W3C home > Mailing lists > Public > public-credentials@w3.org > August 2019

Re: Regarding Text to HTML conversion

From: rhiaro <amy@rhiaro.co.uk>
Date: Sat, 10 Aug 2019 11:56:36 +0200
To: public-credentials@w3.org
Message-ID: <cf37e428-6af6-3503-8ed3-cb704c5e9221@rhiaro.co.uk>
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 Saturday, 10 August 2019 09:57:10 UTC

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