sampler | shell commands execution , visualization and alerting | Data Visualization library
kandi X-RAY | sampler Summary
kandi X-RAY | sampler Summary
Sampler is a tool for shell commands execution, visualization and alerting. Configured with a simple YAML file.
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Currently covering the most popular Java, JavaScript and Python libraries. See a Sample of sampler
sampler Key Features
sampler Examples and Code Snippets
def log_uniform_candidate_sampler(true_classes, num_true, num_sampled, unique,
range_max, seed=None, name=None):
"""Samples a set of classes using a log-uniform (Zipfian) base distribution.
This operation random
def _filter_ds(dataset,
acceptance_dist_ds,
initial_dist_ds,
class_func,
seed,
name=None):
"""Filters a dataset based on per-class acceptance probabilities.
Args:
dat
def uniform_candidate_sampler(true_classes, num_true, num_sampled, unique,
range_max, seed=None, name=None):
"""Samples a set of classes using a uniform base distribution.
This operation randomly samples a tensor of
Community Discussions
Trending Discussions on sampler
QUESTION
This seems to be an issue after I upgraded my iPod Touch to iOS 15 (15.0.1).
When running the example below, it works fine on the first load, playing the sound as many times as I want. However, if I lock the screen on my iPod Touch, then return after a couple minutes, the sound no longer plays. To troubleshoot I set up a state change listener on the AudioContext instance and verified that Safari sets the state back to running
after it was set to interrupted
when the screen was locked. Yet, the sound does not play.
ANSWER
Answered 2021-Oct-12 at 19:34Strangely, if I add an audio element to the HTML that points to some mp3 file, which isn't even referenced in the code at all, then locking the screen for a while then returning to the page no longer affects the audio playback.
Updated code:
QUESTION
I have a map-stype dataset, which is used for instance segmentation tasks. The dataset is very imbalanced, in the sense that some images have only 10 objects while others have up to 1200.
How can I limit the number of objects per batch?
A minimal reproducible example is:
...ANSWER
Answered 2022-Mar-17 at 19:22If what you are trying to solve really is:
QUESTION
Currently i'm able to train a Semantic Role Labeling model using the config file below. This config file is based on the one provided by AllenNLP and works for the default bert-base-uncased
model and also GroNLP/bert-base-dutch-cased
.
ANSWER
Answered 2022-Feb-24 at 02:14The easiest way to resolve this is to patch SrlReader
so that it uses PretrainedTransformerTokenizer
(from AllenNLP) or AutoTokenizer
(from Huggingface) instead of BertTokenizer
. SrlReader
is an old class, and was written against an old version of the Huggingface tokenizer API, so it's not so easy to upgrade.
If you want to submit a pull request in the AllenNLP project, I'd be happy to help you get it merged into AllenNLP!
QUESTION
I'm trying to create a sound using Fourier coefficients.
First of all please let me show how I got Fourier coefficients.
(1) I took a snapshot of a waveform from a microphone sound.
- Getting microphone: getUserMedia()
- Getting microphone sound: MediaStreamAudioSourceNode
- Getting waveform data: AnalyserNode.getByteTimeDomainData()
The data looks like the below: (I stringified Uint8Array, which is the return value of getByteTimeDomainData()
, and added length
property in order to change this object to Array later)
ANSWER
Answered 2022-Feb-04 at 23:39In golang I have taken an array ARR1 which represents a time series ( could be audio or in my case an image ) where each element of this time domain array is a floating point value which represents the height of the raw audio curve as it wobbles ... I then fed this floating point array into a FFT call which returned a new array ARR2 by definition in the frequency domain where each element of this array is a single complex number where both the real and the imaginary parts are floating points ... when I then fed this array into an inverse FFT call ( IFFT ) it gave back a floating point array ARR3 in the time domain ... to a first approximation ARR3 matched ARR1 ... needless to say if I then took ARR3 and fed it into a FFT call its output ARR4 would match ARR2 ... essentially you have this time_domain_array --> FFT call -> frequency_domain_array --> InverseFFT call -> time_domain_array ... rinse N repeat
I know Web Audio API has a FFT call ... do not know whether it has an IFFT api call however if no IFFT ( inverse FFT ) you can write your own such function here is how ... iterate across ARR2 and for each element calculate the magnitude of this frequency ( each element of ARR2 represents one frequency and in the literature you will see ARR2 referred to as the frequency bins which simply means each element of the array holds one complex number and as you iterate across the array each successive element represents a distinct frequency starting from element 0 to store frequency 0 and each subsequent array element will represent a frequency defined by adding incr_freq
to the frequency of the prior array element )
Each index of ARR2 represents a frequency where element 0 is the DC bias which is the zero offset bias of your input ARR1 curve if its centered about the zero crossing point this value is zero normally element 0 can be ignored ... the difference in frequency between each element of ARR2 is a constant frequency increment which can be calculated using
QUESTION
I'm currently writing a gravity-simulation and I have a small problem displaying the particles with OpenGL.
To get "round" particles, I create a small float-array like this:
...ANSWER
Answered 2022-Jan-19 at 16:51You're in a special case where your fragments are either fully opaque or fully transparent, so it's possible to get depth-testing and blending to work at the same time. The actual problem is, that for depth testing even a fully transparent fragment will store it's depth value. You can prevent the writing by explicitly discarding the fragment in the shader. Something like:
QUESTION
I've got a simple wav header reader i found online a long time ago, i've gotten back round to using it but it seems to replace around 1200 samples towards the end of the data chunk with a single random repeated number, eg -126800. At the end of the sample is expected silence so the number should be zero.
Here is the simple program:
...ANSWER
Answered 2022-Jan-07 at 21:55WAV is just a container for different audio sample formats.
You're making assumptions on a wav file that would have been OK on Windows 3.11 :) These don't hold in 2021.
Instead of rolling your own Wav file reader, simply use one of the available libraries. I personally have good experiences using libsndfile
, which has been around roughly forever, is very slim, can deal with all prevalent WAV file formats, and with a lot of other file formats as well, unless you disable that.
This looks like a windows program (one notices by the fact you're using very WIN32API style capital struct names – that's a bit oldschool); so, you can download libsndfile's installer from the github releases and directly use it in your visual studio (another blind guess).
QUESTION
I created a music database application a few years ago in C++ (Code::Blocks + wxWidgets + SQLAPI++) and Firebird as the database server (running as a service in classic mode) on the Windows platform (v10). It creates a SQL database with tables, views, triggers, generators.
So far, it has been running perfectly up to Firebird 3 (Latest version). Now Firebird 4.0 is out, I thought I try it out.
In order to narrow down on the problem, I created a new app that only creates the database, tables, triggers, generators,and only 2 views which are focused around the problem area.
The code for vew_AlbumDetails I use in my test app is:
...ANSWER
Answered 2021-Dec-23 at 06:20I have added 'DataTypeCompatibility = 3.0' to both databases.conf and firebird.conf.
The datatype for Album_NrSeconds is now NUMERIC.
My application runs flawlessly under Firebird 4.0 as a service after these 2 edits.
Thank you Mark Rotteveel for your suggestion. Its much appreciated.
QUESTION
We are working on a project which allows us to record some sounds from a microphone with a 5k Hz sample rate with some Low-Pass filter & HighPass filter.
What we are using
We are using AvaudioEngine for this purpose.
We are using AVAudioConverter for downgrading the sample rate.
We are using AVAudioUnitEQ for the LowPass & HighPass filter.
Code
...ANSWER
Answered 2021-Dec-17 at 10:50I think the main problem with this code was that the AVAudioConverter
was being created before calling engine.prepare()
which can and will change the mainMixerNode
output format. Aside from that, there was a redundant connection of mainMixerNode
to outputNode
, along with a probably incorrect format - mainMixerNode
is documented to be automatically created and connected to the output node "on demand". The tap also did not need a format.
QUESTION
Let's first acknowledge that OpenGL is deprecated by Apple, that the last supported version is 4.1 and that that's a shame but hey, we've got to move forward somehow and Vulkan is the way :trollface: Now that that's out of our systems, let's have a look at this weird bug I found. And let me be clear that I am running this on an Apple Silicon M1, late 2020 MacBook Pro with macOS 11.6. Let's proceed.
I've been following LearnOpenGL and I have published my WiP right here to track my progress. All good until I got to textures. Using one texture was easy enough so I went straight into using more than one, and that's when I got into trouble. As I understand it, the workflow is more or less
- load pixel data in a byte array called
textureData
, plus extra info glGenTextures(1, &textureID)
glBindTexture(GL_TEXTURE_2D, textureID)
- set parameters at will
glTexImage2D(GL_TEXTURE_2D, ... , textureData)
glGenerateMipmap(GL_TEXTURE_2D)
(although this may be optional)
which is what I do around here, and then
glUniform1i(glGetUniformLocation(ID, "textureSampler"), textureID)
- rinse and repeat for the other texture
and then, in the drawing loop, I should have the following:
glUseProgram(shaderID)
glActiveTexture(GL_TEXTURE0)
glBindTexture(GL_TEXTURE_2D, textureID)
glActiveTexture(GL_TEXTURE1)
glBindTexture(GL_TEXTURE_2D, otherTextureID)
I then prepare my fancy fragment shader as follows:
...ANSWER
Answered 2021-Dec-13 at 19:23Instead of passing a texture handle to glUniform1i(glGetUniformLocation(ID, "textureSampler"), ...)
, you need to pass a texture slot index.
E.g. if you did glActiveTexture(GL_TEXTUREn)
before binding the texture, pass n
.
QUESTION
I need to write an OpenCL kernel. Via kernel argument I get an input image with a certain dimension (example: width: 600px, height: 400px
). The algorithm I need to run is: "Nearest Neighbor Interpolation".
In the image below, one starts from pixel (1,1) (left image, green pixel). My scaling factor is 3. This means that in my output image this pixel must appear 9 times.
Now my question is, how can I give each green pixel the value (pixelValue) of the SourceImage
using the code below (using 2 for-loops).
ANSWER
Answered 2021-Dec-13 at 15:40In short: The trick to do nearest neighbor interpolation is integer arithmetic or in some cases float->integer casting.
In your case, the solution for the ?? in the double loop is:
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