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

Re: Regarding Text to HTML conversion

From: Brent Shambaugh <brent.shambaugh@gmail.com>
Date: Mon, 12 Aug 2019 15:19:27 -0500
Cc: public-credentials@w3.org
Message-Id: <CA9D38D6-CC26-4440-9A26-9B122007564F@gmail.com>
To: rhiaro <amy@rhiaro.co.uk>
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

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