Search Results

Search found 98 results on 4 pages for 'fft'.

Page 4/4 | < Previous Page | 1 2 3 4 

• Using Robocopy to Backup to a NAS

  - by Your DisplayName here!

  When using Robocopy to backup data to a NAS, I always had the problem that most files were considered “old” on the NAS device (even if they weren’t) – that kind of defeats the purpose of the /MIR switch. Today I finally decided to search for a solution – and it was remarkably easy. Most NAS devices use Samba or something similar to provide “NTFS shares” – but most of them only implement FAT-style file times with a 2-second-granularity. You can force robocopy to use FAT file time as well using the /FFT switch. Now my backup script works again as expected. See also here.

  Read the article

• which method handle frequency calculation in AurioTouch iphone sample application

  - by MS Nathan

  Hi i am doing some research with aurioTouch FFT implementation for one of my audio related project. now i can found the class and the method which calculate the signal strength / power in decibel (dB), but i couldn't find the method from which i can get the frequency. can anyone help me to find which class and its method handling the frequency calculation? thanks in advance

  Read the article

• Determining when a sound above a certain frequency is played on iPhone

  - by eagle

  I'm wondering what's the easiest/best way to determine if a sound, being recorded on the iPhone mic, is above a certain frequency. My guess is that I would need to use the Audio Queue Services since I don't need to record the audio to a file. I'm using SDK 4.0, so I can use the FFT functions in the Accelerate framework if needed.

  Read the article

• Fourier transform software

  - by CFP

  Hello everyone! After spending a lot of time searching for this, I thought that some SuperUser gurus might know the answer :) I'm searching for an open source application to compute an FFT, that could: * Import a list of points from a text file (in any format, I could write conversion scripts if needed), for example 0,1; 1,2; 4,5 * Compute the associated discrete transform, and output the list of coefficients Ideally, it would also display the plot and the associated fourier decomposition on the same graph, to allow comparison, but this is not absolutely needed. It can be either on Windows or on Linux/Unix. Can you think of a solution? Thanks, CFP.

  Read the article

• Is it illegal to rewrite every line of an open source project in a slightly different way, and use it in a closed source project?

  - by Chris Barry

  There is some code which is GPL or LGPL that I am considering using for an iPhone project. If I took that code (JavaScript) and rewrote it in a different language for use on the iPhone would that be a legal issue? In theory the process that has happened is that I have gone through each line of the project, learnt what it is doing, and then reimplemented the ideas in a new language. To me it seems this is like learning how to implement something, but then reimplementing it separately from the original licence. Therefore you have only copied the algorithm, which arguably you could have learnt from somewhere else other than the original project. Does the licence cover the specific implementation or the algorithm as well? EDIT------ Really glad to see this topic create a good conversation. To give a bit more backing to the project, the code involved does some kind of audio analysis. I believe it is non-trivial to learn or implement, although I was prepared to embark on this task (I'm at the level where I can implement an FFT algorithm, and this was going to go beyond that.) It is a fairly low LOC script, so I didn't think it would be too hard to do a straight port. I really like the idea of rereleasing my port as well as using it in the application. I don't see any problem with that, and it would be a great way to give something back to the community. I was going to add a line about not wanting to discuss the moral issues, but I'm quite glad I didn't as it seems to have fired the debate a bit. I still feel a bit odd about using open source code to learn from. Does this mean that anything one learns from an open source project is not allowed to be used in a closed source project? And how long after or different does an implementation have to be to not be considered violation of the licence? Murky! EDIT 2 -------- Follow up question

  Read the article

• How do audio based games such as Audiosurf and Beat Hazard work?

  - by The Communist Duck

  Note: I am not asking how to make a clone of one of these. I am asking about how they work. I'm sure everyone's seen the games where you use your own music files (or provided ones) and the games produce levels based on them, such as Audiosurf and Beat Hazard. Here is a video of Audiosurf in action, to show what I mean. If you provide a heavy metal song, you would get a completely different set of obstacles, enemies, and game experience from something like Vivaldi. What does interest me is how these games work. I do not know much about audio (well, data-side), but how do they process the song to understand when it is settling down or when it's speeding up? I guess they could just feed the pitch values (assuming those sorts of things exist in audio files) to form a level, but it wouldn't fully explain it. I'm either looking for an explanation, some links to articles about this sort of thing (I'm sure there's a term or terms for it), or even an open-source implementation of this kind of thing ;-) EDIT: After some searching and a little help, I found out about FFT (Fast Fourier Transform). This maybe a step in the right direction, but it is something that does not make any sense to me..or fits with my physics knowledge of waves.

  Read the article

• iTunes Visualizer Plugin in C# - Energy Function

  - by James D

  Hi, iTunes Visualizer plugin in C#. Easiest way to compute the "energy level" for a particular sample? I looked at this writeup on beat detection over at GameDev and have had some success with it (I'm not doing beat detection per se, but it's relevant). But ultimately I'm looking for a stupid-as-possible, quick-and-dirty approach for demo purposes. For those who aren't familiar with how iTunes structures visualization data, basically you're given this: struct VisualPluginData { /* SNIP */ RenderVisualData renderData; UInt32 renderTimeStampID; UInt8 minLevel[kVisualMaxDataChannels]; // 0-128 UInt8 maxLevel[kVisualMaxDataChannels]; // 0-128 }; struct RenderVisualData { UInt8 numWaveformChannels; UInt8 waveformData[kVisualMaxDataChannels][kVisualNumWaveformEntries]; // 512-point FFT UInt8 numSpectrumChannels; UInt8 spectrumData[kVisualMaxDataChannels][kVisualNumSpectrumEntries]; }; Ideally, I'd like an approach that a beginning programmer with little to no DSP experience could grasp and improve on. Any ideas? Thanks!

  Read the article

• TMS320C64x Quick start reference for porgrammers

  - by osgx

  Hello Is thare any quickstart guide for programmers for writing DSP-accelerated appliations for TMS320C64x? I have a program with custom algorythm (not the fft, or usial filtering) and I want to accelerate it using multi-DSP coprocessor. So, how should I modify source to move computation from main CPU to DSPs? What limitations are there for DSP-running code? I have some experience with CUDA. In CUDA I should mark every function as being host, device, or entry point for device (kernel). There are also functions to start kernels and to upload/download data to/from GPU. There are also some limitations, for device code, described in CUDA Reference manual. I hope, there is an similar interface and a documentation for DSP.

  Read the article

• DSP - How are frequency amplitudes modified using DFT?

  - by Trap

  I'm trying to implement a DFT-based equalizer (not FFT) for the sole purpose of learning. To check if it works I took an audio signal, analyzed it and then resynthesized it again with no modifications made to the frequency spectrum. So far so good. Now I tried to silence some frequency bands, just by setting their amplitudes to zero before resynthesis, but definitely it's not the way to go. What I get is a rather distorted signal. I'm using the so-called 'standard way of calculating the DFT' which is by correlation. I first tried to modify the real part amplitudes only, then modifying both the real and imaginary part amplitudes. I also tried to convert the DFT output to polar notation, then modifying the magnitude and convert back to rectangular notation, but none of this is working. Can someone show me what I'm doing wrong? I tried to find info on this subject in the internet but couldn't find any. Thanks in advance.

  Read the article

• How do we calculate the filters in Mel-Frequency Cepstrum Coefficients Algorithm?

  - by André Ferreira

  After calculating the FFT and with the frequency we need to do something like this: http://instruct1.cit.cornell.edu/courses/ece576/FinalProjects/f2008/pae26%5Fjsc59/pae26%5Fjsc59/images/melfilt.png We filter the frequency spectrum with those triangles. I saw that we can use distint ways to calculcate the triangles. I will make the size of the triangles equal till 1kz and after that obtained with log function. What should we do now? With the frequency spectrum and the triangles defined.. - We should filter the frequency (frequencies limited to the triangles, if goes higher only counts till the triangle limit) and calculate the value of each triangle (and after that continue the algorithm). But when does the mel conversation happens? m = 2595 log (f/700 + 1) When do we pass from frequency to mel.. Can someone guide me in the right direction plz :d

  Read the article

• Using Cepstrum for PDA

  - by CziX

  Hey, I am currently deleveloping a algorithm to decide wheather or not a frame is voiced or unvoiced. I am trying to use the Cepstrum to discriminate between these two situations. I use MATLAB for my implementation. I have some problems, saying something generally about the frame, but my currently implementation looks like (I'm award of the MATLAB has the function rceps, but this haven't worked for either): ceps = abs(ifft(log10(abs(fft(frame.*window')).^2+eps))); Can anybody give me a small demo, that will convert the frame to the power cepstrum, so a single lollipop at the pitch frequency. For instance use this code to generate the frequency. fs = 8000; timelength = 25e-3; freq = 500; k = 0:1/fs:timelength-(1/fs); s = 0.8*sin(2*pi*freq*k); Thanks.

  Read the article

• What language/API to use for a standalone live-input audio visualizer app?

  - by knuckfubuck

  I develop with Actionscript and was glad to see that AIR 2.0 was going to give access to mic input data. I planned to use this to create a visualizer set to the tempo of the incoming live audio. After doing a few days of google research it seems unlikely that it will be possible to analyze the data of the mic input in Flash/AIR. If anyone has ideas on how I can achieve this in AIR please let me know. (I'm open to workarounds.) That being said, I don't want to give up on the idea so I'm interested in suggestions for other language/API to use. My requirements for the app are: Run on OSX Two windows - one that can go fullscreen while the other(controller GUI) stays put Able to access live mic input data I've done reading on FFT and understand what needs to be done on the sound side so no need to help with that.

  Read the article

• Audio processing in C# or C++

  - by melculetz

  Hi, I would like to create an application that uses AI techniques and allows the user to record a part of a song and then tries to find that song in a database of wav files. I would have liked to use some already existing libraries for the audio processing part. So, could you recommend any libraries in C# which can read a wav file, get input from microphone, have some audio filters (low pass, high pass, FFT etc) and maybe have the ability to plot the audio signal as well. I would prefer to develop in C#, but if there aren't good libraries for audio processing, I guess I could work in C++ as well. As far as I know, Mathlab already has the above mentioned functionalities, but I can't use it in my application.

  Read the article

• Redundancy algorithm for reading noisy bitstream

  - by Tedd Hansen

  I'm reading a lossy bit stream and I need a way to recover as much usable data as possible. There can be 1's in place of 0's and 0's in palce of 1's, but accuracy is probably over 80%. A bonus would be if the algorithm could compensate for missing/too many bits as well. The source I'm reading from is analogue with noise (microphone via FFT), and the read timing could vary depending on computer speed. I remember reading about algorithms used in CD-ROM's doing this in 3? layers, so I'm guessing using several layers is a good option. I don't remember the details though, so if anyone can share some ideas that would be great! :) Edit: Added sample data Best case data: in: 0000010101000010110100101101100111000000100100101101100111000000100100001100000010000101110101001101100111000101110000001001111011001100110000001001100111011110110101011100111011000100110000001000010111 out: 0010101000010110100101101100111000000100100101101100111000000100100001100000010000101110101001101100111000101110000001001111011001100110000001001100111011110110101011100111011000100110000001000010111011 Bade case (timing is off, samples are missing): out: 00101010000101101001011011001110000001001001011011001110000001001000011000000100001011101010011011001 in: 00111101001011111110010010111111011110000010010000111000011101001101111110000110111011110111111111101 Edit2: I am able to controll the data being sent. Currently attempting to implement simple XOR checking (though it won't be enough).

  Read the article

• CUDA small kernel 2d convolution - how to do it

  - by paulAl

  I've been experimenting with CUDA kernels for days to perform a fast 2D convolution between a 500x500 image (but I could also vary the dimensions) and a very small 2D kernel (a laplacian 2d kernel, so it's a 3x3 kernel.. too small to take a huge advantage with all the cuda threads). I created a CPU classic implementation (two for loops, as easy as you would think) and then I started creating CUDA kernels. After a few disappointing attempts to perform a faster convolution I ended up with this code: http://www.evl.uic.edu/sjames/cs525/final.html (see the Shared Memory section), it basically lets a 16x16 threads block load all the convolution data he needs in the shared memory and then performs the convolution. Nothing, the CPU is still a lot faster. I didn't try the FFT approach because the CUDA SDK states that it is efficient with large kernel sizes. Whether or not you read everything I wrote, my question is: how can I perform a fast 2D convolution between a relatively large image and a very small kernel (3x3) with CUDA?

  Read the article

• MATLAB: Automatic detection of relations between workspace variables through functions

  - by Peterstone

  Hi all, I´m trying to write a function what detect this relation between the variables I have got in the workspace: v1 - fft(v2) = 0 Where v1, v2 are variables of my workspace. Sometimes I need to know which variables have a certain numerical relation. If I have thirty, I don´t want to be looking for this relation in "manual way", just itroducing a sentence for each pair of different variables. I would like a function in which I introuce (or I modify this function every time I need it) the sentence (for instance what I wrote before) and the function show me the pair of variables a I am looking for. Does anyone know how to do it? Thank you so much!

  Read the article

• Linux, C++ audio capturing (just microphone) library

  - by TheOm3ga

  I'm developing a musical game, it's like a singstar but instead of singing, you have to play the recorder. It's called oFlute, and it's still in early development stage. In the game, I capture the microphone input, then run a simple FFT analysis and compare the results to typical recorder's frequencies, thus getting the played note. At the beginning, the audio library I was using was RtAudio, but I don't remember why I switched to PortAudio, which is what I'm currently using. The problem is that, from time to time, either it crashes randomly or stops capturing, like if there were no sound coming from the microphone. My question is, what's the best option to capture microphone input on Linux? I just need to open, read, and close a flow of bytes from the microphone. I've been reading this guide, and (un)surprisingly it says: I don't think that PortAudio is very good API for Unix-like operating systems. So, what do you recommend me?

  Read the article

• BAT file will not run from Task Scheduler but will from Command Line

  - by wtaylor

  I'm trying to run a BAT script from Task Scheduler in Windows 2008 R2 and it runs for 3 seconds and then stops. It says it successfully completes but I know it doesn't. I can run this script from the command line directly, and it runs just fine. The bat file I'm running actually deletes files older than 7 days using "forfiles" then I'm mapping a network drive, moving the files across the network using robocopy, and then closing the network connection. I have taken the network and copy options out of the file and it still does the same thing. Here is how my file looks: rem This will delete the files from BBLEARN_stats forfiles -p "E:\BB_Maintenance_Data\DB_Backups\BBLEARN_stats" -m *.* -d -17 -c "cmd /c del @file" rem This will delete the files from BBLEARN_cms_doc forfiles -p "E:\BB_Maintenance_Data\DB_Backups\BBLEARN_cms_doc" -m *.* -d -14 -c "cmd /c del @path" rem This will delete the files from BBLEARN_admin forfiles -p "E:\BB_Maintenance_Data\DB_Backups\BBLEARN_admin" -m *.* -d -10 -c "cmd /c del @path" rem This will delete the files from BBLEARN_cms forfiles -p "E:\BB_Maintenance_Data\DB_Backups\BBLEARN_cms" -m *.* -d -10 -c "cmd /c del @path" rem This will delete the files from attendance_bb forfiles -p "E:\BB_Maintenance_Data\DB_Backups\attendance_bb" -m *.* -d -10 -c "cmd /c del @path" rem This will delete the files from BBLearn forfiles -p "E:\BB_Maintenance_Data\DB_Backups\BBLEARN" -m *.* -d -18 -c "cmd /c del @path" rem This will delete the files from Logs forfiles -p "E:\BB_Maintenance_Data\logs" -m *.* -d -10 -c "cmd /c del @path" NET USE Z: \\10.20.102.225\coursebackups\BB_DB_Backups /user:cie oly2008 ROBOCOPY E:\BB_Maintenance_Data Z: /e /XO /FFT /PURGE /NP /LOG:BB_DB_Backups.txt openfiles /disconnect /id * NET USE Z: /delete /y This is happening on 2 servers when trying to run commands from inside a BAT file. The other server is giving an error if (0xFFFFFFFF) but that file is running a CALL C:\dir\dir\file.bat -options and I've used commands like that before in Server 2003. Here is the file for this file: call C:\blackboard\apps\content-exchange\bin\batch_ImportExport.bat -f backup_batch_file.txt -l 1 -t archive NET USE Z: \\10.20.102.225\coursebackups\BB_Course_Backups /user:cie oly2008 ROBOCOPY E:\ Z: /move /e /LOG+:BB_Move_Course_Backups.txt openfiles /disconnect /id * NET USE Z: /delete /y Any help would be GREAT. Thanks

  Read the article

• How to write C++ audio processing applications?

  - by cesko82

  Hi everyone, I'm an Electronics and Telecommunications student, next to my graduation. I'm gonna work on a project that involves my knowledge about DSP, music and audio in general. I allready know all the basic mathematic instruments and all the stuff I need to manage it, such as FFT, circular convolution ecc ecc. I want to learn C++ programming basically for one reason: it's very important in the professional world!!! And I think it's one of the most used to write applications working with audio, especially when it's about real time processing. Ok, after this small introduction I would like to know first, which are the most used libraries to work with audio processing in c++?? I was longer looking on the web but i couldn't find a lo of working stuff. (I work under linux with eclipse CDT enviroment). Then I would like to know if there are good sources to learn how to write some working code, such as for example how to write a simple low pass filter. Basically now i will not write real time applications, I would like to start from the processing of a WAV file, or even better an MP3 file, so basically on vectors of samples. Let's say that basically for now I would like to extract the waveform from an audio file, and save it to a thumbnail or to a PNG image. Ok, for now I think it's all I would need. Any ideas, advices, libraries, books, interesting sources about that? Thanks a lot in advance for any kind of answer. Giovanni.

  Read the article

• MKL Accelerated Math Libraries for Java...

  - by Kaopua

  I've looked at the related threads on StackOverflow and Googled with not much luck. I'm also very new to Java (I'm coming from a C# and .NET background) so please bear with me. There is so much available in the Java world it's pretty overwhelming. I'm starting on a new Java-on-Linux project that requires some heavy and highly repetitious numerical calculations (i.e. statistics, FFT, Linear Algebra, Matrices, etc.). So maximizing the performance of the mathematical operations is a requirement, as is ensuring the math is correct. So hence I have an interest in finding a Java library that perhaps leverages native acceleration such as MKL, and is proven (so commercial options are definitely a possibility here). In the .NET space there are highly optimized and MKL accelerated commercial Mathematical libraries such as Centerspace NMath and Extreme Optimization. Is there anything comparable in Java? Most of the math libraries I have found for Java either do not seem to be actively maintained (such as Colt) or do not appear to leverage MKL or other native acceleration (such as Apache Commons Math). I have considered trying to leverage MKL directly from Java myself (e.g. JNI), but me being new to Java (let alone interoperating between Java and native libraries) it seemed smarter finding a Java library that has already done this correctly, efficiently, and is proven. Again I apologize if I am mistaken or misguided (even in regarding any libraries I've mentioned) and my ignorance of the Java offerings. It's a whole new world for me coming from the heavily commercialized Microsoft stock so I could easily be mistaken on where to look and regarding the Java libraries I've mentioned. I would greatly appreciate any help or advice.

  Read the article

• Is there a recommended way to communicate scientific/engineering programming to C developers?

  - by ggkmath

  Hi, I have a lot of MATLAB code that needs to get ported to C (execution speed is critical for this work) as part of a back-end process for a web application. When I attempt to outsource this code to a C developer, I assume (correct me if I'm wrong) few C developers also understand MATLAB code (things like indexing and memory management are different, etc.). I wonder if there are any C developers out there that can recommend a procedure for me to follow to best communicate what the code does? For example, should I provide the MATLAB code and explain what it's doing line by line? Or, should I just provide the math/algorithm, explain it in plain English, and let the C developer implement it with this understanding in his/her own way (e.g. can I assume the developer understands how to work with complex math (i.e. imaginary numbers), how to generate histograms, perform an FFT, etc.)? Or, is there a better method? I expect I'm not the first to need to do this, so I wonder if any C developers out there ran into this situation and can share any conventional wisdom how they'd like this task to be transferred? Thanks in advance for any comments.

  Read the article

• Generic applet style system for publishing mathematics demonstrations?

  - by Alex

  Anyone who's tried to study mathematics using online resources will have come across these Java applets that demonstrate a particular mathematical idea. Examples: http://www.math.ucla.edu/~tao/java/Mobius.html http://www.mathcs.org/java/programs/FFT/index.html I love the idea of this interactive approach because I believe it is very helpful in conveying mathematical principles. I'd like to create a system for visually designing and publishing these 'mathlets' such that they can be created by teachers with little programming experience. So in order to create this app, i'll need a GUI and a 'math engine'. I'll probably be working with .NET because thats what I know best and i'd like to start experimenting with F#. Silverlight appeals to me as a presentation framework for this project (im not worried about interoperability right now). So my questions are: does anything like this exist already in full form? are there any GUI frameworks for displaying mathematical objects such as graphs & equations? are there decent open source libraries that exposes a mathematical framework (Math.NET looks good, just wondering if there is anything else out there) is there any existing work on taking mathematical models/demos built with maple/matlab/octave/mathematica etc and publishing them to the web?

  Read the article

• Writing a managed wrapper for unmanaged (C++) code - custom types/structs

  - by Bobby

  faacEncConfigurationPtr FAACAPI faacEncGetCurrentConfiguration( faacEncHandle hEncoder); I'm trying to come up with a simple wrapper for this C++ library; I've never done more than very simple p/invoke interop before - like one function call with primitive arguments. So, given the above C++ function, for example, what should I do to deal with the return type, and parameter? FAACAPI is defined as: #define FAACAPI __stdcall faacEncConfigurationPtr is defined: typedef struct faacEncConfiguration { int version; char *name; char *copyright; unsigned int mpegVersion; unsigned long bitRate; unsigned int inputFormat; int shortctl; psymodellist_t *psymodellist; int channel_map[64]; } faacEncConfiguration, *faacEncConfigurationPtr; AFAIK this means that the return type of the function is a reference to this struct? And faacEncHandle is: typedef struct { unsigned int numChannels; unsigned long sampleRate; ... SR_INFO *srInfo; double *sampleBuff[MAX_CHANNELS]; ... double *freqBuff[MAX_CHANNELS]; double *overlapBuff[MAX_CHANNELS]; double *msSpectrum[MAX_CHANNELS]; CoderInfo coderInfo[MAX_CHANNELS]; ChannelInfo channelInfo[MAX_CHANNELS]; PsyInfo psyInfo[MAX_CHANNELS]; GlobalPsyInfo gpsyInfo; faacEncConfiguration config; psymodel_t *psymodel; /* quantizer specific config */ AACQuantCfg aacquantCfg; /* FFT Tables */ FFT_Tables fft_tables; int bitDiff; } faacEncStruct, *faacEncHandle; So within that struct we see a lot of other types... hmm. Essentially, I'm trying to figure out how to deal with these types in my managed wrapper? Do I need to create versions of these types/structs, in C#? Something like this: [StructLayout(LayoutKind.Sequential)] struct faacEncConfiguration { uint useTns; ulong bitRate; ... } If so then can the runtime automatically "map" these objects onto eachother? And, would I have to create these "mapped" types for all the types in these return types/parameter type hierarchies, all the way down until I get to all primitives? I know this is a broad topic, any advice on getting up-to-speed quickly on what I need to learn to make this happen would be very much appreciated! Thanks!

  Read the article

< Previous Page | 1 2 3 4