downsampling

问题 I am attempting to perform undersampling of the majority class using python scikit learn. Currently my codes look for the N of the minority class and then try to undersample the exact same N from the majority class. And both the test and training data have this 1:1 distribution as a result. But what I really want is to do this 1:1 distribution on the training data ONLY but test it on the original distribution in the testing data. I am not quite sure how to do the latter as there is some dict

问题 I have a csv file with two columns containing dates and 0 or 1 like so: 17/08/2012 07:47:16 0 17/08/2012 07:54:31 1 17/08/2012 08:02:31 0 17/08/2012 09:22:33 0 17/08/2012 09:58:05 0 17/08/2012 12:26:59 1 17/08/2012 20:56:00 0 18/08/2012 10:04:06 0 18/08/2012 10:42:52 0 20/08/2012 07:22:02 0 20/08/2012 07:54:28 0 20/08/2012 08:01:58 0 20/08/2012 08:16:31 1 20/08/2012 08:26:38 0 20/08/2012 08:55:19 1 20/08/2012 09:00:09 0 20/08/2012 09:26:11 0 20/08/2012 09:50:10 0 20/08/2012 10:33:37 0 20/08

问题 I'm trying to downsample a .wav audio from 22050 to 8000 using AudioInputStream but the conversion returns me 0 data bytes. Here is the code: AudioInputStream ais; AudioInputStream eightKhzInputStream = null; ais = AudioSystem.getAudioInputStream(file); if (ais.getFormat().getSampleRate() == 22050f) { AudioFileFormat sourceFileFormat = AudioSystem.getAudioFileFormat(file); AudioFileFormat.Type targetFileType = sourceFileFormat.getType(); AudioFormat sourceFormat = ais.getFormat(); AudioFormat

问题 I have been working on MNIST dataset to learn how to use Tensorflow and Python for my deep learning course. I could read the data internally/externally and also train it in softmax and cnn thanks to tensorflow tutorial at website. At the end, I could get >%90 in softmax, >%98 in cnn, accuracy. My problem is that I want to resize all images on MNIST as 14x14 and train it again, also to augment all (noising, rotating etc.) and train again. At the end, I want to be able to compare the accuracies

问题 I am using recorder.js to record audio. When I download .wav file it is 48KHz but I want in 16KHz, mono channel. function init(config) { sampleRate = config.sampleRate; //to 16000 numChannels = config.numChannels; //to 1 initBuffers(); } It changes the recorded voice (like a voice of the robot). Help me to get a voice in 16KHz, mono .wav file in a normal voice. (Thanx in Advance) My Code Is Here 来源： https://stackoverflow.com/questions/49043887/need-desired-format-of-wav-in-recoder-js

问题 So here is my problem: I have an image, that image is large (high resolution) and it needs to be small (much lower resolution). So I do the naive thing (kill every other pixel) and the result looks poor. So I try to do something more intelligent (low pass filtering using a Fourier transform and re-sampling in Fourier space) and the result is a little better but still fairly poor. So my question, is there a perceptually motivated image down-sampling algorithm (or implementation)? edit: While I

问题 I've been working on a audio-recognize demo for some time, and the api needs me to pass an .wav file with sample rate of 8000 or 16000 , so I have to downsample it. I have tried 2 algorithms as following. Though none of them solves the problem as I wish, there's some differences of the results and I hope that will make it more clear. This is my first try, and it works fine when sampleRate % outputSampleRate = 0 , however when outputSampleRate = 8000 or 1600 , the outcome audio file is silent

问题 I issue the following command: gs \ -o downsampled.pdf \ -sDEVICE=pdfwrite \ -dDownsampleColorImages=true \ -dColorImageResolution=180 \ -dColorImageDownsampleThreshold=1.0 \ And get the following errors: Subsample filter does not support non-integer downsample factor (1.994360) Failed to initialise downsample filter, downsampling aborted (on some pages) and: Subsample filter does not support non-integer downsample factor (2.000029) Failed to initialise downsample filter, downsampling aborted

问题 SImple question but I haven't been able to find a simple answer. I have a list of data which counts the time in seconds that events occur: [200.0 420.0 560.0 1100.0 1900.0 2700.0 3400.0 3900.0 4234.2 4800.0 etc..] I want to count how many events occur each hour (3600 seconds) and create a new list of these counts. I understand this is called downsampling, but all the information I can find is related to traditional time series. For the example above the new list would look like: [7 3 etc..]

问题 My android application needs to convert PCM(22khz) to AMR , but the API AmrInputStream only supports with pcm of 8khz . How can i downsample the pcm from 22 khz to 8 khz? 回答1: The sample rate is hard coded in AmrInputStream.java. // frame is 20 msec at 8.000 khz private final static int SAMPLES_PER_FRAME = 8000 * 20 / 1000; So you have to convert the PCM to AMR first. InputStream inStream; inStream = new FileInputStream(wavFilename); AmrInputStream aStream = new AmrInputStream(inStream); File