What is the concept of CNTKTextFormatDeserializer and why use? - cntk

I am using the CNTKTextReader to read in my training and test sets. The train file is getting large ( 2.7 GB now, and soon to get bigger ).
I don't understand what is "CNTKTextFormatDeserializer" -- the doc I found didn't explain what the big picture is -- what is it and why use it? The doc I found just went into syntax of it.
So, is it a way to use a binary version of these files to make them more compact?

Readers in general are just a way to make certain aspects of training easier. These include
randomization: SGD generalizes better when the data presented to it are coming in random order. The reader can randomize the data for you with shuffling happening on the fly.
distributed training: For distributed training the reader is aware of the multiple workers and can make sure they receive distinct chunks of data.
memory budget issues: The reader does not load the whole training file in memory.
language agnostic i/o: The reader provides a cross-platform way to read data. (if you want to always be in Python, you might not care about this but others do).
The CTF format is a little verbose and indeed there is a binary format deserializer that was recently added.

Related

Is it better better to open or to read large matrices in Julia?

I'm in the process of switching over to Julia from other programming languages and one of the things that Julia will let you hang yourself on is memory. I think this is likely a good thing, a programming language where you actually have to think about some amount of memory management forces the coder to write more efficient code. This would be in contrast to something like R where you can seemingly load datasets that are larger than the allocated memory. Of course, you can't actually do that, so I wonder how does R get around that problem?
Part of what I've done in other programming languages is work on large tabular datasets, often converted over to a R dataframe or a matrix. I think the way this is handled in Julia is to stream data in wherever possible, so my main question is this:
Is it better to use readline("my_file.txt") to access data or is it better to use open("my_file.txt", "w")? If possible, wouldn't it be better to access a large dataset all at once for speed? Or would it be better to always stream data?
I hope this makes sense. Any further resources would be greatly appreciated.
I'm not an extensive user of Julia's data-ecosystem packages, but CSV.jl offers the Chunks and Rows alternatives to File, and these might let you process the files incrementally.
While it may not be relevant to your use case, the mechanisms mentioned in #Przemyslaw Szufel's answer are used other places as well. Two I'm familiar with are the TiffImages.jl and NRRD.jl packages, both I/O packages mostly for loading image data into Julia. With these, you can load terabyte-sized datasets on a laptop. There may be more packages that use the same mechanism, and many package maintainers would probably be grateful to receive a pull request that supports optional memory-mapping when applicable.
In R you cannot have a data frame larger than memory. There is no magical buffering mechanism. However, when running R-based analytics you could use a disk.frame package for that.
Similarly, in Julia if you want to process data frames larger than memory you need to use am appropriate package. The most reasonable and natural option in Julia ecosystem is JuliaDB.
If you want to do something more low-level solution have a look at:
Mmap that provides Memory-mapped I/O that exactly solves the issue of conveniently handling data too large to fit into memory
SharedArrays that offers a disk mapped array with implementation based on Mmap.
In conclusion, if your data is data frame based - try JuliaDB, otherwise have a look at Mmap and SharedArrays (look at the filename parameter)

Implement data generator in federated training

(I have posted the question on https://github.com/tensorflow/federated/issues/793 and maybe also here!)
I have customized my own data and model to federated interfaces and the training converged. But I am confused about an issue that in an images classification task, the whole dataset is extreme large and it can't be stored in a single federated_train_data nor be imported to memory for one time. So I need to load the dataset from the hard disk in batches to memory real-timely and use Keras model.fit_generator instead of model.fit during training, the approach people use to deal with large data.
I suppose in iterative_process shown in image classification tutorial, the model is fitted on a fixed set of data. Is there any way to adjust the code to let it fit to a data generator?I have looked into the source codes but still quite confused. Would be incredibly grateful for any hints.
Generally, TFF considers the feeding of data to be part of the "Python driver loop", which is a helpful distinction to make when writing TFF code.
In fact, when writing TFF, there are generally three levels at which one may be writing:
TensorFlow defining local processing (IE, processing that will happen on the clients, or on the server, or in the aggregators, or at any other placement one may want, but only a single placement.
Native TFF defining the way data is communicated across placements. For example, writing tff.federated_sum inside of a tff.federated_computation decorator; writing this line declares "this data is moved from clients to server, and aggregated via the sum operator".
Python "driving" the TFF loop, e.g. running a single round. It is the job of this final level to do what a "real" federated learning runtime would do; one example here would be selecting the clients for a given round.
If this breakdown is kept in mind, using a generator or some other lazy-evaluation-style construct to feed data in to a federated computation becomes relatively simple; it is just done at the Python level.
One way this could be done is via the create_tf_dataset_for_client method on the ClientData object; as you loop over rounds, your Python code can select from the list of client_ids, then you can instantiate a new list of tf.data.Datasetsand pass them in as your new set of client data. An example of this relatively simple usage would be here, and a more advanced usage (involving defining a custom client_datasets_fn which takes client_id as a parameter, and passing it to a separately-defined training loop would be here, in the code associated to this paper.
One final note: instantiating a tf.data.Dataset does not actually load the dataset into memory; the dataset is only loaded in when it is iterated over. One helpful tip I have received from the lead author of tf.data.Dataset is to think of tf.data.Dataset more as a "dataset recipe" than a literal instantiation of the dataset itself. It has been suggested that perhaps a better name would have been DataSource for this construct; hopefully that may help the mental model on what is actually happening. Similarly, using the tff.simulation.ClientData object generally shouldn't really load anything into memory until it is iterated over in training on the clients; this should make some nuances around managing dataset memory simpler.

Is there a way to partition a tf.Dataset with TensorFlow’s Dataset API?

I checked the doc but I could not find a method for it. I want to de cross validation, so I kind of need it.
Note that I'm not asking how to split a tensor, as I know that TensorFlow provides an API for that an has been answered in another question. I'm asking on how to partition a tf.Dataset (which is an abstraction).
You could either:
1) Use the shard transformation partition the dataset into multiple "shards". Note that for best performance, sharding should be to data sources (e.g. filenames).
2) As of TensorFlow 1.12, you can also use the window transformation to build a dataset of datasets.
I am afraid you cannot. The dataset API is a way to efficiently stream inputs to your net at run time. It is not a set of tools to manipulate datasets as a whole -- in that regards it might be a bit of a misnomer.
Also, if you could, this would probably be a bad idea. You would rather have this train/test split done once and for all.
it let you review those sets offline
if the split is done each time you run an experiment there is a risk that samples start swapping sets if you are not extremely careful (e.g. when you add more data to your existing dataset)
See also a related question about how to split a set into training & testing in tensorflow.

IPA (International Phonetic Alphabet) Transcription with Tensorflow

I'm looking into designing a software platform that will aid linguists and anthropologists in their study of previously unstudied languages. Statistics show that around 1,000 languages exist that have never been studied by a person outside of their respective speaker groups.
My goal is to utilize TensorFlow to make a platform that will allow linguists to study and document these languages more efficiently, and to help them create written systems for the ones that don't have a written system already. One of their current methods of accomplishing such a task is three-fold: 1) Record a native speaker conversing in the language, 2) Listening to that recording and trying to transcribe it into the IPA, 3) From the phonetics, analyzing the phonemics and phonotactics of the language to eventually create a written system for the speaker.
My proposed platform would cut that research time down from a minimum of a year to a maximum of six months. Before I start, I have some questions...
What would be required to train TensorFlow to transcribe live audio into the IPA? Has this already been done? and if so, how would I utilize a previous solution for this project? Is a project like this even possible with TensorFlow? if not, what would you recommend using instead?
My apologies for the magnitude of this question. I don't have much experience in the realm of machine learning, as I am just beginning the research process for this project. Any help is appreciated!
I guess I will take a first shot at answering this. Since the question is pretty general, my answer will have to be pretty general as well.
What would be required. At the very least you would have to have a large dataset of pre-transcribed data. Ideally a large amount of spoken language audio mapped to characters in the phonetic alphabet, so the system could learn the sound of individual characters rather than whole transcribed words. If such a dataset doesn't exist, a less granular dataset could be used, mapping single words to their transcriptions. Then you would need a model, that is the actual neural network architecture implemented in code. And lastly you would need some computing resources. This is not something you can train casually, you would either have to buy some time in a cloud based machine learning framework (like Google Cloud ML) or build a fairly expensive machine to train at home.
Has this been done? I don't know. I don't think so. There have been published papers reporting various degrees of success at training systems to transcribe speech. Here is one, for example, http://deeplearning.stanford.edu/lexfree/lexfree.pdf It seems that since the alphabet you want to transcribe to is specifically designed to capture the way words sound rather than just write down the words you might have more success at training such a model.
Is it possible with TensorFlow. Yes, most likely. TensorFlow is well suited for implementing most modern deep learning architectures. Unless you end up designing some really weird and very original model for this purpose, TensorFlow should work just fine.
Edit: after some thought in part 1, you would have to use a dataset mapping spoken words to their transcriptions, since I expect that the same sound pronounced separately would be different from when the same sound is used in a word.
This has actually been done, albeit in PyTorch, by a group at CMU: https://github.com/xinjli/allosaurus

I/O performance difference for sequential vs random acess with MxNet data iterators?

I would like to supply to a network many training images that are sampled from a dataset by following certain sampling rules. Now I have two choices:
Use the sampling logic to generate a list of images offline, then convert the .lst file to .rec file and use an sequential DataIter to access it.
Write my own child class of DataIter that can sample the images online. As a result, the class need to support random access, maybe inheriting from MXIndexedRecordIO. I will need to create a .rec file for the original dataset.
My intuition tells me that sequential access will be faster than random access for a .rec file. But I don't know if the difference is big enough to worth the additional time I spend in writing and testing my own iterator class. Could anyone give me a hint on this?
In your case you are better off prepacking images using MXRecordIO. It will give you a boost of performance and also introduce consistency in how you handle the dataset.
It will store the files in a .rec file as a list, where order matters
You can then use mxnet.image.ImageIter to iterate over .rec in order.
http://mxnet.io/api/python/io.html#mxnet.image.ImageIter
Since this is a question about performance, I guess it depends on how fast your network can process images which in turn depends on what hardware you are running your training on.