- From: Brent Shambaugh <brent.shambaugh@gmail.com>
- Date: Mon, 12 Aug 2019 15:19:27 -0500
- To: rhiaro <amy@rhiaro.co.uk>
- Cc: public-credentials@w3.org
I have started looking at this. Would it be good if I was on the call tomorrow? I might get some time off of work. Or I could wait. Sent from my iPhone > On Aug 10, 2019, at 4:56 AM, rhiaro <amy@rhiaro.co.uk> wrote: > > One thing I noticed in the instructions was "Look for any find/replace > suggestions in irc.log and update them (s/../..)" - the w3c scribe.perl > script (which is what RRSAgent uses to publish the static html straight > to w3.org for WGs) does this automatically. Perhaps this can be added to > the online log->html converter. I imagine that would save a lot of > cleanup time. After that it's just a quick skim for typos and obvious > non-meeting related chatter. And fixing anywhere the scribe syntax was > not used correctly... these last things I don't see how could be > automated until we have more advanced AI. > > Amy > >> On 10.8.19. 03:37, Kim Hamilton wrote: >> Thanks for offering Brent! >> >> Overall goal: reduce the tedium of minute generation after the calls. >> It's a burden Manu and team had been carrying for a long time, and >> then I misguidedly volunteered to take over. :) I think it takes about >> 20-30 minutes each week -- just enough time to be a pain given other >> obligations. >> >> I documented the process here (https://w3c-ccg.github.io/publish.html) >> -- also note the link to the video demonstration. Recently I've been >> thinking about ways to automate things. For example, at first I >> assumed the need to edit the audio was a given, and so I started going >> down a path (auto-wave detection and trimming) that made me suspicious >> I needed to learn the bigger picture. The voipbot code and asterisk >> libraries addressed the gap in my understanding, and now I see what >> sort of changes we might make. >> >> The tedious parts I see are: >> *1. Trimming dead audio* >> The need to trim dead audio from beginning and end. We do this >> manually in an audio editor. The need to even install such an app is >> (IMO) a source of friction that prevents people from even wanting to help. >> >> Fortunately there's a better solution. Asterisk exposes "start"/"stop" >> recording commands. I'm thinking we could add a voipbot command so >> that the chair can start recording (or if we're worried that's >> something we'll forget, we could tie it to another trigger that we use >> around the start of the meeting. >> >> *2. Minute cleanup* >> I don't have a good proposal for this yet. This part involves editing >> the irc.log text files. The instructions to do this (currently) are: >> >> * Go to the online scribe tool and copy/paste irc.log into the text >> input box at the bottom. >> * Clean up the IRC log accordingly and overwrite irc.log with the >> edited file. >> * Things to check >> o Look for any find/replace suggestions in irc.log and update >> them (s/../..) >> o Ensure (people name) aliases have matches (there is a >> people.json file in the publishing repo) >> >> But there's another problematic part, which is ensuring everyone on >> the call has been accounting for. This is currently an error-prone >> step because it's up to the publisher to (informally) remember to >> which people joined in and add them. >> >> Progress on 1 or 2 would be great. Manu, I'm curious if you have any >> objections to explicit start/stop recording -- perhaps there's a >> historical reason for that. >> >> As for the existing minutes, I think we have no choice but to suck it >> up and publish them using the existing means. But working on these >> improvements for future recordings would tremendously helpful. >> >> Thanks, >> Kim >> >> >> >> >> On Fri, Aug 9, 2019 at 2:48 PM Brent Shambaugh >> <brent.shambaugh@gmail.com <mailto:brent.shambaugh@gmail.com>> wrote: >> >> Is there some way I can help? What is the goal? >> >> I do see that >> https://github.com/digitalbazaar/voipbot/blob/master/index.js has >> a lot of comments in it already. This could be the first place to >> look. >> >> Running grep "//[A-Za-z0-9]" on this gives: >> >> // read the config file >> // create a lockfile that goes stale after 130 minutes >> // cleanup the lockfile if there is an uncaught exception or an >> interrupt >> // shared variables >> // check to see if there is a channel password >> // connect to the IRC channel >> // check if TLS should be used to connect >> // check to see if the server requires a password >> // says the given message in the main irc channel >> // hook up an error handler to prevent exit on error >> // handle channel join event >> // connect to the Asterisk server >> // get a list of conference participants on join >> // announce when people join the conference >> // confbridge talking the recording file >> // current time in seconds since epoch >> // clear audio events older than 3 minutes >> // build the list of participants >> // announce when people leave the conference >> // listen to IRC channel messages >> // log IRC messages to logfile if a recording directory is >> specified >> // log, ignoring all errors >> // handle non-voipbot directed channel commands >> // handle voipbot specific commands >> // all commands must contain at least the command name which is >> // the second argument >> // show list of participants >> // current time in seconds since epoch >> // overwrite calleridname >> // must have at least two characters to attempt a guess >> // guess by attempting to match the end of the channel name >> // upload log files if they exist >> // search a directory for a file matching regexp larger than fsize >> bytes >> // return early if upload settings are not set >> // get latest audio file larger than 15MB >> // get latest audio and IRC logs >> s3ForcePathStyle: true, // needed with minio? >> // build S3 parameters >> // build S3 parameters >> >> other: >> >> /*************************** Helper functions ******************************/ >> >> /** >> * Converts a queue to a string. >> * >> * @param queue the queue to convert to a string >> * @return a string representing the queue >> */ >> >> /** >> * Removes a given nick from a queue. >> * >> * @param queue the queue to modify. >> * @param nick the nickname to remove from the queue >> * >> * @return the removed value. >> */ >> >> /** >> * Pretty-prints a channel name for human readability. >> * >> * @param channel the channel name >> * @return the pretty-printed channel >> */ >> >> /** >> * Attempts to guess a channel given some text. The test currently tries >> * to match the last section of the channel name. >> * >> * @param text the text to try and match against the end of the channel name. >> * @return the guessed channel name, or false if the guess failed. >> */ >> >> /** >> * Removes a given nick from a queue. >> * >> * @param queue the queue to modify. >> * @param nick the nickname to remove from the queue >> * >> * @return the removed value. >> */ >> >> // search a directory for a file matching regexp larger than fsize bytes >> >> >> It looks like .wav files are handled in the program. I see no >> mention of .mp4. I guess the conversion program from .wav to .mp4 >> elsewhere? >> It looks like the script works with https://aws.amazon.com/s3/ . >> >> At least this documentation might be functionally equivalent. >> https://www.voip-info.org/asterisk-manager-api/ >> The wiki might be useful, but it is difficult to find a starting >> point. >> https://wiki.asterisk.org/wiki/display/AST/Beginning+Asterisk >> >> -Brent Shambaugh >> >> GitHub: https://github.com/bshambaugh >> Website: http://bshambaugh.org/ >> LinkedIN: https://www.linkedin.com/in/brent-shambaugh-9b91259 >> Skype: brent.shambaugh >> Twitter: https://twitter.com/Brent_Shambaugh >> WebID: http://bshambaugh.org/foaf.rdf#me >> >> >> On Fri, Aug 9, 2019 at 2:08 PM Kim Hamilton >> <kimdhamilton@gmail.com <mailto:kimdhamilton@gmail.com>> wrote: >> >> Excellent thanks! I’ll take a look >> >> On Fri, Aug 9, 2019 at 10:09 AM Manu Sporny >> <msporny@digitalbazaar.com <mailto:msporny@digitalbazaar.com>> >> wrote: >> >>> On 8/9/19 1:05 AM, Kim Hamilton wrote: >>> Where is the info/source for voipbot? >> >> https://github.com/digitalbazaar/voipbot >> >> Documentation is horrible, integration into Asterisk is beyond >> painful... it's a volunteer project built over the last >> decade in our >> spare time. :) >> >> That said, it's the heart and soul of the system that >> manages the phone >> lines, audio recording, log publishing, and queue >> management for our >> weekly CCG calls. We'd be happy to deploy upgrades if >> anyone would like >> to contribute new features, bug fixes, etc. >> >> -- manu >> >> -- >> Manu Sporny (skype: msporny, twitter: manusporny) >> Founder/CEO - Digital Bazaar, Inc. >> blog: Veres One Decentralized Identifier Blockchain Launches >> https://tinyurl.com/veres-one-launches >> >
Received on Monday, 12 August 2019 20:19:53 UTC