Best practices for internal api calls to external apis with buffer - python-3.x

I have different external APIs doing basically the same things but in a different way : add product informations (ext_api).
I would like to make an adapter API that would call, behind the scene, the different external APIs (adapter_api).
My problem is the following : the external APIs are optimised when calling them with a batch of products attributes. However, my API would be optimised on a product by product basis.
I would like to somehow make a buffer of product attributes that would grow when I call my adapter_api. When the number of product attributes reach a certain limit, the ext_api would be called and the buffer would be reset and ready to receive more product attributes.
I'm wondering how to achieve that. I was thinking of making a REST api in python that would store the buffer of product attributes. I would like this REST api to be able to scale on a Kubernetes cluster : it would need low latency, and several instance of this API would write in the buffer of products until one of them reach the limit and make the call to the external API.
Here is what I have in mind :
Are there any best practices concerning the buffer on this use case ? To add some extra informations : my main purpose here is to hide from internal business APIs (not drawn) the complexity of calling many different external APIs each of which have their own rules and credentials.
Thank you very much for your help.

You didn't tell us your performance evaluation criteria.
You did tell us this:
don't know how to store the buffer : I would like to avoid databases or files.
which makes little sense,
since there's a simple answer to this question:
Is there any best practices on this use case ?
Yes. The best practice is to append requests to buffer.txt
and send the batch when that file exceeds some threshold.
A convenient way to implement the threshold would be
to send when getsize() reports a large enough value.
If requests are of quite different size and the batch
size really matters to you, then append a single byte
to a 2nd file, and use size of that to indicate how
many entries are enqueued.
requirements
The heart of your question seems to revolve around
what was left unsaid:
What is the cost function for sending too many "small" batches to ext_api?
What is the cost function for the consumer of the adapter_api, what does it care about? Low latency return, perhaps?
If ext_api permanently fails (say, a day of downtime), do we have some responsibility for quickly notifying the consumer that its updates are going into a black hole?
And why would using the filesystem be inappropriate?
It seems a perfect match for your needs.
Consider using a global in-memory object,
such as list or queue for the batch you're accumulating.
You might want to protect accesses with a lock.
Maybe your client doesn't really want a
one-product-at-a-time API.
Maybe you'd prefer to have your client
accumulate items,
sending only when its batch size is big enough.

Related

Best way for Node.js server to return an error before process runs out of heap memory

I'm running Node.js / Express server on a container with pretty strict memory constraints.
One of the endpoints I'd like to expose is a "batch" endpoint where a client can request a list of data objects in bulk from my data store. The individual objects vary in size, so it's difficult to set a hard limit on how many objects can be requested at one time. In most cases a client could request a large amount of objects without any issues, but it certain edge cases even requests for a small amount of objects will trigger an OOM error.
I'm familiar with Node's process.memoryUsage() & process.memoryUsage.rss(), but I'm worried about the performance implications of constantly checking heap (or service) memory usage while serving an individual batch request.
In the longer term, I might consider using memory monitoring to bake in some automatic pagination for the endpoint. In the short term, however, I'd just like to be able to return an informative error to the client in the event that they are requesting too many data objects at a given time (rather than have the entire application crash with an OOM error).
Are there any more effective methods or tools I could be using to solve the problem instead?
you have couple of options.
Options 1.
what is biggest object you have in store. I would say that you allow some {max object count} on api and set container memory to biggestObject x {max allowed objects size}. You can even have some pagination concept added if required where page size = {max object count}
Option 2.
Also using process.memoryUsage() should be fine too. I don't believe it is a not a costly call unless you have read this somewhere. Before each object pull check current memory and go ahead only if safe amount of memory is available.The response in this case can contain only pulled data and lets client to pull remaining ids in next call. Implementable via some paging logic too.
options 3.
explore streams. This I will not be able to add much info for now.

How to deal with Command which is depend on existing records in application using CQRS and Event sourcing

We are using CQRS with EventSourcing.
In our application we can add resources(it is business term for a single item) from ui and we are sending command accordingly to add resources.
So we have x number of resources present in application which were added previously.
Now, we have one special type of resource(I am calling it as SpecialResource).
When we add this SpecialResource , id needs to be linked with all existing resources in application.
Linked means this SpecialResource should have List of ids(guids) (List)of existing resources.
The solution which we tried to get all resource ids in applcation before adding the special
resource(i.e before firing the AddSpecialResource command).
Assign these List to SpecialResource, Then send AddSpecialResource command.
But we are not suppose to do so , because as per cqrs command should not query.
I.e. command cant depend upon query as query can have stale records.
How can we achieve this business scenario without querying existing records in application?
But we are not suppose to do so , because as per cqrs command should not query. I.e. command cant depend upon query as query can have stale records.
This isn't quite right.
"Commands" run queries all the time. If you are using event sourcing, in most cases your commands are queries -- "if this command were permitted, what events would be generated?"
The difference between this, and the situation you described, is the aggregate boundary, which in an event sourced domain is a fancy name for the event stream. An aggregate is allowed to run a query against its own event stream (which is to say, its own state) when processing a command. It's the other aggregates (event streams) that are out of bounds.
In practical terms, this means that if SpecialResource really does need to be transactionally consistent with the other resource ids, then all of that data needs to be part of the same aggregate, and therefore part of the same event stream, and everything from that point is pretty straight forward.
So if you have been modeling the resources with separate streams up to this point, and now you need SpecialResource to work as you have described, then you have a fairly significant change to your domain model to do.
The good news: that's probably not your real requirement. Consider what you have described so far - if resourceId:99652 is created one millisecond before SpecialResource, then it should be included in the state of SpecialResource, but if it is created one millisecond after, then it shouldn't. So what's the cost to the business if the resource created one millisecond before the SpecialResource is missed?
Because, a priori, that doesn't sound like something that should be too expensive.
More commonly, the real requirement looks something more like "SpecialResource needs to include all of the resource ids created prior to close of business", but you don't actually need SpecialResource until 5 minutes after close of business. In other words, you've got an SLA here, and you can use that SLA to better inform your command.
How can we achieve this business scenario without querying existing records in application?
Turn it around; run the query, copy the results of the query (the resource ids) into the command that creates SpecialResource, then dispatch the command to be passed to your domain model. The CreateSpecialResource command includes within it the correct list of resource ids, so the aggregate doesn't need to worry about how to discover that information.
It is hard to tell what your database is capable of, but the most consistent way of adding a "snapshot" is at the database layer, because there is no other common place in pure CQRS for that. (There are some articles on doing CQRS+ES snapshots, if that is what you actually try to achieve with SpecialResource).
One way may be to materialize list of ids using some kind of stored procedure with the arrival of AddSpecialResource command (at the database).
Another way is to capture "all existing resources (up to the moment)" with some marker (timestamp), never delete old resources, and add "SpecialResource" condition in the queries, which will use the SpecialResource data.
Ok, one more option (depends on your case at hand) is to always have the list of ids handy with the same query, which served the UI. This way the definition of "all resources" changes to "all resources as seen by the user (at some moment)".
I do not think any computer system is ever going to be 100% consistent simply because life does not, and can not, work like this. Apparently we are all also living in the past since it takes time for your brain to process input.
The point is that you do the best you can with the information at hand but ensure that your system is able to smooth out any edges. So if you need to associate one or two resources with your SpecialResource then you should be able to do so.
So even if you could associate your SpecialResource with all existing entries in your data store what is to say that there isn't another resource that has not yet been entered into the system that also needs to be associated.
It all, as usual, will depend on your specific use-case. This is why process managers, along with their state, enable one to massage that state until the process can complete.
I hope I didn't misinterpret your question :)
You can do two things in order to solve that problem:
make a distinction between write and read model. You know what read model is, right? So "write model" of data in contrast is a combination of data structures and behaviors that is just enough to enforce all invariants and generate consistent event(s) as a result of every executed command.
don't take a rule which states "Event Store is a single source of truth" too literally. Consider the following interpretation: ES is a single source of ALL truth for your application, however, for each specific command you can create "write models" which will provide just enough "truth" in order to make this command consistent.

FetchXml optimization

I am looking for the ways how to optimize bulk download of CRM2011 data. Here are the two main scenarios:
a) Full synchronization: Download of all data - first all accounts, then all contacts etc etc.
b) Incremental synchronization: Download of all entities modified since given date
We use multithread downloader with 3 threads. Each thread performs FetchXml for one entity type that is downloaded page by page. Parsed objects are stored in the downloader cache and the downloader goes on for the next page. There is another thread that pulls the downloaded data from the cache and processes them. This organization increases the download speed more than 2x.
The problems I see:
a) FetchXml protocol is very inefficient. For example it contains lots of unneeded data. Example: FormattedValues take 10-15% bandwidth (my data show ~15% in the source Xml stream or ~10% in the zipped stream), although all we do with it is a) Xml parsing, b) throwing away. (Note that the parsing is not negligible either - iOs/Android Mono parsers are surprisingly slow.)
b) In case of incremental synchronization most of the FetchXml requests return zero items. In this case it would be highly desirable to combine several FetchXml requests into one. (AFAIK it is impossible.) Or maybe use another trick such as to ask for the counts of modified objects I did not investigate what is possible yet.
Does anybody have any advice how to optimize FetchXml traffic?
Your fastest method would be to use SQL server directly for something like this (unless you are using online).
To make the incremental faster, your best bet is to use the aggregate functionality FetchXML provides which is both extremely quick and less verbose.
Why parse on the iOS/Android Mono? If you are instead sharing this to a large number of devices, you'd be better off having a central caching server that could send back this data in a json (zipped) format to the devices (or possibly bson). Then the caching server would request an update of the changes, process those and then send back incremental changes in whatever format to the clients. Would be considerably faster on the clients and far less bandwidth.
I'm not sure of a way to further optimize FetchXML. I would question why you're not using the OData Endpoints and REST, especially if you're primarily concerned with the about of data being sent over the wire.
I have talked to some brilliant CRM MVPs, and I know they have used REST to migrate data to CRM. I'm not sure if they did it because it was faster, but I assumed that was why.
I do know that you are going to minimize the amount of data that is being sent to the client, since XML is extremely bloated.
Have a look at the ExecuteMultipleRequest to allow you to perform multiple requests/queries at once... http://msdn.microsoft.com/en-us/library/microsoft.xrm.sdk.messages.executemultiplerequest.aspx

Working with large Backbone collections

We're designing a backbone application, in which each server-side collection has the potential to contain tens of thousands of records. As an analogy - think of going into the 'Sent Items' view of an email application.
In the majority of Backbone examples I've seen, the collections involved are at most 100-200 records, and therefore fetching the whole collection and working with it in the client is relatively easy. I don't believe this would be the case with a much larger set.
Has anyone done any work with Backbone on large server-side collections?
Have you encountered performance issues (especially on mobile devices) at a particular collection size?
What decision(s) did you take around how much to fetch from the server?
Do you download everything or just a subset?
Where do you put the logic around any custom mechanism (Collection prototype for example?)
Yes, at about 10,000 items, older browsers could not handle the display well. We thought it was a bandwidth issue, but even locally, with as much bandwidth as a high-performance machine could throw at it, Javascript just kinda passed out. This was true on Firefox 2 and IE7; I haven't tested it on larger systems since.
We were trying to fetch everything. This didn't work for large datasets. It was especially pernicious with Android's browser.
Our data was in a tree structure, with other data depending upon the presence of data in the tree structure. The data could change due to actions from other users, or other parts of the program. Eventually, we made the tree structure fetch only the currently visible nodes, and the other parts of the system verified the validity of the datasets on which they dependent independently. This is a race condition, but in actual deployment we never saw any problems. I would have liked to use socket.io here, but management didn't understand or trust it.
Since I use Coffeescript, I just inherited from Backbone.Collection and created my own superclass, which also instantiated a custom sync() call. The syntax for invoking a superclass's method is really useful here:
class Dataset extends BaseAccessClass
initialize: (attributes, options) ->
Dataset.__super__.initialize.apply(#, arguments)
# Customizations go here.
Like Elf said you should really paginate loading data from the server. You'd save a lot of load on the server from downloading items you may not need. Just creating a collection with 10k models locally in Chrome take half a second. It's a huge load.
You can put the work on another physical CPU thread by using a worker and then use transient objects to sent it to the main thread in order to render it on the DOM.
Once you have a collection that big rendering in the DOM lazy rendering will only get you so far. The memory will slowly increase until it crashes the browser (that will be quick on tablets). You should use object pooling on the elements. It will allow you to set a small max size for the memory and keep it there.
I'm building a PerfView for Backbone that can render 1,000,000 models and scroll at 120FPS on Chrome. The code is all up on Github https://github.com/puppybits/BackboneJS-PerfView. It;s commented so theres a lot of other optimizations you'd need to display large data sets.

Has anybody some advice on programming realtime audio synthesis?

I'm currently working on a personal project: creating a library for realtime audio synthesis in Flash. In short: tools to connect wavegenarators, filters, mixers, etc with eachother and supply the soundcard with raw (realtime) data. Something like max/msp or Reaktor.
I already have some working stuff, but I'm wondering if the basic setup that I wrote is right. I don't want to run into problems later on that force me to change the core of my app (although that can always happen).
Basically, what I do now is start at the end of the chain, at the place where the (raw) sounddata goes 'out' (to the soundcard). To do that, I need to write chunks of bytes (ByteArrays) to an object, and to get that chunk I ask whatever module is connected to my 'Sound Out' module to give me his chunk. That module does the same request to the module that's connected to his input, and that keeps happening until the start of the chain is reached.
Is this the right approach? I can imagine running into problems if there's a feedbackloop, or if there's another module with no output: if i were to connect a spectrumanalyzer somewhere, that would be a dead end in the chain (a module with no outputs, just an input). In my current setup, such a module wouldnt work because i only start calculating from the sound-output module.
Has anyone experience with programming something like this? I'd be very interested in some thoughts about the right approach. (For clarity: i'm not looking for specific Flash-implementations, and that's why i didnt tag this question under flash or actionscript)
I did a similar thing a while back, and I used the same approach as you do - start at the virtual line out, and trace the signal back to the top. I did this per sample though, not per buffer; if I were to write the same application today, I might choose per-buffer instead though, because I suspect it would perform better.
The spectrometer was designed as an insert module, that is, it would only work if both its input and its output were connected, and it would pass its input to the output unchanged.
To handle feedback, I had a special helper module that introduced a 1-sample delay and would only fetch its input once per cycle.
Also, I think doing all your internal processing with floats, and thus arrays of floats as the buffers, would be a lot easier than byte arrays, and it would save you the extra effort of converting between integers and floats all the time.
In later versions you may have different packet rates in different parts of your network.
One example would be if you extend it to transfer data to or from disk. Another example
would be that low data rate control variables such as one controlling echo-delay may, later, become a part of your network. You probably don't want to process control variables with the same frequency that you process audio packets, but they are still 'real time' and part of the function network. They may for example need smoothing to avoid sudden transitions.
As long as you are calling all your functions at the same rate, and all the functions are essentially taking constant-time, your pull-the-data approach will work fine. There will
be little to choose between pulling data and pushing. Pulling is somewhat more natural for playing audio, pushing is somewhat more natural for recording, but either works and ends up making the same calls to the underlying audio processing functions.
For the spectrometer you've got
the issue of multiple sinks for
data, but it is not a problem.
Introduce a dummy link to it from
the real sink. The dummy link can
cause a request for data that is not
honoured. As long as the dummy link knows
it is a dummy and does not care about
the lack of data, everything will be
OK. This is a standard technique for reducing multiple sinks or sources to a single one.
With this kind of network you do not want to do the same calculation twice in one complete update. For example if you mix a high-passed and low-passed version of a signal you do not want to evaluate the original signal twice. You must do something like record a timer tick value with each buffer, and stop propagation of pulls when you see the current tick value is already present. This same mechanism will also protect you against feedback loops in evaluation.
So, those two issues of concern to you are easily addressed within your current framework.
Rate matching where there are different packet rates in different parts of the network is where the problems with the current approach will start. If you are writing audio to disk then for efficiency you'll want to write large chunks infrequently. You don't want to be blocking your servicing of the more frequent small audio input and output processing packets during those writes. A single rate pulling or pushing strategy on its own won't be enough.
Just accept that at some point you may need a more sophisticated way of updating than a single rate network. When that happens you'll need threads for the different rates that are running, or you'll write your own simple scheduler, possibly as simple as calling less frequently evaluated functions one time in n, to make the rates match. You don't need to plan ahead for this. Your audio functions are almost certainly already delegating responsibility for ensuring their input buffers are ready to other functions, and it will only be those other functions that need to change, not the audio functions themselves.
The one thing I would advise at this stage is to be careful to centralise audio buffer
allocation, noticing that buffers are like fenceposts. They don't belong to an audio
function, they lie between the audio functions. Centralising the buffer allocation will make it easy to retrospectively modify the update strategy for different rates in different parts of the network.

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