Babble | VoIP client/server inspired by the Teamspeak project | TCP library

You can't do this.

There is no way to keep the microphone open with a custom skill.

When you activate the skill "open my skill", Alexa starts talking and then, when she stops, you have only 8 + 8 seconds to speak, otherwise, the session will be closed.

So you have 8 seconds, then if you don't speak there is a re-prompt phrase asking you to tell something, and another 8 seconds where you can speak.

If you don't, the session will be closed.

There is a way to keep the session open more than 8 seconds (playing some mute music for example), but the problem remains because you can only speak after that "music" is finished.

document.getElementsByClassName("flip-card") will return an DOM Array elements, just add [0] to select the first element of the array, like this:

Use MID to parse the string breaking on the Char(10) and return the max

In your case you needs to create a recursive function which will get the nested value of underlying interface{} till last depth. There is no way we can get nested value from interface{} with unknown underlying type. Here is an example of what you can do in your code

The str.replace method returns a copy of the original string with all occurrences of the given substring replaced with the given new string, so you can simply slice the input string from the second character and replace all occurrences of the first character with '*', and concatenate it after the first character of the orignal string:

Code is complicated but I found few mistakes which resolved problem with error.

bestWordHelper() needs 2D list but problem is [] in some places and + between list's elements so sometimes it creates 1D list instead of 2D list and then it can't get second dimention when you use L[0][1] - and finally you get integer[0] instead of list[0] so you get your error message "int object is not subscriptable"

First: you forgot [1] in

gst-launch-1.0 ximagesrc xid=$XID ! video/x-raw,framerate=30/1 ! queue ! videoconvert ! videorate ! queue ! x264enc ! queue ! avimux name=mux ! queue ! filesink location=out.avi pulsesrc device=$DEV ! queue ! audioconvert ! queue ! lamemp3enc bitrate=192 ! queue ! mux.

The above pipeline should play the audio video at proper speed.

I would also appreciate if you correct me on the usage of ! queue !. Where is it needed? In the current setup I almost never get warnings that samples were dropped.

queue are just buffers, these need to be used in places where one element is slower and one is faster, so for example video generating (ximagesrc) is much faster compared with x264enc (software encoding), so you would add a queue in between them so as to buffers aren’t dropped.

gst-launch-1.0 ximagesrc ! video/x-raw,framerate=30/1 ! queue ! videoconvert ! queue ! x264enc key-int-max=5 ! queue ! mp4mux name=mux reserved-bytes-per-sec=100 reserved-max-duration=20184000000000 reserved-moov-update-period=100000000 ! queue ! filesink location=out.mp4 audiotestsrc ! queue ! audioconvert ! queue ! lamemp3enc bitrate=192 ! queue ! mux.

The above pipeline would create a mp4 file mp4mux, but the moov atom will be at end itself also note make sure you change mp4mux properties as per your need.

Check your data for lines that are empty ... you got empty lines in it (maybe the last line has got a \n at its end?) resulting in raise JSONDecodeError("Expecting value", s, err.value) from None.

This fixes that error:

In python strings are iterables. So, your line for i in s will run n times, one for every letter in the string, and will print "not found". The first time the loop runs, it replaces a with*. So, for all subsequent runs, it will print "not found".

If I understand what you are trying to do, it would be something like