trickle | A small library for composing asynchronous code | Chat library
kandi X-RAY | trickle Summary
kandi X-RAY | trickle Summary
We’re using Trickle internally at Spotify in core, production-critical services that would break Spotify completely if they failed. This means we have a fairly high degree of confidence that it works. It is, however, a young library and you shouldn’t be surprised if there are API changes in the next few months.
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Top functions reviewed by kandi - BETA
- The main entry point
- Returns the list of call information
- Makes a simple graph of the name and a hello node
- String representation of the call
- Gets the parameters
- Returns the value of the property
- Runs a prepared graph
- Executes the plan
- Declares the graph variables for sending
- Creates a graph from the supplied function
- Gets the future for the traversal
- Returns a String representation of this attribute
- This method builds a hashCode value
- Adds debugging information to the graph
- Returns the list of arguments of this node
- Constructs a new error node
- Gets the bind value associated with the input
- This method returns a hashCode of the variable name
- Compares against input
- Compares this parameter with another object
- Returns a future that can be used asynchronously
- Compares two call info objects
- Returns the name of this model
- Returns the predecessors of this graph
- Returns the name of the graph
- Returns a function that returns an AsyncFunction
trickle Key Features
trickle Examples and Code Snippets
Community Discussions
Trending Discussions on trickle
QUESTION
Am new to webRTC and am trying to create a react native app with video calling functionality using this tutorial here as an example to follow https://dipanshkhandelwal.medium.com/video-calling-using-firebase-and-webrtc-14cc2d4afceb
However i keep getting this error on iOS and on android the app just closes once i try to join a call. The error i get on iOS says:
...ANSWER
Answered 2021-Jun-05 at 06:38I guess you are trying to use firebase as a signalling medium and want to use react-native-webrtc for the video calling.
Here is the sample code I have for the same solution with the latest libraries and react-native version.
Firebase Installation React Native.
Just set up ios and android using this above link and then use the below code for reference.
QUESTION
In WebRTC, 'offer' must be generated by initiating peer (i.e. the caller) and shared with other peer (i.e. the callee) via signaling channel.
The other peer, of she wishes to answer, is supposed to generate 'answer' and send it back to initiating peer. Again, via signaling channel.
In this exchange it is clear who should be preparing which piece of data, and to whom it should be shared.
The next part of the ritual is the gathering of ICE candidates. Both peers will gather their own candidate slowly (trickle ICE) and on each discovered candidate, that candidate should be shared too. Sharing is also to be done via signaling channel
What is not clear to me here is at this point, who should be sharing candidates to whom?
Should caller share with callee? Or the other way around? Or both
...ANSWER
Answered 2021-May-25 at 16:30The ICE candidates are shared with the remote peer. If you think in terms of Offer/Answer
- The offerer will send their ICE Candidates to the answerer
- The answerer will send their ICE Candidates to the offerer
So the answer to your final question would be both
!
QUESTION
I have a web.api action that accepts multipart/form-data via streaming, so it doesn't have any arguments:
...ANSWER
Answered 2021-May-21 at 10:21Since in .Net web api the controllers derive from ControllerBase
, they all have a HttpContext
instance property, you can use its RequestAborted
token.
corresponding docs
So something like this:
QUESTION
I'm working on WebRTC streaming which streams video from a device to a browser. This streaming works in Chromium browsers just fine but fails in Firefox. There is a failure with the SDP exchange which then halts the rest of the connection (no ICE candidates sent after SDP exchange).
There are some issues with Firefox's answer SDP I've found but I haven't discovered a reason for the issues: SDP mentions VP8 but we use H264 only; m=video 0 has port 0 but typically that's non zero; I typically get an a=inactive line; a=sendrecv should probably be a=recvonly; many other lines are missing (for example, ICE-specific lines)
SDP examples below:
...ANSWER
Answered 2021-May-13 at 18:03Firefox likely doesn't support the profile level id 0x4d4016. Then you have no codecs in common and the media is rejected (which is what port 0 means). Without any non-rejected m-line your connection will fail.
QUESTION
I have a web app that plays video broadcasted using WebRTC. I'm using a COTURN simple implementation (as described here: https://gabrieltanner.org/blog/turn-server). My turnserver.conf looks like this:
...ANSWER
Answered 2021-Apr-04 at 22:56I'm not sure if it was some sort of update on the RTCPeerConnection constructor (because it just stopped working without anyone touching the code), but in case anyone needs this, I solved it by adding
QUESTION
"Answer" of SDP offer is not sent to Web App(running on Windows/Mac/Linux) from Android/iOS application after updation of Chrome to latest v89 (released on 9 March 2021).
It is working fine on Chrome v88 and below.
Offer SDP:
...ANSWER
Answered 2021-Mar-22 at 15:14The issue is caused by Chrome enabling extmap-allow-mixed
by default (see https://www.chromestatus.com/features#offerExtmapAllowMixed). It can be solved by the receiver mangling the offer SDP to remove the line containing a=extmap-allow-mixed
QUESTION
I am writing an Angular front end for an API that occasionally serves Infinity
and -Infinity
(as bare words) in the JSON response object. This is of course not compliant with the spec, but is handled a few different JSON libraries, albeit optionally. I have an Angular service in place that can successfully retrieve and handle any retrieved entity that does not have these non-conforming values. Additionally, I have managed to get an HttpInterceptor
in place which just logs when events trickle through, just to be sure I have it connected properly.
The issue that I am facing is that the HttpInterceptor
seems to allow me to do one of two things:
- Catch/mutate the request before it is sent to the API, or
- Catch/mutate the request after it comes back from the API, and also after it is parsed.
What I would like to do is very similar to this question for native javascript, but I have not been able to determine if it is possible to tie into the replacer
function of JSON.parse in the Angular Observable pipe (I think that if tying into that is possible it would solve my issue).
I have also found this question for Angular which is close, but they appear to have been able to handle changing the response to something other than the bare-words, which I don't have the liberty of doing.
This is the current implementation of my HttpInterceptor
, note that it does not actually make any changes to the body. When retrieving an entity without these bare-word values, it logs to the console and all is well. When retrieving an entity with any of these bare-word values, an error is thrown before the HERE
line is hit.
ANSWER
Answered 2021-Mar-14 at 20:36I was able to figure out how to achieve this, and it came down to:
- Modifying the request to return as text instead of json
- Catch the text response and replace the bare word symbols with specific string flags
- Parse the text into an object using
JSON.parse
, providing areviver
function to replace the specific string flags with the javascript version of+/-Infinity
andNaN
Here's the Angular HttpInterceptor
I came up with:
QUESTION
I want to set up a WebRTC datachannel and send some text. Therefore I create a RTCPeerConnection object with an "onicecandidate" and "ondatachannel" handler. (I keep "configuration" for RTCPeerConnection empty because connection is between localhost only.)
...ANSWER
Answered 2021-Feb-23 at 03:01You must call myConnection.createDataChannel
at least once before negotiating, to have the connection negotiate the use of datachannels. They're not included by default.
There's no media and no datachannel before then, nothing to negotiate. This makes your first negotiation redundant, a no-op, which is why the state never leave "new"
.
Once you've done this, there's no need to do it again, as all subsequent datachannels are multiplexed over the same sctp transport.
To have negotiation only happen when needed, consider setting up:
QUESTION
I am using a recursive CTE for "Groups" in a table using ID and Parent.
The first table "Groups" (Project_Estimate_Group) is hierarchical with ID and parent. The second table "GroupOptions" (Project_Estimate_Group_Option) accommodates a many-to-many relationship between Groups and Options. If you add an Option to any level it trickles down to any child in that branch.
I need to join the parent and recursive children of Groups to GroupOptions and have those joins "trickle" down through each recursion level so that at each level you have a join to the current Options as well as a join to the Options for each parent in the branch.
I need a row for each group without the OptionID and a row for each joined OptionID. Some levels will have no Options.
Can this be done with a recursive CTE?
Table1 "Groups" has
...ANSWER
Answered 2021-Feb-20 at 10:49Try this:
QUESTION
I want to deploy an RDS database to AWS with a secret from AWS Secrets Manager. I have:
...ANSWER
Answered 2021-Feb-16 at 18:57Define an output in the secrets
module. Define an input in the db
module. Pass the output value from secrets
to the input property in db
.
For example if you defined an output named "password" in secrets
and an input named "password" in db
, then in your db module declaration you would pass the value like this:
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