[Ffmpeg-devel-irc] ffmpeg-devel.log.20160723

[burek]

[00:46:25 CEST] <kode54> https://gist.github.com/kode54/5224b2446c871b5251945f990bdce507
[00:46:36 CEST] <kode54> that introduces two changes
[00:46:45 CEST] <kode54> 1) read id3v2 "COMM" frames
[00:47:05 CEST] <kode54> 2) parse "iTunSMPB" comment frames for file info
[00:48:17 CEST] <atomnuker> mp3_parse_itunes_tag() doesn't look good at all
[00:49:58 CEST] <atomnuker> do if (!s->metadata || !(de = av_dict_get(s->metadata, "iTunSMPB", NULL, 0))) return
[00:50:13 CEST] <atomnuker> that'll get the indentation level down by 1
[00:51:20 CEST] <atomnuker> do the same for the sscanf
[00:51:39 CEST] <kode54> okay, then
[00:51:46 CEST] <pomaranc> does anyone know if someone is working on a mp4 dash muxer?
[00:51:50 CEST] <atomnuker> sanitize the hell out of the string before sscanf too
[00:55:57 CEST] <kode54> and how do I do that?
[00:56:13 CEST] <kode54> this is supposed to be a machine generated string
[00:56:42 CEST] <kode54> for loop and make sure it contains nothing but hex and whitespace?
[00:57:46 CEST] <atomnuker> just do a basic sanity check on the length
[01:05:55 CEST] <kode54> done
[01:06:18 CEST] <kode54> it now checks if length is too short for the minimum size of the known number of fields, and too long for the same
[01:06:26 CEST] <kode54> and the indenting should be significantly better
[01:13:50 CEST] <atomnuker> make the i in for (i = 0; i < 8; ++i) { go i++ instead of ++i
[01:32:46 CEST] <kode54> is it pleasing now? I also forgot something with calculating the duration
[01:33:09 CEST] <kode54> aww
[01:33:11 CEST] <kode54> I botched it
[01:35:08 CEST] <kode54> there we go
[01:37:59 CEST] <atomnuker> yeah, test it and post it on the ML
[01:46:39 CEST] <kode54> verified working with this iTunes rip I originally designed it for, based on a published topic on the field format
[01:50:10 CEST] <kode54> do I need to filter my email addresses when formatting the patches?
[01:52:51 CEST] <atomnuker> filter?
[01:56:47 CEST] <Chloe> kode54: use git format-patch -1 to generate the patch (note, dont use git diff, or diff), and then send it as an attachment or use git send-email to send it
[01:57:03 CEST] <kode54> k
[02:00:35 CEST] <kode54> ffmpeg-devel at ffmpeg.org ?
[02:01:42 CEST] <Chloe> yes
[02:04:26 CEST] <Chloe> kode54: I did `git config sendemail.to ffmpeg-devel at ffmpeg.org` to setup the repository's email so that I can just use `git send-email 0001-my-patch.patch` (after setting up the global smtp settings as well)
[02:04:47 CEST] <kode54> k
[02:05:50 CEST] <Chloe> kode54: this may also have more information: https://ffmpeg.org/developer.html#Submitting-patches-1
[02:06:26 CEST] <Chloe> oh yeah, remember to `make fate`
[02:07:15 CEST] <JEEB> pomaranc: you probably missed libavformat/dashenc.c?
[02:13:06 CEST] <pomaranc> JEEB: wow, I did
[02:13:21 CEST] <pomaranc> JEEB: the documentations says there is only webm_dash_manifest
[02:13:43 CEST] <pomaranc> I will try it out, thanks
[02:14:23 CEST] <JEEB> yeah, many muxers don't have documentation :P you just see them on the -formats listing
[02:14:35 CEST] <JEEB> and if you just specify output file name as .mpd
[02:48:05 CEST] <Chloe> kode54: there seem to be a few whitespace issues with your first patch: http://sprunge.us/MZHQ your second patch applies cleanly though
[02:48:14 CEST] <kode54> oh crap
[02:48:26 CEST] <kode54> did I use tabs or something?
[02:48:30 CEST] <kode54> dammit
[02:50:17 CEST] <kode54> weird with the first one
[02:50:30 CEST] <kode54> I even remade the patch
[02:51:03 CEST] <Chloe> don't worry about it, just send a v2 with a comment 'fixed whitespace' or equivalent
[02:59:30 CEST] <kode54> I hope that does it
[02:59:54 CEST] <kode54> dammit, I set --subject
[03:00:57 CEST] <kode54> clearly, git send-email is not as useful as I'd like it to be
[03:04:56 CEST] <Chloe> you can use --annotate
[03:05:03 CEST] <kode54> ah, oops
[03:05:11 CEST] <Chloe> And then you can edit the subject directly
[03:39:50 CEST] <Chloe> kode54: could a copy of the sample you're using to test?
[03:40:33 CEST] <kode54> it's a full album ripped using iTunes
[03:40:43 CEST] <kode54> should I package it up somewhere?
[03:42:48 CEST] <Chloe> kode54: see PM.
[03:53:33 CEST] <kode54> weird, the delay/padding isn't being subtracted as far as ffprobe is concerned
[03:54:27 CEST] <Chloe> kode54: I haven't looked at any of the IDv2 stuff before today, but I'd say that putting it in mp3dec is probably the wrong approach
[03:55:07 CEST] <kode54> is there a better place to put it so it's picked up by the stream data?
[03:55:44 CEST] <cone-941> ffmpeg 03Mark Reid 07master:0a088dea3d33: ffmpeg.c: copy video profile when using stream_copy
[03:57:57 CEST] <Chloe> Well the reason I think that is because iTunes' AAC also supports gapless (not just mp3)
[04:26:06 CEST] <Chloe> kode54: I'm not entirely sure where else it could go, I was just wondering how it could be added to AAC as well with minimal code duplication
[04:36:47 CEST] <kode54> Chloe, AAC/MP4 has a different iTunsMPB field format, though
[04:36:51 CEST] <kode54> slightly different, at least
[04:40:41 CEST] <Chloe> kode54: Oh, I didn't realise this. It should be fine to have an mp3 version then, the only issue is that there's no output in ffprobe I guess, and I think ffmpeg may already have aac gapless support. (ffprobe shows something with my sample). ugh, it's 3:40am, too late to be doing this. bye
[04:41:21 CEST] <kode54> see you tomorrow, or something :]
[04:41:34 CEST] <kode54> I need to fix this ID3v2 code too
[04:41:44 CEST] <Chloe> good luck :)
[04:53:10 CEST] <kode54> I compiled my ID3v2 patch and it definitely came out in the ffprobe, as far as the tags being displayed
[04:53:18 CEST] <kode54> now to test the gapless thing
[04:54:01 CEST] <kode54> comes out correct length
[04:54:37 CEST] <kode54> are you sure you used the copy you freshly built (./ffprobe) to test the patch?
[04:54:42 CEST] <kode54> you can recheck in the morning
[09:24:34 CEST] <rcombs> holy fuck DTS CD is the most idiotic thing I've ever seen
[09:24:51 CEST] <rcombs> the format itself, and the ridiculous hacks people have come up with to work with it
[09:25:05 CEST] <rcombs> "let's pretend DTS is PCM and put it on CDDAs"
[09:25:17 CEST] <rcombs> "wait won't that blow out people's speakers if they play it on CDDA players"
[09:25:35 CEST] <rcombs> "hmm good point, let's add 2 padding bits for every 14 bits of audio data"
[09:25:48 CEST] <rcombs> <later&>
[09:26:03 CEST] <rcombs> "let's rip DTS CDs as WAVs as if they were regular PCM CDDAs"
[09:26:36 CEST] <rcombs> "let's make players support reading that"
[09:26:39 CEST] <rcombs> <later&>
[09:27:30 CEST] <rcombs> "hmm I know how to add metadata tags to FLACs but not to WAVs& guess I'll pretend this WAV really does contain PCM and compress it with FLAC, then tag that"
[12:16:21 CEST] <Chloe> kode54: ok, yes it does work. I'm not sure what happened. I think I was just missing a make clean
[12:19:40 CEST] <cone-036> ffmpeg 03Chris Cunningham 07master:d59820f6fec3: libavformat/matroskadec: fix unsigned overflow to improve seeking
[12:19:40 CEST] <cone-036> ffmpeg 03Nikolay Aleksandrov 07master:8ec599caa4b8: MAINTAINERS: add myself to the project server
[14:06:38 CEST] <cone-036> ffmpeg 03Mark Reid 07master:e47981dab7fb: libavformat/movenc: add dnxhr compatibility for apple players
[14:06:39 CEST] <cone-036> ffmpeg 03Mark Reid 07master:c2cd1a5ea7a2: tests/fate/dnxhd: add dnxhr prefix tests
[15:30:55 CEST] <kasper93> Hi
[15:31:20 CEST] <kasper93> Any reason that ffio_init_context doesn't initialize protocol_whitelist field?
[15:31:35 CEST] <kasper93> (and protocol_blacklist for that matter)
[15:32:57 CEST] <cone-036> ffmpeg 03Mark Reid 07master:cdca1902ab35: libavcodec/dnxhdenc: add support for variable mircoblock counts
[15:32:58 CEST] <cone-036> ffmpeg 03Mark Reid 07master:ece12db4f1a5: libavcodec/dnxhdenc: add support for dnxhr encoding
[15:33:23 CEST] <kasper93> Right now rtp/asf crashes because of that.
[15:34:07 CEST] <JEEB> is that because of uninit data access or because nothing is whitelisted?
[15:34:24 CEST] <JEEB> although I guess crash means the previous one
[15:34:34 CEST] <JEEB> s/previous/former/
[15:34:44 CEST] <kasper93> uninit data access. 
[15:35:09 CEST] <kasper93> in utils.c:501 "if (!s->protocol_whitelist && s->pb && s->pb->protocol_whitelist) {"
[15:35:26 CEST] <kasper93> s->pb->protocol_whitelist is uninit.
[15:35:41 CEST] <kasper93> av_strdup crashes.
[15:58:32 CEST] <michaelni> kasper93, is there a ticket or testcase about this ?
[16:04:44 CEST] <kasper93> michaelni: I have regional locked stream that crashes. But looking at the code it will be reproducible with every asf stream. Take a look at rtpdec_asf.c:118. init_packetizer inits AVIOContext but the protocol_whitelist is not initialized. 
[16:05:18 CEST] <kasper93> I'm just not sure if it should be init in the rtpdec_asf.c or actually in ffio_init_context for everyone.
[16:37:43 CEST] <michaelni> kasper93, is whitelist ever initialized before ffio_init_context() so that it would wipe it out ?
[17:04:44 CEST] <michaelni> kasper93, you could post a patch that sets the lists in rtpdec_asf.c and point in the commit message out that you are unusure if a more generic solution would make sense
[17:05:16 CEST] <michaelni> aka implement the simpler and see if everyone is happy with that
[20:09:42 CEST] <cone-303> ffmpeg 03Jan Sebechlebsky 07master:75bd5d3e2dde: avformat/tee: Rescale ts using av_packet_rescale_ts
[22:54:58 CEST] <Compn> ubitux : what kind of rtmpdump abuse mails did you geT? 
[22:55:28 CEST] <Compn> or if you can, forward them to me compn at ffmpeg.org
[22:55:37 CEST] <Compn> if thats easy for you, otherwise no problem
[00:00:00 CEST] --- Sun Jul 24 2016