| 00:04.40 | coreyfarrell | I'm doing a test merge of media_formats-reviewed to trunk, no part of format.c was successful. but only 16 files left to manually merge, most don't have much to fix |
| 00:06.25 | mjordan | \o/ |
| 00:06.29 | mjordan | I have translation paths |
| 00:08.12 | file | mjordan, You have unlocked "I'm going to have to ask you to come in on Sunday, mmmk?" |
| 00:08.34 | mjordan | I asked myself |
| 00:08.44 | mjordan | And I told myself to !&*) off |
| 00:08.49 | mjordan | sometimes, I like to stick it to the man |
| 00:22.06 | mjordan | Hm. I'll feel better once we start getting audio flowing. |
| 00:23.12 | file | if Rome was built using IKEA furniture it would have been built in a day |
| 00:24.15 | coreyfarrell | and would have fallen the same day |
| 00:24.53 | file | coreyfarrell, efficiency |
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| 01:11.03 | mjordan | I think I've reached my stopping point - got everything to compile, and the first new error is ast_parse_allow_disallow not knowing what 'ulaw' is |
| 01:11.09 | mjordan | going to post this up for review |
| 01:14.33 | coreyfarrell | I'm attempting to compile trunk. got almost everything to merge. |
| 01:18.02 | coreyfarrell | instead of merging format.c I just copied from media_formats-reviewed, and it also failed to merge ast_translator_best_choice, so I'm probably just going to copy that procedure too see where it gets me |
| 01:18.47 | mjordan | hm |
| 01:18.57 | mjordan | I don't think I've touched that one yet |
| 01:19.47 | mjordan | coreyfarrell: cleaning things up right now. In channel.c, you removed an explicit check for the T140 format type (which called the write_text channel callback) and instead made everything call send_text unilaterally. Any particular reason why? |
| 01:20.22 | coreyfarrell | whatever was being checked for T140 was never set (grepped the entire asterisk source) |
| 01:23.33 | mjordan | hm |
| 01:23.47 | mjordan | that's odd. |
| 01:24.50 | coreyfarrell | that's what I thought. |
| 01:25.14 | mjordan | res_rtp_asterisk should be able to create frames with that format type |
| 01:26.32 | coreyfarrell | set fr.subclass.integer ? |
| 01:28.39 | mjordan | line 3875 |
| 01:28.46 | mjordan | ast_format_set(&rtp->f.subclass.format, AST_FORMAT_T140, 0); |
| 01:29.10 | coreyfarrell | how would that set fr.subclass.integer ? |
| 01:29.38 | coreyfarrell | maybe the check I removed should exist but never worked? |
| 01:30.24 | mjordan | it wouldn't. |
| 01:30.27 | mjordan | Well |
| 01:30.29 | mjordan | It probably used to work |
| 01:30.38 | mjordan | But real time text isn't something that anyone ever tests |
| 01:30.52 | mjordan | And the lack of anything approximating an automated test for it means it gets 0 love |
| 01:31.04 | mjordan | I'll add it back, just because it does match code in other locations as well |
| 01:31.11 | mjordan | as far as determining which channel tech callback gets used |
| 01:31.22 | mjordan | but I suspect something in the RTP layer doesn't build the frames correctly |
| 01:33.38 | coreyfarrell | oh any objection if I create /team/group/media_formats-reviewed-trunk ? |
| 01:34.38 | mjordan | none |
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| 01:37.46 | coreyfarrell | great. I'm probably not going to do automerge at this point, I'm just going do the svn copy then commit what I've merged |
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| 01:53.40 | TazzNZ | so guys - if I want to add configuration options and/or channel vars to Asterisk, where should I look for naming stds ? |
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| 01:57.55 | mjordan | TazzNZ: generally, that depends on the channel driver. We tend to use snake_case for variables/configuration parameters |
| 01:58.28 | mjordan | TazzNZ: Channel variables are typically all upper case, and again, should fit the nomenclature of the channel driver/module they are being added to. It's nice if the name chosen doesn't conflict with someone's possible usage of the same name |
| 01:58.42 | mjordan | i.e., adding "TRUNK" would probably be a bad idea |
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| 02:24.54 | mjordan | hm |
| 02:25.00 | mjordan | ast_parse_allow_disallow had an easy fix |
| 02:25.05 | mjordan | and now I have sip peers with codecs |
| 02:33.44 | coreyfarrell | meda_formats-reviewed-trunk now contains all reviewed code merged to trunk |
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| 07:33.58 | skrusty | morning |
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| 07:57.44 | oej | Morning folks |
| 08:01.27 | TazzNZ | hey oej, skrusty |
| 08:04.31 | skrusty | good w/e? |
| 08:04.47 | TazzNZ | yeah - wasn't too bad thanks :) yours ? |
| 08:05.28 | skrusty | warm which is good for the UK ;) normally wet :) |
| 08:05.59 | TazzNZ | ah - I was wondering about your TZ - most ppl in here are US based :) |
| 08:11.54 | TazzNZ | anybody got an idea how big the svn is for asterisk ? |
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| 08:23.10 | skrusty | yeah, it's quiet here in the mornings :) |
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| 12:31.16 | file | Just SIP It (tm) |
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| 15:36.40 | tzafrir | makeopts:AST_MARCH_NATIVE=@AST_MARCH_NATIVE@ - what's this? |
| 15:48.54 | vlad_starkov | malcolmd: Hello Malcolm, could you please point me how to can I use "audiofilter" utility to tune my custom audio prompts as you showed here https://wiki.asterisk.org/wiki/display/AST/About+the+Sounds+Tools ? Thanks |
| 15:49.48 | file | tzafrir, it's for -march=native support in GCC, the configure script checks for support for it |
| 15:50.59 | tzafrir | AC_SUBST(AST_NATIVE_ARCH) |
| 15:51.26 | tzafrir | Nothing seems to use AST_MARCH_NATIVE |
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| 16:07.25 | tzafrir | hmm... when building in dev mod (specifically: COMPILE_DOUBLE), OPTIMIZE is always added to build flags, |
| 16:07.32 | tzafrir | regardless of DONT_OPTIMIZE |
| 16:08.02 | file | if I remember right kpfleming made it so it builds two - an optimized version and an unoptimized version |
| 16:08.13 | file | as optimization caught stuff |
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| 16:55.34 | TazzNZ | Hey guys - so a bit more details on my question "yesterday" about the vars - I am looking to add RFC4904 support to Asterisk. From what I can see it can't be to bad. I am thinking om implementing it using channel vars so that it can be adjusted based on the call. How do I bouch that off "the powers that be" so that I don't run off on the wrong track ? |
| 16:56.10 | TazzNZ | channel vars==SIP channels only :) |
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| 17:11.14 | malcolmd | so, audiofilter's something that was provided by the author of speex - i can't tell you a lot about what it's doing. you can see an example of its use in the converters script in the scripts sub-directory of sound_tools. |
| 17:24.13 | coreyfarrell | TazzNZ: it sounds like you want to add a new feature, so if you don't know testsuite check out https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation (and sub pages). your chances of adding a new feature to a channel driver is greatly increased if you provide automated testing. |
| 17:33.30 | TazzNZ | cool - will do coreyfarrell |
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| 17:40.36 | coreyfarrell | TazzNZ: you should submit patches for a couple bugs in asterisk if you have not already. this is not mandatory but I think will help you understand the process / coding guidelines / etc. that way when you have your new feature you will not have to learn those things. |
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| 18:14.32 | TazzNZ | coreyfarrell: I will try |
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| 18:37.39 | file | newtonr, can you see if you can find a JIRA issue talking about the "rtp set debug on" output for sending packets having the incorrect "len" when ICE is in use? |
| 18:43.30 | file | thinks he went for a walk |
| 18:47.47 | file | there it is |
| 19:03.12 | newtonr | file, just now saw your message. you already found it? |
| 19:03.17 | file | yes |
| 19:03.39 | newtonr | yeah i went for a walk. it is crazy hot out |
| 19:04.02 | newtonr | and now, for the rest of my lunch, food |
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| 20:35.34 | mjordan | coreyfarrell: sorry, got way laid today by meetings. I think I've got all the findings now on r3665 |
| 20:38.33 | coreyfarrell | mjordan: it's cool. taking a look now |
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| 20:42.11 | mjordan | thanks! |
| 20:43.15 | mjordan | from a "what's next" perspective, would you rather have me go ahead with the renaming, or save that for later? A part of me kind of wants to hold off for a little bit until we get chan_sip/pjsip/iax2/others off the ground, as the renaming will definitely hold things up until it is approved |
| 20:43.51 | coreyfarrell | mjordan: only one issue I might see |
| 20:46.42 | coreyfarrell | I think we should go for res_rtp_asterisk next. once we have that and one channel driver we will be able to start doing some testing |
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| 20:57.14 | coreyfarrell | mjordan: on my last finding one other way to deal with it would be to just zero the ast_rtp_mime_types[x] when adding one |
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| 21:41.27 | coreyfarrell | mjordan / file: how do you feel about moving forward with media_formats based on trunk? I think we're too late to consider ever merging to /branches/12. |
| 21:41.49 | mjordan | I'm fine with that |
| 21:42.12 | mjordan | we did it against 12 in case it was a candidate for that branch, but I think we both realized awhile back that it was a lost cause :-) |
| 21:43.00 | mjordan | coreyfarrell: do you want to merge the changes into your trunk, and then we start working out of that one? |
| 21:43.06 | coreyfarrell | also related question did I do right with the creation of /team/group/media_formats-reviewed-trunk? I'm more used to working with GIT, so it feels a bit dirty that the merge of code from media_formats-reviewed doesn't properly account for the source of the original commits |
| 21:43.13 | mjordan | heh |
| 21:43.18 | mjordan | it is dirty |
| 21:44.50 | coreyfarrell | I mean if that's the crappy way things are done with SVN I'm ok with it |
| 21:46.04 | coreyfarrell | I ran svn diff on media_formats-reviewed that included all changes since it was branched from 12, then ran 'svn patch' on my new team branch. then manually resolved the issues |
| 21:47.29 | coreyfarrell | but to answer your question, if I did an acceptable job merging the existing reviewed changes into the new branch, I'd prefer we stop using the branch based on 12 |
| 22:05.59 | mjordan | coreyfarrell: works for me. |
| 22:06.16 | mjordan | And yes, that's pretty much what happens. When this behemoth gets merged in, you do lose the commit messages from the team branch |
| 22:06.32 | mjordan | Hence: we need to move to git. |
| 22:06.41 | mjordan | But after 7/16, cus, ya know, feature freeze. |
| 22:09.18 | coreyfarrell | mjordan: unrelated question, what are the chances of getting a bamboo job going that does Asterisk Testsuite full and Unit tests in the same job? The purpose would be to get coverage information for all tests. even once a week would be useful to identify code that is never tested |
| 22:10.01 | mjordan | coreyfarrell: highly likely |
| 22:10.17 | mjordan | https://bamboo.asterisk.org/bamboo/browse/ATB-MRB |
| 22:10.22 | mjordan | I just have re-enable it |
| 22:10.30 | mjordan | The only problem right now is that it will still try to build all modules |
| 22:10.55 | mjordan | er |
| 22:10.56 | mjordan | AH |
| 22:11.02 | mjordan | you mean a general test suite run that does everything |
| 22:11.24 | coreyfarrell | yes normal testing of 1.8/11/12/13/trunk |
| 22:11.26 | mjordan | Well, we have one problem right now, which is that res_mwi_external (and things that use it) conflict with app_voicemail |
| 22:11.37 | mjordan | so no matter what, there won't be code coverage for those items |
| 22:12.09 | coreyfarrell | so for example adding unit tests to run from https://bamboo.asterisk.org/bamboo/browse/AST-ATSF |
| 22:12.16 | mjordan | Yeah, that's doable |
| 22:12.27 | mjordan | And unit tests don't add much time, so we might as well just add them to those test runs |
| 22:12.45 | coreyfarrell | well from what I saw the full testsuite is 400 minutes, unit tests are 200 |
| 22:12.58 | coreyfarrell | maybe I just looked at a slow run of them |
| 22:13.16 | mjordan | well... those times are an illusion, since they include how long they were blocked |
| 22:13.22 | mjordan | plus, they have to rebuild Asterisk on each run |
| 22:13.28 | mjordan | so a lot of that time is compilation (for the unit tests) |
| 22:13.41 | mjordan | the actual unit tests probably take less than a minute |
| 22:13.47 | coreyfarrell | ah yes I'm looking at unit tests again it says 37 minutes so that's much better |
| 22:15.05 | coreyfarrell | mjordan: did you forget to switch branches or are you committing to both? |
| 22:16.02 | mjordan | hm |
| 22:16.27 | mjordan | wasn't sure if you were going to merge it over or me :-) But I'd be happy to merge it over |
| 22:17.14 | coreyfarrell | makes no difference to me |
| 22:17.19 | mjordan | I got it then |
| 22:17.33 | mjordan | file: I suggest we delete the old branch after this |
| 22:17.43 | file | I'm down with that |
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| 22:24.20 | coreyfarrell | The trunk based branch has a couple of build error's due to code that was added to trunk, I'll try to look through those later and get a review up to fix them |
| 22:28.01 | mjordan | I saw func_talkdetect, but that's not surprising since that was added recently in 12 |
| 22:28.16 | mjordan | and now I hit the ones in channel.c |
| 22:28.22 | mjordan | I'm going to assume those are new and commit the merge |
| 22:30.59 | coreyfarrell | please do |
| 22:31.15 | coreyfarrell | well you dealt with the merge conflicts? |
| 22:31.26 | coreyfarrell | mjordan: ^^ |
| 22:31.26 | mjordan | I dealt with the merge conflicts from my patch, yes |
| 22:31.38 | mjordan | there are compilation errors that are new in channel.c that weren't from merge conflicts |
| 22:31.45 | coreyfarrell | commit away |
| 22:32.06 | mjordan | should be in |
| 22:32.20 | mjordan | Unless there are any objections, I shall now delete the old branch ... |
| 22:33.03 | coreyfarrell | it's SVN, so if we really needed to we can still check-out the revision before you deleted so I saw burn it |
| 22:33.03 | mjordan | 3 |
| 22:33.04 | mjordan | 2 |
| 22:33.05 | mjordan | 1 |
| 22:33.07 | mjordan | yup :-) |
| 22:33.37 | mjordan | okay! |
| 22:34.12 | TazzNZ | blast off ? :D |
| 22:34.16 | mjordan | So: res_rtp_asterisk. It would probably get messy to have two people working on it. I'm more than happy to tackle it, unless you're itching for it coreyfarrell |
| 22:34.22 | mjordan | TazzNZ: something like that :-) |
| 22:34.33 | file | burn baby burn burn burn |
| 22:34.47 | TazzNZ | "How can we sleep when our SVN is burning...." |
| 22:34.54 | coreyfarrell | mjordan: I would be thrilled to not work on res_rtp_asterisk :) |
| 22:35.27 | mjordan | k! I'm going to be out of pocket most of tomorrow - doctor's visits in Nashville (yay) - but I'll be working on it while sitting in waiting rooms |
| 22:35.36 | mjordan | I'll shoot for getting something up for review by Wednesday |
| 22:35.55 | mjordan | I know chan_iax2 won't load currently |
| 22:35.58 | mjordan | linkage issues |
| 22:36.01 | coreyfarrell | sounds great. i'll go after any loose ends that resulted from the merge |
| 22:36.20 | coreyfarrell | ok well anything that's not chan_iax2 |
| 22:36.25 | mjordan | lol |
| 22:36.34 | mjordan | I'd be happy to look at that after res_rtp_asterisk |
| 22:36.41 | mjordan | after this week, we can probably put some more Digium folks behind whatever loose ends are left as well. |
| 22:37.48 | mjordan | is already curious why we hard coded the rate of G722 - and G722 only - as 8khz |
| 22:37.57 | coreyfarrell | once we have res_rtp_asterisk my goal is to be able to run this with chan_sip and call into the system with dialplan: Answer(); Playback(invalid); Hangup(); |
| 22:38.52 | coreyfarrell | I choose chan_sip since it's the driver I'm most familiar with. |
| 22:39.31 | file | mjordan, because SDP |
| 22:40.26 | mjordan | so, sounds like I'm holding things up :-D |
| 22:40.49 | coreyfarrell | mjordan: ? |
| 22:40.51 | mjordan | file: SDP expects g722 to have a format rate of 8khz? Or were you just trolling SDP? |
| 22:40.59 | mjordan | coreyfarrell: you're waiting on res_rtp_asterisk |
| 22:41.01 | file | no, it's 8000 in the SDP |
| 22:41.04 | file | despite being 16000 |
| 22:41.13 | mjordan | file: ... |
| 22:41.27 | mjordan | #troll |
| 22:41.54 | coreyfarrell | I can't blame you for needed 2 days to work on a module that I don't want to touch :) |
| 22:41.59 | file | I wish I were kidding |
| 22:42.13 | file | G.722 VoIP is typically carried in RTP payload type 9.[4] Note that IANA records the clock rate for type 9 G.722 as 8 kHz (instead of 16 kHz), RFC3551[5] clarifies that this is due to a historical error and is retained in order to maintain backward compatibility. Consequently correct implementations represent the value 8,000 where required but encode and decode audio at 16 kHz. |
| 22:43.01 | coreyfarrell | compatibility sucks |
| 22:43.43 | mjordan | see: IAX2. |
| 22:43.47 | mjordan | AND ITS BITFIELDS. |
| 22:43.49 | mjordan | WHY. |
| 22:44.00 | mjordan | anywho, off to dinner |
| 22:44.16 | file | and then we lose mjordan for eternity |
| 22:45.14 | TazzNZ | does he eat *that* much ? :) |
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| 23:22.31 | coreyfarrell | file: so is ast_format_cap locking gone? I'm updating test_format_api.c, it does tests with and without locking |
| 23:22.43 | file | yes |
| 23:23.05 | file | unless we come up with a good reason to need it again |
| 23:23.43 | coreyfarrell | ok so when we test with and without locking I should get rid of the test with locking? |
| 23:23.54 | file | yes |
| 23:23.59 | coreyfarrell | k, thanks |
| 23:24.21 | file | hrm, almost 8:30PM and it's still 23C out |
| 23:24.28 | file | I don't know how I feel about this |
| 23:25.40 | TazzNZ | file: I was in NV a month ago....topped out at 108 |
| 23:25.46 | TazzNZ | I almost died |
| 23:26.16 | Juggie | file, its summer? |
| 23:26.39 | Juggie | its 7:30 and 28c here |
| 23:26.49 | file | Juggie, this is the first night it's really been like this :P |
| 23:26.58 | Juggie | ah ic |
| 23:27.01 | Juggie | no air conditioning? |
| 23:27.14 | file | I have AC but that doesn't help me when I'm outside |
| 23:27.16 | TazzNZ | that is why he hasn't left work :D |
| 23:27.18 | TazzNZ | hehe |
| 23:27.25 | file | TazzNZ, I work from home |
| 23:27.25 | Juggie | heat wave coming this weekend too |
| 23:27.35 | TazzNZ | nice file |
| 23:27.39 | file | er, well, that's a lie now these days... I work from home and from an office downtown |
| 23:28.40 | file | but work time is over |
| 23:28.53 | TazzNZ | GAME TIME !!!! |
| 23:28.55 | TazzNZ | :D |
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| 23:44.37 | *** mode/#asterisk-dev [+o snuff-work] by ChanServ |