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• SPARC Solaris Momentum

  - by Mike Mulkey-Oracle

  Following up on the Oracle Solaris 11.2 launch on April 29th, if you were able to watch the launch event, you saw Mark Hurd state that Oracle will be No. 1 in high-end computing systems "in a reasonable time frame”.  "This is not a 3-year vision," he continued.Well, According to IDC's latest 1QCY14 Tracker, Oracle has regained the #1 UNIX Shipments Marketshare! You can see the report and read about it here: Oracle regains the #1 UNIX Shipments Marketshare, but suffice to say that SPARC Solaris is making strong gains on the competition.  If you have seen the public roadmap through 2019 of Oracle's commitment to continue to deliver on this technology, you can see that Mark Hurd’s comment was not to be taken lightly.  We feel the systems tide turning in Oracle's direction and are working hard to show our partner community the value of being a part of the SPARC Solaris momentum.We are now planning for the Solaris 11.2 GA in late summer (11.2 beta is available now), as well as doing early preparations for Oracle OpenWorld 2014 on September 28th.  Stay tuned there!Here is a sampling of the coverage highlights around the Oracle Solaris 11.2 launch:“Solaris is still one of the most advanced platforms in the enterprise.” – ITBusinessEdge“Oracle is serious about clouds now, just as its customers are, whether they are building them in their own datacenters or planning to use public clouds.” – EnterpriseTech"Solaris is more about a layer of an integrated system than an operating system.” — ZDNet

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• Origin of common list-processing function names

  - by Heatsink

  Some higher-order functions for operating on lists or arrays have been repeatedly adopted or reinvented. The functions map, fold[l|r], and filter are found together in several programming languages, such as Scheme, ML, and Python, that don't seem to have a common ancestor. I'm going with these three names to keep the question focused. To show that the names are not universal, here is a sampling of names for equivalent functionality in other languages. C++ has transform instead of map and remove_if instead of filter (reversing the meaning of the predicate). Lisp has mapcar instead of map, remove-if-not instead of filter, and reduce instead of fold (Some modern Lisp variants have map but this appears to be a derived form.) C# uses Select instead of map and Where instead of filter. C#'s names came from SQL via LINQ, and despite the name changes, their functionality was influenced by Haskell, which was itself influenced by ML. The names map, fold, and filter are widespread, but not universal. This suggests that they were borrowed from an influential source into other contemporary languages. Where did these function names come from?

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• GPU based procedual terrain borders?

  - by OnePie

  I'm working on a game that preferibly should feature a combination of designed and procedually generated terrain where the designer specifies in somewhat detailed terms what type of terrain a given area will have (grasslands, forest etc...) and then a precedual algorithm takes care of the rest. I'm not talking about minecraft style biomoes, but rather the game map for a strategy game. Each 'area' will not take up that much of the screen, and thus be more akin to a tile whose texture is procedually generated. While procedually generating terrain textures on the GPU are not that difficult, the hard part is making the borders between them look good. Currently, the 'tiles' are large enough to be visible (due to memory constraints mainly, we are talking planetary sized textures for a game taking place in space and on a continental ground view with seamless transitions between them) and creating good borders between them with an algorithm that is fast enough to be useful has proven difficult. Sampling the n-surrounding pixels and using the combiened result did not yield very good borders and was fairly slow on the GPU to boot (ca 12ms for me, that is without any lighning or shading and with very simple terrain texture shaders). So are there any practical known methods to solve this problem?

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• M-Audio Delta starts up at wrong sample rate

  - by steevc

  When the PC starts my M-Audio Delta 66 is using 48000kHz sampling rate when it is set for 44100 in Envy24. This causes audio to play slower than it should. This is in Kubuntu 14.04 on my new PC using an AMD A8 6500 with 8GB. When I first installed it seemed okay, but at some point it went wrong and has been doing this consistently since then. Kernel is 3.13.0-24-generic #47-Ubuntu SMP Fri May 2 23:31:42 UTC 2014 i686 athlon i686 GNU/Linux steve@slarti:~$ cat /proc/asound/card2/pcm0p/sub0/hw_params access: MMAP_INTERLEAVED format: S32_LE subformat: STD channels: 10 rate: 48000 (48000/1) period_size: 441 buffer_size: 3528 I can get it to switch to 44100 if I disable/enable the Delta in Pulseaudio volume control, but I have to do this every day and the sound still seems distorted. I can't see any issues in any of the config or log files I can find. If I boot the PC with a Mint Live USB it starts at 44100 and sound fine. Originally reported on Youtube is playing at the wrong speed - maybe soundcard related, but I'll close that and have this more relevant question instead.

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• EDQ Technical Enablement for OPN (Prague - June 17-19)

  - by milomir.vojvodic

  Oracle Enterprise Data Quality (EDQ) Technical Enablement and Partner Training Trusted Data for Your Enterprise Applications Oracle Enterprise Data Quality helps organizations achieve maximum value from their business-critical applications by delivering fit-for-purpose data. These products also enable individuals and collaborative teams to quickly and easily identify and resolve any problems in underlying data. With Oracle Enterprise Data Quality, customers can identify new opportunities, improve operational efficiency, and more efficiently comply with industry or governmental regulation. Oracle Enterprise Data Quality is designed to serve as a very channel friendly platform to OPN.  This means that pre-built extensions, components and even complete business solutions can readily be built and shared.  This allows our customers/partners to be highly efficient in how they deploy custom business solutions, but also allows our partners to develop specialized components, domain knowledge and even complete business solutions. Training is suitable for: · Database administrators · Architects · Technical staff Objectives of the training: After completing this course, participants should: · Have an understanding of the core functionality of EDQ across profiling, auditing, transforming, parsing and matching data · Be able to describe some of the key capabilities and benefits delivered by EDQ · Be able to create and run standalone EDQ processes and jobs · Be ready to start working with data from customers and (with practice) be able to demonstrate EDQ to customers Agenda 17th June Fundamentals For Demoing (Profile, Audit, Transform and More) Profiling Auditing Transforming Writing and exporting data Jobs and scheduling Publishing, packaging and copying EDQ processes Introduction to the Customer Data Extension Pack Realtime Processing via Web Services The Server Console Run Profiles Data Interfaces Sampling Publishing metrics to the Dashboard Users and security 18th June Matching Matching overview Basic matching configuration Matching rule hierarchies Clustering Merging Reviewing possible matches Outputting Match Data Case study 19th June Address Verification Address Verification Overview Configuration Accuracy Flags Parsing Parsing Overview Phrase profiling Tailoring a CDEP Parser Base Tokenization Classification Reclassification Selection Resolution Register Here Don’t miss this FREE event. Space is limited. Oracle University V Parku 2294/4 148 00 Praha 4 17.6. – 19.6. 2014 09:00 a.m.– 17:30 p.m.

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• Laser range finder, what language to use? Beginner advice

  - by DrOnline

  I hope this is the right place. I am a programming beginner, and I want to make a laser range finder, and I need advice about how to proceed etc. In a few weeks I will get a lot of dirt cheap 3-5V lasers and some cheap usb webcams. I will point the laser and webcam in parallel, and somehow use trigonometry and programming to determined distance. I have seen online that others made done it this way, I have purposefully not looked at the details too much because I want to develop it on my own, and learn, but I know the general outline. I have a general idea of how to proceed. The program loads in a picture from the webcam, and I dunno how images work really, but I imagine there is a format that is basically an array of RGB values.. is this right? I will load in the red values, and find the most red one. I know the height difference between the laser and the cam. I know the center dot in the image, I know the redmost dot. I'm sure there's some way to figure out some range there. TO THE POINT: 1) Is my reasoning sound thus far, especially in terms of image analysis? I don't need complete solutions, just general points 2) What I need to figure out, is what platform to use. I have an arduino... apparently, I've read it's too weak to process images. Read that online. I know some C I know some Python I have Matlab. Which is the best option? I do not need high sampling rates, I have not decided on whether it should be automated or whether I should make a GUI with a button to press for samples. I will keep it simple and expand I think. I also do not need it to be super accurate, I'm just having fun here. Advice!

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• Soft Shadows in Raytracing 3D to 2D

  - by Myx

  Hello: I wish to implement soft shadows produced by area lights in my raytracer. I'm having trouble generating the random samples. So I have a scene in which I have an area light (represented as a circle) whose world (x,y,z) coordinates of the center are given, the radius is given, the normal of the plane on which the circle lies is given, as well as the color and attenuation factors. The sampling scheme I wish to use is the following: generate samples on the quadrilateral that encompasses the circle and discard points outside the circle until the required number of samples within the circle have been found. I'm having trouble understanding how I can transform the 3D coordinates of the center of the circle to its 2D representation (I don't think I can assume that the projection of the circle is on the x-y axis and thus just get rid of the z-component). I think the plane normal information should be used but I'm not sure how. Any and all suggestions are appreciated.

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• Shark was unable to find symbol information for this address range - iPhone

  - by Elliot

  I'm trying to use Shark to determine which method(s) are taking the most time in my iPhone app. After sampling, I get this: Clicking the "!" button yields: Shark was unable to find symbol information for this address range. Typically this happens because the application was compiled without symbols or they have been subsequently stripped away. In Xcode, make sure the "Generate Debug Symbols" checkbox is selected (passes the -g flag to the compiler). Note that this does not affect code optimization, and does not typically alter performance significantly. However, the extra symbol information does consume significantly more space and may bloat the size of the executable. But I AM using the Debug option, and I am running on my Device. And Generate Debug Symbols IS checked. So what's wrong?

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• How can I encode four unsigned bytes (0-255) to a float and back again using HLSL?

  - by Statement

  Hello! I am facing a task where one of my hlsl shaders require multiple texture lookups per pixel. My 2d textures are fixed to 256*256, so two bytes should be sufficient to address any given texel given this constraint. My idea is then to put two xy-coordinates in each float, giving me eight xy-coordinates in pixel space when packed in a Vector4 format image. These eight coordinates are then used to sample another texture(s). The reason for doing this is to save graphics memory and an attempt to optimize processing time, since then I don't require multiple texture lookups. By the way: Does anyone know if encoding/decoding 16 bytes from/to 4 floats using 1 sampling is slower than 4 samplings with unencoded data?

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• Matlab: Analysis of signal

  - by Mateusz

  Hi, I have a problem with this task: For free route perform frequency analysis and give parametrs of each signal component: time of beginning and ending of each component beginning and ending frequency amplitude (in time domain) in the beginning and end of each signal's component level of noise in dB Assume, that, the parametrs of each component like amplitude, frequency is changing lineary in time. Frequency of sampling is 1000Hz For example I have signal like this: Nx=64; fs=1000; t=1/fs*(0:Nx-1); %========================== A1=1; A2=4; f1=500; f2=1000; x1=A1*cos(2*pi*f1*t); x2=A2*sin(2*pi*f2*t); %========================== x=x1+x2;

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• How can I subsample data from a time series with LINQ to SQL?

  - by Chris Farmer

  I have a database table full of time points and experimental values at those time points. I need to retrieve the values for an experiment and create a thumbnail image showing an XY plot of its data. Because the actual data set for each experiment is potentially 100,000 data points and my image is only 100 pixels wide, I want to sample the data before creating the image. My current query (which retrieves all the data without sampling) is something simple like this: var points = from p in db.DataPoints where p.ExperimentId == myExperimentId orderby p.Time select new { X = p.Time, Y = p.Value } So, how can I best take every nth point from my result set in a LINQ to SQL query?

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• Sample the deltas between values using boost::accumulators

  - by Checkers

  I have a data set with N integers (say, 13, 16, 17, 20) where each next sample is incremented by some value (3, 1, 3 in this case) and I want to use boost::accumulators::accumulator_set to find various statistics of the second sequence. I want to be able to do something like this: accumulator_set< double, features< tag::mean > > acc; ... acc(13); acc(16); acc(17); acc(20); ...BUT sampling the differences instead of the actual values. How can I do that with accumulator_set without keeping track of the last value manually?

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• Working examples of winform/WPF apps

  - by melody

  I'm a user interface designer. The project I'm working on is a dashboard that will have several interactive and customizable widgets (think Netvibes, etc.). The development group is going to be using either WinForm or WPF to build the dashboard and widgets. I need to understand the capabilities of each technology better and see a sampling of actual working apps to help accomplish this. I can't download and build examples... need live examples somewhere on the web. Can you point me to live, existing apps? Thx.

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• Smoothing Small Data Set With Second Order Quadratic Curve

  - by Rev316

  I'm doing some specific signal analysis, and I am in need of a method that would smooth out a given bell-shaped distribution curve. A running average approach isn't producing the results I desire. I want to keep the min/max, and general shape of my fitted curve intact, but resolve the inconsistencies in sampling. In short: if given a set of data that models a simple quadratic curve, what statistical smoothing method would you recommend? If possible, please reference an implementation, library, or framework. Thanks SO!

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• Efficient way to create/unpack large bitfields in C?

  - by drhorrible

  I have one microcontroller sampling from a lot of ADC's, and sending the measurements over a radio at a very low bitrate, and bandwidth is becoming an issue. Right now, each ADC only give us 10 bits of data, and its being stored in a 16-bit integer. Is there an easy way to pack them in a deterministic way so that the first measurement is at bit 0, second at bit 10, third at bit 20, etc? To make matters worse, the microcontroller is little endian, and I have no control over the endianness of the computer on the other side.

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• How to improve performance

  - by Ram

  Hi, In one of mine applications I am dealing with graphics objects. I am using open source GPC library to clip/merge two shapes. To improve accuracy I am sampling (adding multiple points between two edges) existing shapes. But before displaying back the merged shape I need to remove all the points between two edges. But I am not able to find an efficient algorithm that will remove all points between two edges which has same slope with minimum CPU utilization. Currently all points are of type PointF Any pointer on this will be a great help.

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• Strange profiling results: definitely non-bottleneck method pops up

  - by jkff

  I'm profiling a program using sampling profiling in YourKit and JProfiler, and also "manually" (I launch it and press Ctrl-Break several times to get thread dumps). All three methods give me extremely strange results: some tens of percents of time spent in a 3-line method that does not even do any allocation or synchronization and doesn't have loops etc. Moreover, after I made this method into a NOP and even removed its invocation completely, the observable program performance didn't change at all (although it got a negligible memory leak, since it was a method for freeing a cheap resource). I'm thinking that this might be because of the constraints that JVM puts on the moments at which a thread's stacktrace may be taken, and it somehow turns out that in my program it is exactly the moments where this method is invoked, although there is absolutely nothing special about it or the context in which it is invoked. What can be the explanation for this phenomenon? What are the aforementioned constraints? What further measurements can I take to clarify the situation?

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• SOS: AudioFormat when writing to file in FreeTTS

  - by user330793

  Very annoying problem. I have developed a freeTTS application of the freetts class that write captured audio to file however I am having some very annoying problems. When setting the audio player to singlefileaudio player I try to also set the audioformat with my own default values for sampleRate, sampleSizeInBits, channels, signed and bigEndian. Now I access AudioPlayer.get methods to show these values in runtime just to ensure they are set to what I set them and they match those values. However when file writing completes and I check the properties of the resulting wave file, they are set to the audioPlayer default settings. Normally this will be fine except I have to read the files into another application which has fixed audio property settings so I always get a resulting output that sounds like am fast forwarding the sound and listening to it at the same time. Obviously because of the different sampling rates. I need help please. Thanx, Henry

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• help with matlab and Discrete Fourier transform

  - by user504363

  Hi all I have previous experience with Matlab, but the problem that I face some problems in apply a problem in (DSP : Digital signal processing) which is not my study field, but I must finish that problems in days to complete my project. all i want is help me with method and steps of solving this problem in matlab and then I can write the code with myself. the problem is about the signal x(t) = exp(-a*t); 1) what's the Discrete Fourier transform of the sampled signal with sample rate fs 2) if a=1 and fs =1 , plot the amplitude spectrum of sampled signal 3) fix the sampling frequency at fs = 1(hz) [what's it mean ?] and plot the magnitude of the Fourier Transform of the sampled signal at various values of a thanks

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• BlackBerry Technical Specification

  - by Sam

  I'm having trouble locating BlackBerry techical specifications and their website is a mess. They also don't have a number that I can use to easily contact them. This isn't exactly a coding question, but what does the BlackBerry audio API look like, and where can I get technical specifications on audio? Specifically, I'm trying to find out more information on Audio-In, specifically, through the Mic-In on the 3.5 mm jack. Unfortunately, before I can proceed, I need to know such things like sampling rate, data width, etc. Direction to the right resource or if you know off of the top of your head is appreciated.

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• Efficiently remove points with same slope

  - by Ram

  Hi, In one of mine applications I am dealing with graphics objects. I am using open source GPC library to clip/merge two shapes. To improve accuracy I am sampling (adding multiple points between two edges) existing shapes. But before displaying back the merged shape I need to remove all the points between two edges. But I am not able to find an efficient algorithm that will remove all points between two edges which has same slope with minimum CPU utilization. Currently all points are of type PointF Any pointer on this will be a great help.

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• C++Math evaluating incorrectly

  - by Hayden

  I thought I can make life a little easier in data statistics by making a small program which returns the results of sampling distribution of the mean (with standard error). It does this part successfully but in an attempt to return the z-score by using the formula I found here, it returns -1#IND. My interpretation of that formula is: ((1/(sqrt(2*pi)*stdev))*pow(e, (normalpow)) where double normalpow=-0.5*((mean-popmean)*(mean-popmean)/stdev) I did a little more investigating and found that (mean-popmean)*(mean-popmean) was evaluating to 0 no matter what. How can I get around this problem of normalpow evaluating to 0.

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• Microphone input

  - by George

  I'm trying to build a gadget that detects pistol shots using Android. It's a part of a training aid for pistol shooters that tells how the shots are distributed in time and I use a HTC Tattoo for testing. I use the MediaRecorder and its getMaxAmplitude method to get the highest amplitude during the last 1/100 s but it does not work as expected; speech gives me values from getMaxAmplitude in the range from 0 to about 25000 while the pistol shots (or shouting!) only reaches about 15000. With a sampling frequency of 8kHz there should be some samples with considerably high level. Anyone who knows how these things work? Are there filters that are applied before registering the max amplitude. If so, is it hardware or software? Thanks, /George

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• Microphone input

  - by George

  I'm trying to build a gadget that detects pistol shots using Android. It's a part of a training aid for pistol shooters that tells how the shots are distributed in time and I use a HTC Tattoo for testing. I use the MediaRecorder and its getMaxAmplitude method to get the highest amplitude during the last 1/100 s but it does not work as expected; speech gives me values from getMaxAmplitude in the range from 0 to about 25000 while the pistol shots (or shouting!) only reaches about 15000. With a sampling frequency of 8kHz there should be some samples with considerably high level. Anyone who knows how these things work? Are there filters that are applied before registering the max amplitude. If so, is it hardware or software? Thanks, /George

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• Pseudorandom Number Generation with Specific Non-Uniform Distributions

  - by carnun

  Hello all, I'm writing a program that simulates various random walks (with differing distributions). At each timestep, I need randomly generated, two dimensional step distances and angles from the distribution of the random walk. I'm hoping someone can check my understanding of how to generate these random numbers. As I understand it I can use Inverse Transform Sampling as follows: If f(x) is the pdf of our random walk that has a non-uniform distribution, and y is a random number from a uniform distribution. Then if we let f(x) = y and solve to find x then we have a random number from the non-uniform distribution. Is this a feasible solution?

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