multiple chrome windows on different Xvfb screens ? and record from each one but than wouldn't the output have the same sound source ?
[15:19:57 CET] <CaimAstraea> So I would hear the sound from both chrome windows.. is it possible to specify the sound "source" for each window ? My instinct says no
[15:21:19 CET] <CaimAstraea> In the guide here https://trac.ffmpeg.org/wiki/Capture/Desktop though i saw somethng like i hw:0 so maybe that's a start no idea what that means since there is only 1 hw source for the sound even if it's virtual
[15:30:57 CET] <CaimAstraea> i ran pactl list sources and it says i only have source #0
[15:31:03 CET] <CaimAstraea> auto_null monitor
[15:41:33 CET] <c_14> you should be able to use pulse to record the monitor output for every individual sound source
[15:41:40 CET] <c_14> that only works if chrome opens a sound source per window though
[15:46:52 CET] <c_14> or sound output rather
[15:47:12 CET] <c_14> whatever the terminology is for this shi^wwonderful stuff
[15:49:13 CET] <klaxa> systemd-ffmpegd when?
[15:50:06 CET] Action: DHE <img src="killicon-backstab.png"> systemd
[15:56:13 CET] <CaimAstraea> Ah yes i found  pacmd I need to create some new audio sink and something called pavucontrol will try it out when I get home can't wait https://askubuntu.com/questions/60837/record-a-programs-output-with-pulseaudio :)  worst case scenario i wreck the VM but can easily spin another one
[16:26:38 CET] <lyncher> hi all. does anyone knows if livav* has a generic list/queue?
[16:29:57 CET] <DHE> there is a fifo type. see libavutil/fifo.h
[16:30:16 CET] <DHE> there's also a thread-safe queue system built on it
[16:30:35 CET] <lyncher> thanks!
[16:56:26 CET] <mifritscher> hi
[17:19:06 CET] <norayr> how does ffmpeg support gifs? i use gentoo, there's no gif USE flag, but ffmpeg -format gives me 'gif'.
[17:19:17 CET] <norayr> i guess gif is supported in one of the dependencies. which?
[17:19:32 CET] <c_14> there's an internal demuxer/decoder
[17:19:49 CET] <c_14> as with most codecs/formats that ffmpeg supports
[17:20:00 CET] <norayr> oh.
[17:20:09 CET] <norayr> and is it a configure option?
[17:20:18 CET] <c_14> I mean, you _can_ disable it
[17:20:19 CET] <norayr> i mean, may it be gentoo folks disabled it?
[17:20:24 CET] <c_14> nah
[17:20:31 CET] <c_14> don't think they enable/disable individual codecs
[17:20:44 CET] <c_14> why?
[17:22:23 CET] <norayr> i have 'Unrecognized option 'crf'. Error splitting the argument list: Option not found' error, when ffmpeg gets called from ruby software.
[17:23:04 CET] <norayr> yes that's a different issue, not related to the gif support.
[17:25:09 CET] <c_14> build with x264 support
[17:25:23 CET] <c_14> that or libvpx probably
[17:25:38 CET] <c_14> or x265
[17:25:57 CET] <c_14> in any case, it's trying to use an encoder that wasn't built into ffmpeg
[17:25:59 CET] <DHE> or make sure you put the option in the right place
[17:26:04 CET] <c_14> or that
[17:27:18 CET] <norayr> http://pasted.co/6bd1c327
[17:27:24 CET] <DHE> generally ffmpeg's commandline is ordered "ffmpeg [options for file1] [file1] [options for file2] [file2] ..." where input files are written as "-i inputfile" and outputs are just "outputfile"
[17:27:28 CET] <norayr> DHE: it's not me.
[17:27:32 CET] <norayr> it's ruby software i use.
[17:27:41 CET] <norayr> i need to communicate with developers as well.
[17:27:53 CET] <c_14> what's the command it's trying to execute?
[17:27:54 CET] <norayr> but i think may be i need to compile ffmpeg with other use flags.
[17:27:56 CET] <DHE> you never set a video codec
[17:28:06 CET] <c_14> oh, didn't see the paste
[17:28:18 CET] <norayr> i don't even understand, sometimes they call ffmpeg on gif files, sometimes they don't. it depends on gif file.
[17:28:51 CET] <c_14> rebuild with x264 support
[17:28:56 CET] <DHE> other issues include requesting an audio codec when the input has no audio
[17:29:20 CET] <DHE> yeah the configure build of ffmpeg includes --disable-libx264
[17:29:29 CET] <c_14> If you don't set a video codec and you link against libx264 that's the default, otherwise it's the internel mpeg4 encoder. The mpeg4 encoder doesn't have a crf option
[17:29:31 CET] <c_14> that's why it fails
[17:29:33 CET] <DHE> and I doublechecked, mp4 does default to h264 output
[17:29:50 CET] <c_14> the ruby program is incorretly assuming that libx264 will always be the default and not requesting it explicitly
[17:29:56 CET] <c_14> but then using options that only work for it
[17:30:05 CET] <norayr> okay, i'll rebuild ffmpeg with x264 support.
[17:30:07 CET] <norayr>  thank you!
[17:32:19 CET] <CaimAstraea> Anyone knows where the ngxing rtmp plugin thingie comes in ? I'm currently creating an HLS stream using the ffmpeg HLS muxer but in some tutorials they said to install some plugin for nginx https://github.com/arut/nginx-rtmp-module ? What's the point of that ? Is the performance better than ffmpeg ?
[17:32:38 CET] <DHE> it's a third party module you have to add to nginx at build time
[17:33:05 CET] <DHE> but if you're using ffmpeg's HLS muxer, you don't need it. just nginx (or any other HTTP server) in static content mode is fine
[17:33:35 CET] <CaimAstraea> oh i see yes that's what i'm doing. Was thinking of adding a s3 fuse bucket and write the m3u8 there
[17:33:42 CET] <CaimAstraea> and then hooking up cloudfront to it ?
[17:34:03 CET] <CaimAstraea> the stream is a bit laggy on an amazon t2 micro instance
[17:34:04 CET] <DHE> seems messy, but it could work...
[18:23:24 CET] <RingoTheDog> Does anyone know how to solve "non monotonically increasing dts to muxer in stream" errors...resulting in audio sych issues?
[18:23:48 CET] <RingoTheDog> (I just posted to the pastebin...)
[18:24:01 CET] <RingoTheDog> https://pastebin.com/iGySyxdN
[18:56:00 CET] <alexpigment> RingoTheDog: I can't recall the exact reason for this message, but i've certainly run into it before
[18:56:40 CET] <alexpigment> what I do is run the source file through Tsmuxer (free application), and it inserts sps/pps, and generally kinda fixes formatting problems
[18:56:52 CET] <alexpigment> i've just confirmed that this works on your source file too
[18:59:48 CET] <Fyr> guys, what is vc-1 codec?
[19:00:01 CET] <alexpigment> Fyr: it's an extension of windows media video 9
[19:00:09 CET] <alexpigment> it's a bit more advanced
[19:00:10 CET] <Fyr> I thought the video file is H264, however, FFPROBE shows VC-1.
[19:00:16 CET] <alexpigment> and it's one of the 3 blu-ray video standards
[19:00:24 CET] <Fyr> thanks
[19:00:35 CET] <alexpigment> if i recall, vc-1 is often used for interlaced content on blu-ray
[19:00:49 CET] <alexpigment> if that's the case, i remember hearing that a lot of players had problems with interlaced vc-1
[19:00:56 CET] <alexpigment> software players, that is
[19:03:32 CET] <Fyr> alexpigment, does MP4 sufficiently support VC-1?
[19:03:46 CET] <alexpigment> i don't think so
[19:04:07 CET] <alexpigment> vc-1 should probably either be in MKV, WMV, or TS
[19:04:16 CET] <alexpigment> i'm just speaking off the cuff here
[19:04:17 CET] <RingoTheDog> thanks...is there a CL version of TSMUXER?
[19:04:25 CET] <alexpigment> RingoTheDog: yes
[19:04:27 CET] <Fyr> I remuxed it into MP4, MediaInfo doesn't show shit about it, though, FFPROBE reports, PotPlayer plays it.
[19:04:42 CET] <alexpigment> RingoTheDog: it's included in the download (a zip file, if I recall)
[19:05:01 CET] <RingoTheDog> dope...you are correct
[19:05:30 CET] <alexpigment> Fyr: i'd try MKV or TS
[19:07:05 CET] <Fyr> FFMPEG muxes VC-1 into MP4 without a glitch.
[19:07:54 CET] <alexpigment> right, but I was under the assumption that you were asking about compatibility when you said "does MP4 sufficiently support VC-1"
[19:08:15 CET] <Fyr> ok
[19:08:31 CET] <alexpigment> because ffmpeg can mux codecs into formats when it's not very well supported by other applications
[19:08:59 CET] <Fyr> ok
[19:11:42 CET] <RingoTheDog> Alex...thank you very much!  Yes that tsmuxer appears to be a work around.  Just wondering is this an FFMPEG issue or is the way the source file was created?
[19:12:57 CET] <alexpigment> RingoTheDog: I really have no clue. I just random run into that from time to time. There are bugs logged on trac.ffmpeg.org that have been fixed about it, but I still see it
[19:13:24 CET] <alexpigment> my guess is that the original file has some sort of non-standard situation going on, and tsmuxer kinda just makes the file more compliant
[19:13:39 CET] <RingoTheDog> yes...I read that it was fixed 6 years ago...but seems there may still be an issue
[19:13:56 CET] <RingoTheDog> I have found a few files recently that cause this....most are from twitter videos I believe
[19:17:32 CET] <systemfreq> I'm having an issue with loading a png from file into an AVFrame and then writing the raw image data to file.  The written file starts off with garbage.  Looks like the AVFrame is saying the image data is starting at an earlier address than it should.  Anyone have thoughts?
[19:17:37 CET] <alexpigment> RingoTheDog: it may be worth logging a new issue on trac with your sample
[19:18:36 CET] <systemfreq> Will do.  in a time crunch to get this out, i'll have to switch libraries then for now.
[19:19:16 CET] <alexpigment> systemfreq: sorry, i was talking to another user ;)
[19:22:09 CET] <RingoTheDog> here is another example   http://s000.tinyupload.com/?file_id=93500756624945791830
[19:22:10 CET] <M6HZ> Hello, I encounter an issue since several days when playing http audio streams with ffmpeg based programs. After many hours of playing (more than 24 hours), the audio begins to jolt for few hours before letting place to silence ("rdft" flat with ffplay). This happens at least with mpv, ffplay, mplayer
[19:22:11 CET] <M6HZ> I don't know if it's a problem related to my ISP, which would be weird because the streams are carried over TCP, or if it's a policy from the streamers, but this problem happens for at least three different Media servers owned by different organizations.
[19:22:11 CET] <M6HZ> Or maybe is there something to dig into about ffmpeg ?
[19:22:14 CET] <RingoTheDog> also from twitter video
[19:31:31 CET] <c_14> M6HZ: can you download the stream and reproduce with a file?
[19:38:52 CET] <M6HZ> c_14, I can try
[19:51:55 CET] <saml> ffmpeg -i a.mp4 -i b.mp4       how can I scale a.mp4 to the size of b.mp4?  they have same aspect ratio but b.mp4 is smaller
[19:52:03 CET] <saml> this is so that i can calculate psnr
[19:52:46 CET] <saml> ffmpeg -i a.mp4 -i b.mp4 -filter_complex '[0]scale=640:480[a];[a][1]psnr'
[20:04:41 CET] <alexpigment> saml: presumably you want to make them the same height?
[20:04:59 CET] <alexpigment> if so scale=-2:480
[20:05:43 CET] <alexpigment> (-2 means that the width will be scaled to be an even number. it's probably not important here, but x264 will fail if you try to encode something with an odd numbered resolution)
[20:15:23 CET] <kepstin> saml: in this case, you'd probably just want to use the scale2ref filter. assuming you want to scale the first video to match the second, something like "-filter_complex scale2ref,psnr" would probably be enough.
[20:25:35 CET] <saml> alexpigment, wow thanks. didn't know about -2
[20:25:55 CET] <saml> wow thanks kepstin
[20:26:15 CET] <saml> we need automated assistant for ffmpeg
[20:26:24 CET] <saml> ok google, how do I do this and that with ffmpeg?
[20:30:54 CET] <AeroNotix> trying to play an mp3 file with ffplay but it doesn't exit. Even with -autoexit and -t $n. It exits when playing mp4 files.
[20:32:56 CET] <saml> what's exact command you're running?  and do you have the file somewhere i can download?
[20:42:36 CET] <M6HZ> c_14, I will use: curl -L '[URL]' | tee >(ffplay -) > "$(date --rfc-3339=seconds -u | sed 's/ /_/g')"_-_radio-record.mp3
[20:42:53 CET] <AeroNotix> saml: ffplay -autoexit $mp3file
[20:43:08 CET] <M6HZ> c_14, If you have a better idea or recommandations please let me know.
[20:43:44 CET] <c_14> should be fine
[20:44:24 CET] <M6HZ> great
[20:44:28 CET] <c_14> that way there's at least a reproducer (hopefully)
[20:44:35 CET] <M6HZ> Yes
[20:45:25 CET] <prelude2004c> hey everyone.. i am trying to update my ffmpeg to latest release 3.4.1 but i am having an issue...  -hwaccel cuvid -hwaccel_device 1 used to work but now does not. I did compile it again with cuvid and everything else and no errors.  Only the hwaccess_device does not use 2nd GPU .. and also hwaccel cuvid does not use GPU to decode anymore. Using old version works fine. Something change ?
[20:46:04 CET] <saml> AeroNotix, i think you need both -autoexit -t 5 ?
[20:46:19 CET] <AeroNotix> saml: with -t it stops the audio but doesn't exit
[20:46:32 CET] <saml> hrm
[20:47:41 CET] <c_14> prelude2004c: from what release did you update?
[20:49:19 CET] <alexpigment> AeroNotix: it works over here for what it's worth
[20:49:28 CET] <AeroNotix> alexpigment: interesting.
[20:49:37 CET] <alexpigment> do you have another mp3 to test with?
[20:49:43 CET] <alexpigment> like one that comes from a different source?
[20:50:03 CET] <AeroNotix> alexpigment: I've actually only got a few mp3s on my system. Will find one online
[20:50:49 CET] <alexpigment> not that i think you necessarily have a bad mp3, but given that your command seems to be correct, and it works with other files (your mp4 files), it's good to rule out the possibility that it's content-specific
[20:50:51 CET] <AeroNotix> http://www.sample-videos.com/download-sample-audio.php using the mp3s from here
[20:50:58 CET] <AeroNotix> it still doesn't exit
[20:51:09 CET] <AeroNotix> using `ffplay -autoexit -t 5 $file`
[20:52:44 CET] <AeroNotix> alexpigment: does that file work for you?
[20:52:52 CET] <AeroNotix> ffplay version 3.4.1 here
[20:53:13 CET] <alexpigment> checking now...
[20:53:39 CET] <AeroNotix> with -t 5 the audio stops but ffplay doesn't exit
[20:54:13 CET] <alexpigment> yep, ffplay stops over here
[20:54:19 CET] <AeroNotix> which version of ffplay?
[20:54:37 CET] <alexpigment> well, i just built it yesterday, so it's a nightly
[20:54:41 CET] <AeroNotix> ok
[20:54:52 CET] <AeroNotix> 3.4.1 doesn't work here at least
[20:55:03 CET] <c_14> I have a relatively old git snapshot and it works here
[20:55:08 CET] <alexpigment> out of curiosity, have you tried taking your variable out of the equation?
[20:55:16 CET] <alexpigment> you're doing $file
[20:55:26 CET] <alexpigment> but just try the actual filename itself via command line
[20:55:26 CET] <AeroNotix> I'm not using a variable, I just did that to make it clear what I'm running for you
[20:55:32 CET] <alexpigment> ok, just checking
[20:56:19 CET] <c_14> AeroNotix: where did you get your ffmpeg build from?
[20:56:24 CET] <AeroNotix> c_14: arch repos
[20:56:42 CET] <AeroNotix> another user in #archlinux is complaining of the same behaviour. I thought I'd ask here
[20:56:52 CET] <AeroNotix> it might be related to the build we're using, but seems a strange feature to break
[20:56:59 CET] <c_14> yeah
[20:57:00 CET] <AeroNotix> since it looks like -autoexit was added in 2010
[20:57:03 CET] <c_14> going to try with a 3.4.1 here
[20:57:06 CET] <AeroNotix> thanks!
[20:57:23 CET] <alexpigment> 32-bit or 64-bit?
[20:57:25 CET] <AeroNotix> 64
[20:57:28 CET] <alexpigment> i'm downloading a version for windows right now
[20:57:47 CET] <AeroNotix> ~ uname -a
[20:57:49 CET] <AeroNotix> Linux xenocorp 4.14.13-1-ARCH #1 SMP PREEMPT Wed Jan 10 11:14:50 UTC 2018 x86_64 GNU/Linux
[20:58:31 CET] <saml> mplayer -endpos 1 SampleAudio_0.4mb.mp3     ends.  ffplay -autoexit -t 1 SampleAudio_0.4mb.mp3   ends as well
[20:58:32 CET] <alexpigment> oh weird. i get some WASAPI initialization error when i try the 3.4.1 static build from Zeranoe
[20:58:44 CET] <saml> ffplay -t 1 SampleAudio_0.4mb.mp3    stops audio but never exits
[20:58:57 CET] <c_14> yeah, you need the autoexit flag
[20:58:59 CET] <AeroNotix> saml: 3.4.1?
[20:59:14 CET] <saml> 3.3.6
[20:59:24 CET] <AeroNotix> I guess I can try that
[20:59:36 CET] <AeroNotix> but need to go afk. Thanks for helping so far people :)
[20:59:41 CET] <c_14> 3.4.1 autoexits here
[20:59:46 CET] <c_14> just built it
[21:00:57 CET] <alexpigment> well, i don't know what they borked in 3.4.1 for windows, but ther's an audio driver initialization error. works fine in the build i made yesterday...
[21:01:09 CET] <alexpigment> anyway, unrelated to the issue at hand
[21:09:54 CET] <AeroNotix> I'll be back in a couple of hours, I will try building 3.4.1 locally and see
[21:11:45 CET] <prelude2004c> c_14 , i went to the lates version
[21:12:04 CET] <prelude2004c> the old version was ffmpeg version N-83514-g27a49a2
[21:12:06 CET] <prelude2004c> that was working fine
[21:13:06 CET] <c_14> that doesn't seem like a valid commit hash?
[21:13:44 CET] <c_14> But there were a few relatively recent changes to cuvid stuff
[21:13:47 CET] <c_14> not sure if there's any docs
[21:14:29 CET] <c_14> there's the new nvdec hwacell for one
[21:21:48 CET] <prelude2004c> ok cool... nvdec
[21:21:54 CET] <prelude2004c> so maybe i have to use nvdec to decode
[21:22:07 CET] <prelude2004c> do i have to compile with it you think?
[21:22:24 CET] <c_14> yeah, I think it's a separate configure option
[21:23:25 CET] <saml> similar to scale2ref,  is there a filter to match framerate of two files?
[21:24:02 CET] <saml> and also match pixel format
[21:25:06 CET] <prelude2004c> Unknown option "--enable-nvdec".
[21:29:05 CET] <prelude2004c> so that's one thing.. another thing is... i can't choose the second GPU
[21:29:05 CET] <c_14> it's listed here
[21:29:12 CET] <c_14> maybe it's not in 3.4.1
[21:29:53 CET] <prelude2004c> its not in 3.3.6 iether
[21:29:56 CET] <prelude2004c> i checked there too
[21:30:18 CET] <alexpigment> prelude2004c: not that i have any idea what's going, but just to make sure, do you have a monitor plugged into the second gpu (if it's a desktop)?
[21:30:24 CET] <prelude2004c> so the two items are ...  i can't chose 2nd GPU  and it wont use nv decoder
[21:30:44 CET] <alexpigment> i know hardware *encoding* will fail if there's not a monitor plugged into it
[21:30:47 CET] <prelude2004c> alex, no these system are basic shells just for transcoding
[21:31:03 CET] <prelude2004c> hardware encoding is fine
[21:31:18 CET] <prelude2004c> its the decoding and only on latest release .. the old version i been playing with seems to be fine
[21:31:19 CET] <alexpigment> ok, just wanted to mention it in case it was relevant
[21:31:47 CET] <alexpigment> prelude2004c: understandable. i wasn't implying that it wasn't new behavior :)
[21:32:03 CET] <prelude2004c> the problem i have with the old version is im getting random " ffmpeg non-monotonous dts in output stream "
[21:32:11 CET] <prelude2004c> i'm hoping new versions have been able to addres this
[21:34:26 CET] <SortaCore> if you guys want to test with NVIDIA, I have an Intel QSV motherboard + NVIDIA external, and a screen plugged into both
[21:34:47 CET] <SortaCore> windows 10 x64, NVIDIA GTX 970
[21:36:15 CET] <alexpigment> Sorta: yep, that's the same setup i've got over here ;)
[21:36:37 CET] <alexpigment> once the AMF encoding gets further along (if/when), i'll have an AMD card in here too
[21:36:59 CET] <alexpigment> and i will enjoy those lovely BSODs that come with it
[21:41:38 CET] <SortaCore> but less Spectre problems right
[21:42:42 CET] <alexpigment> well, the cpu will still be intel, so no :)
[21:42:59 CET] <alexpigment> actually, i run win 7
[21:43:13 CET] <alexpigment> and all of my systems are haswell or earlier
[21:43:43 CET] <alexpigment> and apparently the slowdowns from the patches affect pre-win10 and pre-skylake the most :(
[22:00:40 CET] <SortaCore> today fixing DTS/PTS, tomorrow the world
[22:09:55 CET] <prelude2004c> c_14 , so any other idea ?
[22:09:59 CET] <ddubya> I'm trying to fix a bug in a transcoder. They are encoding audio, and ignoring any PTS values set in the packet (by the encoder). The result is that an empt space is created at the start of the exported file, because the first PTS is 0 when it should be negative
[22:10:04 CET] <c_14> prelude2004c: not really, sorry
[22:10:27 CET] <ddubya> my question is then, if the encoder provides no pts what should I be setting it to (for audio encoding)
[22:12:33 CET] <ddubya> more specifically, avcodec_encode_audio2() provides no pts, what pts do I set before av_interleaved_write_frame()
[22:15:28 CET] <DHE> ddubya: are you decoding a file, or generating the bitstream directly?
[22:15:41 CET] <DHE> even reading a .wav file should produce AVFrames with some pts value you could leverage
[22:15:47 CET] <ddubya> source audio is generated
[22:16:23 CET] <ddubya> the source audio starts at 0 PTS always. But the output PTS sometimes has to be negative at the start, eample m4a format
[22:17:09 CET] <ddubya> but wav format it doesn't have to be
[22:23:28 CET] <saml> filtergraph, how do I do this? its syntax is crazy
[22:24:58 CET] <DHE> -filter_complex "[inputtag] filtername=option1=value1:option2=value2 [outputtag] ; [anotherinput] filtername=... [anotheroutput]"
[22:25:44 CET] <DHE> where an input could be [0:v] for the video of the first input, or a label that is an output for another filter. outputs could go into other filter inputs, or delivered to an output file with "-map [outputtag]"
[22:33:16 CET] <saml> if i'm dealing with two inputs, how do I apply framerate to both and continue on with the chain?
[22:34:13 CET] <alexpigment> do they have different frame rates?
[22:34:17 CET] <saml> scale2ref,[0]framerate[0];[1]framerate[1],format=pix_fmts=yuv420p,psnr=stats_file=-
[22:34:36 CET] <alexpigment> oh right, the psnr thing :)
[22:35:42 CET] <kepstin> saml: if the two inputs have different framerates, I don't think you can usefully do anything with psnr
[22:35:50 CET] <kepstin> you really need the videos to match frame-for-frame
[22:36:03 CET] <saml> that's why i'm trying to force framerate and pix_fmts to some constant
[22:37:03 CET] <alexpigment> can we reasonably assume that the framerates are evenly divisible by one another?
[22:37:13 CET] <alexpigment> like, for instance, 29.97 and 59.94
[22:38:02 CET] <saml> current example videos are: 59.94 vs 60
[22:38:39 CET] <saml> there should be a cli called psnr  that takes any two videos with different resolution, framerate, ... etc and works
[22:42:50 CET] <alexpigment> saml: how would that work, exactly?
[22:43:20 CET] <alexpigment> it's going to compare pixels that don't exist, frames that don't exist, etc
[22:44:07 CET] <saml> 1. scale the larger video (Reference video) down to match main video.  2. adjust framerate of both videos to gcd ?    3. psnr
[22:44:12 CET] <alexpigment> psnr is a very strict, cold, and robotic way of measuring quality. it's not a good approximation for a human's visual perception of quality
[22:44:28 CET] <saml> what other options do i have?
[22:44:37 CET] <alexpigment> saml: start with explaining your overall goal
[22:44:42 CET] <alexpigment> maybe you're going about this wrong
[22:44:51 CET] <saml> yeah i'm just given a ticket :P
[22:45:06 CET] <alexpigment> does the ticket explain the goal?
[22:45:30 CET] <saml> i want to start recording some kind of quality metrics. so that as I introduce different ffmpeg parameters, I can observe how it affects this "quality metrics"
[22:45:46 CET] <alexpigment> for example, is it like "figure out the visual quality lost by downscaling to 480p and dropping frames"?
[22:46:10 CET] <furq> !filter libvmaf @saml
[22:46:10 CET] <nfobot> saml: http://ffmpeg.org/ffmpeg-filters.html#libvmaf
[22:46:15 CET] <saml> i think it's mostly about video codec. if i switch to vp9, will customer complain due to visual artifacts.. etc
[22:46:20 CET] <kepstin> hmm, as far as I know, there's no automatic metrics that cover motion perception due to framerate changes
[22:46:27 CET] <alexpigment> saml: that's fine
[22:46:28 CET] <furq> you probably want to look at that if you want quality metrics that aren't useless
[22:46:34 CET] <furq> i have no idea if it works with different framerates etc though
[22:46:36 CET] <furq> probably not
[22:46:42 CET] <kepstin> but if you avoid that, and have frame-per-frame video matches, then vmaf is a good start, yeah
[22:46:46 CET] <alexpigment> different codecs, different bitrates, etc are fine for comparing psnr
[22:48:14 CET] <saml> man we don't build ffmpeg with vmf
[22:48:22 CET] <alexpigment> saml: but if you want to know how much quality is lost by, say, downscaling from 1080p to 480p, then the answer is 4.5x
[22:49:02 CET] <alexpigment> or by dropping frames, from 59.94 to 29.97, the answer is 2x
[22:49:03 CET] <saml> x264  vs vp9  this kind of stuff doesn't change the "metric"?
[22:49:15 CET] <alexpigment> although obviously halving the framerate is much more distracting than dropping the resolution (within reason)
[22:49:19 CET] <furq> it'll mess with psnr/ssim because both of those use psy optimisations
[22:49:32 CET] <furq> you'll still get the "correct" result but it won't of much value
[22:49:35 CET] <furq> be
[22:49:36 CET] <saml> yeah, i think that's why we want to start with psnr
[22:49:38 CET] <alexpigment> but at least the principle will work
[22:49:49 CET] <saml> i thought psnr would be the simplest.. but looks real difficult
[22:50:03 CET] <kepstin> psnr is the simplest, which is why it's not really all that useful
[22:50:20 CET] <saml> so for all comparisons, i need to match framerate and resolution
[22:50:26 CET] <kepstin> turns out that "simple" is not accurately representative of how people see :)
[22:50:34 CET] <saml> yeah
[22:50:46 CET] <saml> i just want something simple to start. and can iterate
[22:50:50 CET] <furq> yeah it's not really suitable for modern codecs
[22:51:07 CET] <saml> but what i'm gathering is that getting two videos match resolution and framerate is a hard problem?
[22:51:36 CET] <furq> resolution is easy-ish although the scaling may introduce artifacts which aren't present in the other video
[22:51:47 CET] <saml> that's true
[22:51:49 CET] <alexpigment> saml: it's easy. but comparing quality between two videos when the basic elements of those videos aren't the same is a bit more difficult
[22:51:58 CET] <furq> or may hide artifacts if you scale and then pass straight to psnr without reencoding it first
[22:52:06 CET] <saml> i see
[22:52:19 CET] <furq> and yeah fps is more difficult because the frames kept in the comparison clip might be different
[22:52:27 CET] <kepstin> saml: these metrics are intended to compare two different encodings of the same original signal, so the underlying signal in both cases is assumed to match.
[22:52:46 CET] <saml> that's a good way to put it kepstin
[22:52:54 CET] <AndrewPRS> Hi there! is there any tool in ffmpeg that I can use to analyze the audio tracks in dvd?
[22:53:07 CET] <kepstin> AndrewPRS: analyze them to find out what?
[22:53:10 CET] <furq> ffmpeg's handling of dvd is pretty poor
[22:53:29 CET] <furq> you should usually just be able to ffprobe a vob to find the audio tracks
[22:53:42 CET] <AndrewPRS> I have a music album that I bought and it has also a DVD with extra high quality tracks
[22:53:56 CET] <AndrewPRS> I wanna get the tracks from my DVD to flac
[22:54:11 CET] <furq> oh, dvd audio
[22:54:13 CET] <AndrewPRS> I was considering using handbrake, but it is specially designed for video, not audio
[22:54:17 CET] <furq> i'm not really sure how that worls on account of i've never seen one
[22:54:29 CET] <alexpigment> AndrewPRS: are you on windows?
[22:54:33 CET] <AndrewPRS> nope
[22:54:34 CET] <AndrewPRS> debian
[22:54:39 CET] <alexpigment> hmmm
[22:54:52 CET] <alexpigment> linux + these types of projects = not the best solution usually
[22:55:01 CET] <furq> that's annoying because they removed transcode from debian for some unknown reason
[22:55:02 CET] <alexpigment> anyway, maybe dvd decrypter exists for linux
[22:55:20 CET] <furq> i forget if i use dvdbackup or dvdunauthor on *nix
[22:55:32 CET] <kepstin> I'd probably use dvdbackup to pull the individual chapters to vobs, then ffmpeg can transcode to flac
[22:55:55 CET] <alexpigment> but basically you will probably want to find something that rips vobs by title or chapter. then you can use ffmpeg to encode the audio from PCM/whatever to FLAC
[22:55:57 CET] <AndrewPRS> kepstin, I have AOB and VOB files on my dvd
[22:56:09 CET] <kepstin> dvdbackup lets you select by title/chapter, yeah
[22:56:10 CET] <AndrewPRS> but I think the info of their layout is in the NFO files
[22:56:14 CET] <AndrewPRS> which is not plain text
[22:56:32 CET] <furq> i don't know if either of those handle audio_ts though
[22:56:42 CET] <kepstin> hmm, it's probably not actually an audio dvd?
[22:56:42 CET] <alexpigment> what if you just take an AOB file and do ffmpeg -i [file] -c:a flac output.flac
[22:56:59 CET] <furq> if it's anything like video dvd then the track splits will be in the IFO
[22:57:05 CET] <AndrewPRS> alexpigment, then it contains several tracks all mixed in a single flac file
[22:57:07 CET] <furq> right
[22:57:10 CET] <kepstin> AndrewPRS: when you look at the dvd files, do you have an 'audio_ts' and/or 'video_ts' directory? does one of them have files?
[22:57:14 CET] <AndrewPRS> and I don't know how the tracks map
[22:57:19 CET] <furq> there are windows tools that do this but idk anything on *nix
[22:57:31 CET] <AndrewPRS> kepstin, I have audio_ts and video ts
[22:57:35 CET] <furq> AndrewPRS: http://avisynth.nl/index.php/DVDAGuide
[22:57:35 CET] <AndrewPRS> and both have VOB files
[22:57:41 CET] <AndrewPRS> also audio_ts has AOB file
[22:57:46 CET] <kepstin> both have files, eh? so it probably is dvd-audio then :/
[22:57:47 CET] <AndrewPRS> and both have NFO
[22:58:01 CET] <furq> yeah if you have AOBs then it's DVDA
[22:58:06 CET] <AndrewPRS> kepstin, it is a dvd that includes audio and also a live performance from a concert
[22:58:29 CET] <AndrewPRS> I am stunned that no one has made a simple gui tool for this :/
[22:58:34 CET] <kepstin> yeah, I don't know of any linux software that correctly handles stuff like chapters for dvd-audio
[22:58:35 CET] <furq> http://dvd-audio.sourceforge.net/
[22:58:36 CET] <furq> maybe this?
[22:58:37 CET] <AndrewPRS> I mean, I'm a c++ dev so I could :P
[22:58:37 CET] <alexpigment> AndrewPRS: does dvdbackup allow you to choose streams?
[22:58:43 CET] <furq> this is the only foss tool i can find that does it
[22:58:53 CET] <kepstin> alexpigment: dvdbackup is video only i think :/
[22:58:56 CET] <AndrewPRS> furq, looks like that's for authoring
[22:58:57 CET] <alexpigment> ahh
[22:59:34 CET] <furq> oh
[22:59:36 CET] <furq> yeah maybe
[22:59:40 CET] <alexpigment> AndrewPRS: this is a complete non-answer and isn't even particularly helpful, but you really want to have a Windows box around for this stuff
[22:59:44 CET] <ddubya> any ideas why my transcoder produces a file with first pts/dts = 0 and ffmpeg cmdline is -1024/-1024 (which is the size of one audio frame in samples)
[23:00:03 CET] <furq> apparently dvd audio explorer works in wine
[23:00:18 CET] <kepstin> ddubya: what codec/container?
[23:00:25 CET] <furq> i'd probably just do that, honestly
[23:00:30 CET] <furq> much as i dislike having to use wine
[23:00:38 CET] <AndrewPRS> I could also develop a tool to extract audio :P !
[23:00:39 CET] <ddubya> kepstin, container is aac/mp4
[23:00:52 CET] <AndrewPRS> there must be some library ...
[23:01:12 CET] <furq> there's libdvdread and libdvdnav but i have no idea if they deal with dvda
[23:01:30 CET] <kepstin> ddubya: hmm. aac has approximately 1 frame of preroll, so the actual audio start is in the second frame. so that kinda makes sense.
[23:01:39 CET] <furq> oh
[23:01:43 CET] <furq> AndrewPRS: https://github.com/jdeblese/dvda-decode
[23:01:45 CET] <furq> this looks promising
[23:02:02 CET] <ddubya> kepstin, is there a way to get the codec preroll or know it based on some flag
[23:02:57 CET] <kepstin> ddubya: you're writing your own transcoding software using the libavcodec apis?
[23:03:07 CET] <ddubya> kepstin, yes
[23:03:50 CET] <furq> AndrewPRS: you will probably need to have libdvdcss installed, which you normally need to build yourself
[23:04:03 CET] <furq> but dvdread/dvdnav will dlopen it if it's present so you don't need to build those
[23:04:17 CET] <furq> obviously if the disc's not encrypted then you don't need to bother
[23:04:42 CET] <AndrewPRS> well I hope that they aren0t
[23:04:49 CET] <AndrewPRS> I'm not sure though :/
[23:05:15 CET] <kepstin> encrypted dvd stuff isn't really a problem, libdvdcss will crack it in a few seconds on a modern box.
[23:05:19 CET] <furq> yeah
[23:05:41 CET] <furq> i should "acquire" an audio dvd and see if the regular dvd tools will do anything useful with it
[23:05:56 CET] <AndrewPRS> I have three on my desktop T_T
[23:06:56 CET] <alexpigment> i think i technically have one somewhere, but i think i bought it on accident thinking it was a dvd with music videos on it ;)
[23:07:13 CET] <alexpigment> no clue where it is or even what it is
[23:07:28 CET] <AndrewPRS> I wonder why no library supports DVDA
[23:07:30 CET] <AndrewPRS> :/ !
[23:07:35 CET] <furq> it's relatively uncommon
[23:07:40 CET] <alexpigment> yeah, it was a niche format
[23:07:45 CET] <alexpigment> post sacd
[23:07:49 CET] <AndrewPRS> I guess it can't be vert different from DVDvideo
[23:07:50 CET] <AndrewPRS> *very
[23:08:10 CET] <AndrewPRS> I may have some sacd's too
[23:08:11 CET] <furq> i assume it's just IFO chapters, so yeah
[23:08:33 CET] <kepstin> now sacd is actually something different :/
[23:08:33 CET] <furq> i'm guessing regular dvd tools just haven't bothered supporting all the extra audio formats
[23:09:12 CET] <saml> what's difference between , and ; in filtergraph?  both seems to be sequence
[23:09:16 CET] <furq> oh lol
[23:09:22 CET] <furq> DVD-Audio streams can be encrypted. The encryption is called Content Protection for Prerecorded Media (CPPM), which uses a media key block (MKB) to authenticate DVD-Audio players.
[23:09:22 CET] <saml> http://ffmpeg.org/ffmpeg-filters.html#Filtering-Introduction
[23:09:26 CET] <furq> is this different encryption
[23:09:41 CET] <kepstin> saml: "," sends the output of the filter before the , to the input of the filter after the ,
[23:09:41 CET] <furq> if it is then that's a lot of fun
[23:09:43 CET] <saml> not sure why it's split;crop,vflip;overlay   why comman between crop and vflip
[23:10:03 CET] <kepstin> saml: ";" separates unrelated portions of the filtergraph with independent inputs and outputs
[23:10:23 CET] <furq> right
[23:10:33 CET] <furq> , needs one output on the left and one input on the right
[23:10:36 CET] <saml> so,  instead of , I can just use ; and specify input and output sequentially?
[23:10:42 CET] <furq> you can if you want
[23:10:48 CET] <saml> ah thanks
[23:11:01 CET] <kepstin> saml: yes. "filtera[pad];[pad]filterb" is similar to "filtera,filterb"
[23:11:10 CET] <kepstin> except for the case where there's multiple outputs and inputs
[23:12:06 CET] <alexpigment> i'm trying to find that DVD Audio dvd i have
[23:12:17 CET] <saml> ffmpeg -i "$video" -i "$reference" -filter_complex "scale2ref [vid][ref]; [vid] framerate [vid]; [ref] framerate [ref]; [vid][ref] psnr=stats_file=-" -f null -
[23:12:38 CET] <alexpigment> and i found one that *isn't* dvda but has high res audio on it... what is the point of dvda? just not having to make any video stream?
[23:12:39 CET] <saml> now i can produce garbage psnr
[23:12:52 CET] <furq> vobs need to have a dummy video stream, yeah
[23:12:57 CET] <furq> also i think dvda supports additional codecs
[23:13:00 CET] <furq> such as the ever-popular mlp
[23:13:01 CET] <kepstin> saml: that makes no sense, you can't output to the [vid] from multiple fitlers, you need to use different names for each pad
[23:13:06 CET] <alexpigment> furq: ahh
[23:13:20 CET] <alexpigment> the one i'm looking at is dts 24/96
[23:13:28 CET] <saml> ah i see. i thought names would be overwritten
[23:13:28 CET] <saml> thanks
[23:13:31 CET] <alexpigment> but it just has a picture of the album that sits there
[23:13:36 CET] <kepstin> saml: how is the psnr filter suppose to know whether to read from the output of scale2ref or framerate? :)
[23:13:54 CET] <AndrewPRS> alexpigment, how are you looking at it ?
[23:14:00 CET] <AndrewPRS> what software are you using?
[23:14:06 CET] <furq> AndrewPRS: it's a video dvd
[23:14:12 CET] <alexpigment> it's a video dvd, yes ;)
[23:14:15 CET] <alexpigment> it's playing in VLC
[23:14:17 CET] <AndrewPRS> oh
[23:14:25 CET] <alexpigment> not that i like VLC, but it's quicker than mounting the ISO :)
[23:14:42 CET] <furq> yeah it looks like dvda has a different encryption system than dvd video
[23:14:54 CET] <furq> so presumably libdvdcss doesn't work there
[23:15:01 CET] <furq> you're probably just going to have to use dvd audio explorer in wine
[23:15:01 CET] <AndrewPRS> cra'
[23:15:03 CET] <AndrewPRS> crap!
[23:15:11 CET] <saml>  PSNR y:inf u:inf v:inf average:inf min:inf max:inf    hahah same input and reference.
[23:15:28 CET] <kepstin> you can throw 48kHz 16-bit LCPM on dvd-video, and I suspect they're easier/cheaper to author, which is presumably why most of the music i've seen is actually on dvd-video.
[23:15:55 CET] <alexpigment> kepstin: you can even do 24bit
[23:16:11 CET] <alexpigment> somewhat rare, but i've definitely got quite a few like that
[23:16:15 CET] <furq> there is a cli tool that does it but it appears to be windows-only
[23:16:17 CET] <alexpigment> just music video dvds
[23:18:09 CET] <alexpigment> so i think i've just kinda realized how pointless DVDA discs are :)
[23:18:30 CET] <alexpigment> i have a feeling the only reason for their existence is to allow standalone DVDA players
[23:22:02 CET] <alexpigment> well, i guess MLP allow you to do 6-channel...
[23:23:02 CET] <AndrewPRS> looks like I can open it with DVDExplorer
[23:23:06 CET] <AndrewPRS> :D
[23:24:39 CET] <AndrewPRS> thank you guys! furq alexpigment Kei_N
[23:24:41 CET] <AndrewPRS> kepstin,
[23:26:24 CET] <furq> alexpigment: you can have 192k lpcm on dvda as well
[23:26:44 CET] <furq> obviously that doesn't make it any less pointless, but it is a difference
[23:27:09 CET] <furq> and yeah mlp will save quite a bit on size so you can pack more data in
[23:27:14 CET] <furq> without having to resort to dts
[23:27:17 CET] <AndrewPRS> after extracting the mlp files, can I convert them with ffmpeg?
[23:27:42 CET] <furq> yeah ffmpeg has an mlp decoder
[23:27:47 CET] <AndrewPRS> ahh ok great
[23:27:48 CET] <AndrewPRS> furq
[23:28:06 CET] <AndrewPRS> do you also think that over 16/48kHz is pointless?
[23:28:12 CET] <furq> it is for playback
[23:28:22 CET] <AndrewPRS> that's what I've read
[23:28:30 CET] <AndrewPRS> do you have a portable player?
[23:28:37 CET] <furq> i do but i don't use it often
[23:28:48 CET] <AndrewPRS> what kind of equipment do you use to listen to music?
[23:28:57 CET] <AndrewPRS> I'm just curious
[23:29:05 CET] <furq> usually just my pc, headphones and an amp
[23:29:11 CET] <furq> nothing particularly fancy
[23:29:21 CET] <AndrewPRS> favourite band?
[23:29:23 CET] <AndrewPRS> :P !
[23:30:21 CET] <furq> https://www.youtube.com/watch?v=aowa2mBRGec
[23:30:35 CET] <alexpigment> furq: yeah, the difference is apparently that you have a maximum bitrate for audio, which requires that you use MLP when you want to use high sampling rate + 6 channels
[23:31:05 CET] <alexpigment> so while you can have 24/96 on both, you can't do 6-channels of PCM
[23:31:33 CET] <alexpigment> and it looks like the max bitrate for the audio portion of dvd video is lower, although i have a feeling most players don't care
[23:33:13 CET] <furq> https://en.wikipedia.org/wiki/DVD-Audio#Sound_quality
[23:33:14 CET] <furq> lol
[23:33:24 CET] <furq> those parens are doing a lot of work
[23:34:04 CET] <alexpigment> AndrewPRS: the bit depth thing is about the noise floor, which isn't particularly important on a digital medium once something has been mixed and mastered. it's really important when you're recording and you may need to boost your signals and compress that same track without destroying dynamics
[23:34:54 CET] <alexpigment> as for the 96khz thing... i'd like to think i have good ears. i've done a lot of work in the music realm, including recording and mixing bands
[23:34:58 CET] <alexpigment> i really can't tell the difference
[23:35:15 CET] <alexpigment> but you use it because you might as well ;)
[23:35:42 CET] <alexpigment> having said that, there is some validity to having your sampling frequency being an even multiple of your delivery frequency
[23:35:48 CET] <alexpigment> and that usually means 44.1
[23:36:12 CET] <AndrewPRS> and what is the frequency of love?
[23:36:13 CET] <alexpigment> your primary delivery format is rarely 48khz or 96khz
[23:36:27 CET] <alexpigment> the frequency of love is only 32khz, oddly enough ;)
[23:36:28 CET] <furq> it's not that uncommon for it to be 48k these days
[23:36:35 CET] <alexpigment> furq: delivery though
[23:36:42 CET] <alexpigment> CD, itunes, spotify, etc
[23:36:52 CET] <furq> itunes supports 48k doesn't it
[23:36:59 CET] <alexpigment> of course
[23:37:01 CET] <furq> if you're a digital only label then there's no need to go near 44.1
[23:37:05 CET] <furq> and there's plenty of those around
[23:37:06 CET] <alexpigment> but if buy from the itunes store, it's 44.1
[23:37:13 CET] <utack> at this point probably not evne apple knows what itunes does or does not do or support
[23:37:15 CET] <furq> yeah i meant the store
[23:38:04 CET] <alexpigment> utack: well, they *did* cannibalize their own store with their streaming service ;)
[23:38:04 CET] <furq> then again itunes store is still 100% aac isn't it
[23:38:11 CET] <alexpigment> pretty sure
[23:38:14 CET] <furq> so downsampling losses are the least of your worries
[23:38:22 CET] <alexpigment> last i recall it was 256kbpc aac at 44.1
[23:39:02 CET] <utack> so where does alac come from? apple must sell that somewhere?
[23:39:07 CET] <furq> apparently spotify uses whatever sample rate they get sent
[23:39:08 CET] <utack> why else make their own lossless format?
[23:39:18 CET] <furq> i expect other streaming services do the same
[23:39:24 CET] <alexpigment> furq: well, i'd assume that means up to 48k
[23:39:30 CET] <alexpigment> but i have no way to tell what they do
[23:39:31 CET] <furq> and obviously if you're distributing on bandcamp or whatever then you can use whatever you want
[23:39:36 CET] <furq> and yeah i meant between 44.1 and 48
[23:39:55 CET] <alexpigment> still though, i haven't seen much evidence of people moving past 44.1 for delivery
[23:40:09 CET] <furq> as long as cd exists then most people won't
[23:40:12 CET] <alexpigment> unless you go to a source that specializes in flac
[23:40:15 CET] <furq> i have plenty of 16/48 though
[23:40:42 CET] <furq> a lot of electronic labels are either just web or web/vinyl
[23:40:59 CET] <alexpigment> i have a ton of download cards from vinyl
[23:41:09 CET] <alexpigment> i've never really looked to see if any of that is 48khz
[23:41:56 CET] <alexpigment> well, my Baroness album is 48khz/320kbps mp3 with the vinyl download code
[23:42:26 CET] <alexpigment> which, admittedly, makes sense for audio that was pressed onto new vinyl
[23:42:45 CET] <alexpigment> Manchester Orchestra was 44.1
[23:44:32 CET] <alexpigment> yeah, most of these are just 44.1khz 320kbps mp3s
[23:46:40 CET] <alexpigment> granted, i haven't redeemed most of them in the past 2 years because of spotify. just makes the whole point of a local music library kinda blurry
[00:00:00 CET] --- Sat Jan 20 2018