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  • Silverlight 3 Data Binding: Imperative and Mixed Approaches

    In the first part of this multi-part series on data binding in Silverlight we learned how to use the declarative XAML syntax approach. In this second part we ll learn how to use the imperative approach and how to combine the two.... Test Drive the Next Wave of Productivity Find Microsoft Office 2010 and SharePoint 2010 trials, demos, videos, and more.

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  • Using ASP.NET 3.5 ListView in a Web Application

    This tutorial will show an example of how to use the ListView web control featuring data updating and validation before the data is inserted or updated to the MS SQL server database. Examples of how to use ListView controls to retrieve information from the data are featured in the first part of this tutorial which appeared yesterday.... Test Drive the Next Wave of Productivity Find Microsoft Office 2010 and SharePoint 2010 trials, demos, videos, and more.

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  • A tale of two (and more) apps

    Robert Cooper gave a great lightning talk at our recent Atlanta GTUG meetup, where he discussed using a single codebase to target multiple mediums (e.g. Android, Facebook, Wave...

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  • Using the FormView Web Control in ASP.NET 3.5

    A FormView web control works much like a DetailsView web control it will display one record at a time to the browser from the database. The difference is that FormView is a template-based layout for which a developer can make detailed changes that affect the final output when rendered in the browser. This tutorial will explain how it works and walk you through setting up a FormView web control.... Test Drive the Next Wave of Productivity Find Microsoft Office 2010 and SharePoint 2010 trials, demos, videos, and more.

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  • How one could use a live editor

    - by Sathvik
    I was thinking about a live editing environment where code / a source file is synchronized so that, changes made by one user would be carried across to all others editing the file. Something like Google Wave, but for code. Could this kind of an environment be better for the code, as changes are shared instantly? (with revision-control, of course) Has anyone tried (or has had a need for) using a shared environment for code?

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  • Convert MySQL to an MS SQL Server 2008 Database

    Converting a MySQL database to an MS SQL Server 2 8 database is a bit tricky. It is however an important database migration conversion. Is there some way to do it without resorting to costly database conversion software or facing issues with ODBC connectivity This article will teach you a new method to help you accomplish this conversion.... Test Drive the Next Wave of Productivity Find Microsoft Office 2010 and SharePoint 2010 trials, demos, videos, and more.

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  • Is SEO Dead?

    Google has release yet another update - Caffeine, and with it comes the next wave of people claiming that SEO is dead. This happens almost once a year, it seems. Not Google's updates, but the clamoring for the death of SEO. Let's examine some reasons why people think it this time around, and whether or not SEO is really dead.

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  • Data Binding in ASP.NET 3.5 with a Basic Example

    Data binding is a method of binding ASP.NET 3.5 web controls and the database column fields. This method of binding is necessary to produce a certain level of interactivity within the web control. This article will explain how and why to use data binding in your web applications.... Test Drive the Next Wave of Productivity Find Microsoft Office 2010 and SharePoint 2010 trials, demos, videos, and more.

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  • Bind to Collection Objects with Silverlight 3

    In this third part of the series we will continue to discuss Silverlight 3 data binding. This time however we ll cover more complex topics such as how to bind to collection objects. The fourth and fifth parts will cover how to deal with the validation during data binding not to mention the possible data conversion .... Test Drive the Next Wave of Productivity Find Microsoft Office 2010 and SharePoint 2010 trials, demos, videos, and more.

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  • Get Started with Silverlight 3 Data Binding

    If you ve learned about data binding from other Microsoft technologies you ll be glad to hear that Silverlight 3 also gives you a smooth way to handle data binding. This article the first one in a multi-part series gets you started by teaching you some of the techniques you ll need to handle data binding successfully.... Test Drive the Next Wave of Productivity Find Microsoft Office 2010 and SharePoint 2010 trials, demos, videos, and more.

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  • Validate Data Binding with Silverlight 3

    In this fourth part of the series we will take a look at how to validate data binding. We ll start by explaining why this is important and then walk through a step-by-step process that shows you how to do it. The next and final part of the series will discuss data conversion.... Test Drive the Next Wave of Productivity Find Microsoft Office 2010 and SharePoint 2010 trials, demos, videos, and more.

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  • Mobile Broadband Connection terminating automatically after 2-3 minutes

    - by HussainNagri
    I am connecting my mobile device with Ubuntu to access INTERNET, It connects easily but after 2-3 minutes it automatically disconnect and then I have to disconnect my mobile device and then reconnect it again to establish a connection. Cant understand what is the problem. No problem with my device as it works flawless with windows. As asked here is more info: Device is Samsung wave 575 Network Provider is Vodafone and Ubuntu is 12.04

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  • How do I lower the hardware volume? (volume too high)

    - by Zom-B
    I have a 4yo Dell laptop with Windows XP Pro (modern ones unfortunately don't have a physical volume knob), and lately I'm using my Apple earphones, because they have much better low frequency response than my $10 earphones. They also have the side effect of being much louder. To give an indication of my agony, for most tasks (movie, music, games) I have my main volume at 3 ticks: drag to 0 with the mouse and press the up key 3 times (the handle does not even raise 1 pixel) and my wave volume at 50%. I notice that when I do this, I have a lot of digital noise, because I'm using just a tiny fraction of the 16-bit space. If I drag the Wave slider down until I barely hear the audio, it becomes really distorted and noisy, indicating that this is digital volume (in the DirectSound driver or something) and not hardware volume. I experimented in Audition. When I make a tone of 1000Hz at -50db, (all windows volumes at max) the volume is just below my pain threshold. When I zoom in to see how high the sample values reach, I see that just 8 of the 16 bits are used (about -100 ~ 100). When I generate such tone at -80db (minimum I can specify) then I can still clearly hear the tone, although really noisy. When I zoom in, I see that just 3 out of 16 bits are used. I created a squarewave tone that is just 1 bit high, and I can still hear it! For most uses, this is not a big problem (audiophiles will disagree!), as I just have more noise than usual (about the same as old 8 bit hardware), but I'm also in the process of programming a hearing test program, in which case this problem is a death blow as the test subjects will even hear tones at the bottom of the theoretical range (lowering the windows volume is futile, see above) (I cannot update drivers, as Dell has discontinued XP support for my model)

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  • Using sound forge 6.0 what will be the need to upgrade to latest version

    - by Jayapal Chandran
    I had been using sound forge 6.0 not recently but long back. I edit mp3 files for my purpose and some more filters like flange, pan, fade in out, recording, line in recording, extracting sound from video files (mpg, avi(divx), etc...), increasing the default volume, editing treble and bass effects, and etc... I am not going to use it professionally. I use it just like that. Now when i checked i could see Sound Forge Audio Studio 10 is the latest version for my purpose. Others are too high i think. Besides, i had been using Gold Wave version 4 very extensively just to edit sound files mostly mp3. and here is the reason for me to change to sound forge. It is when we edit mp3 files it deflashes(making it raw i think) before editing. after editing if i save it asks for the format to save and i will choose mp3. At this point it again applies the compression process which makes the sound file lossy. When i did the same with sound forge it did not deflash. It just edited the file as mp3. May be i dont know whether gold wave has the same option. So, please suggest. oh i had asked a question already like this... here it is goldwave vs sound forge in editing mp3 files

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  • Autocorrelation returns random results with mic input (using a high pass filter)

    - by Niall
    Hello, Sorry to ask a similar question to the one i asked before (FFT Problem (Returns random results)), but i've looked up pitch detection and autocorrelation and have found some code for pitch detection using autocorrelation. Im trying to do pitch detection of a users singing. Problem is, it keeps returning random results. I've got some code from http://code.google.com/p/yaalp/ which i've converted to C++ and modified (below). My sample rate is 2048, and data size is 1024. I'm detecting pitch of both a sine wave and mic input. The frequency of the sine wave is 726.0, and its detecting it to be 722.950820 (which im ok with), but its detecting the pitch of the mic as a random number from around 100 to around 1050. I'm now using a High pass filter to remove the DC offset, but it's not working. Am i doing it right, and if so, what else can i do to fix it? Any help would be greatly appreciated! double* doHighPassFilter(short *buffer) { // Do FFT: int bufferLength = 1024; float *real = malloc(bufferLength*sizeof(float)); float *real2 = malloc(bufferLength*sizeof(float)); for(int x=0;x<bufferLength;x++) { real[x] = buffer[x]; } fft(real, bufferLength); for(int x=0;x<bufferLength;x+=2) { real2[x] = real[x]; } for (int i=0; i < 30; i++) //Set freqs lower than 30hz to zero to attenuate the low frequencies real2[i] = 0; // Do inverse FFT: inversefft(real2,bufferLength); double* real3 = (double*)real2; return real3; } double DetectPitch(short* data) { int sampleRate = 2048; //Create sine wave double *buffer = malloc(1024*sizeof(short)); double amplitude = 0.25 * 32768; //0.25 * max length of short double frequency = 726.0; for (int n = 0; n < 1024; n++) { buffer[n] = (short)(amplitude * sin((2 * 3.14159265 * n * frequency) / sampleRate)); } doHighPassFilter(data); printf("Pitch from sine wave: %f\n",detectPitchCalculation(buffer, 50.0, 1000.0, 1, 1)); printf("Pitch from mic: %f\n",detectPitchCalculation(data, 50.0, 1000.0, 1, 1)); return 0; } // These work by shifting the signal until it seems to correlate with itself. // In other words if the signal looks very similar to (signal shifted 200 data) than the fundamental period is probably 200 data // Note that the algorithm only works well when there's only one prominent fundamental. // This could be optimized by looking at the rate of change to determine a maximum without testing all periods. double detectPitchCalculation(double* data, double minHz, double maxHz, int nCandidates, int nResolution) { //-------------------------1-------------------------// // note that higher frequency means lower period int nLowPeriodInSamples = hzToPeriodInSamples(maxHz, 2048); int nHiPeriodInSamples = hzToPeriodInSamples(minHz, 2048); if (nHiPeriodInSamples <= nLowPeriodInSamples) printf("Bad range for pitch detection."); if (1024 < nHiPeriodInSamples) printf("Not enough data."); double *results = new double[nHiPeriodInSamples - nLowPeriodInSamples]; //-------------------------2-------------------------// for (int period = nLowPeriodInSamples; period < nHiPeriodInSamples; period += nResolution) { double sum = 0; // for each sample, find correlation. (If they are far apart, small) for (int i = 0; i < 1024 - period; i++) sum += data[i] * data[i + period]; double mean = sum / 1024.0; results[period - nLowPeriodInSamples] = mean; } //-------------------------3-------------------------// // find the best indices int *bestIndices = findBestCandidates(nCandidates, results, nHiPeriodInSamples - nLowPeriodInSamples - 1); //note findBestCandidates modifies parameter // convert back to Hz double *res = new double[nCandidates]; for (int i=0; i < nCandidates;i++) res[i] = periodInSamplesToHz(bestIndices[i]+nLowPeriodInSamples, 2048); double pitch2 = res[0]; free(res); free(results); return pitch2; } /// Finds n "best" values from an array. Returns the indices of the best parts. /// (One way to do this would be to sort the array, but that could take too long. /// Warning: Changes the contents of the array!!! Do not use result array afterwards. int* findBestCandidates(int n, double* inputs,int length) { //int length = inputs.Length; if (length < n) printf("Length of inputs is not long enough."); int *res = new int[n]; double minValue = 0; for (int c = 0; c < n; c++) { // find the highest. double fBestValue = minValue; int nBestIndex = -1; for (int i = 0; i < length; i++) { if (inputs[i] > fBestValue) { nBestIndex = i; fBestValue = inputs[i]; } } // record this highest value res[c] = nBestIndex; // now blank out that index. if(nBestIndex!=-1) inputs[nBestIndex] = minValue; } return res; } int hzToPeriodInSamples(double hz, int sampleRate) { return (int)(1 / (hz / (double)sampleRate)); } double periodInSamplesToHz(int period, int sampleRate) { return 1 / (period / (double)sampleRate); } Thanks, Niall. Edit: Changed the code to implement a high pass filter with a cutoff of 30hz (from What Are High-Pass and Low-Pass Filters?, can anyone tell me how to convert the low-pass filter using convolution to a high-pass one?) but it's still returning random results. Plugging it into a VST host and using VST plugins to compare spectrums isn't an option to me unfortunately.

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  • How to store a shmup level?

    - by pek
    I am developing a 2D shmup (i.e. Aero Fighters) and I was wondering what are the various ways to store a level. Assuming that enemies are defined in their own xml file, how would you define when an enemy spawns in the level? Would it be based on time? Updates? Distance? Currently I do this based on "level time" (the amount of time the level is running - pausing doesn't update the time). Here is an example (the serialization was done by XNA): <?xml version="1.0" encoding="utf-8"?> <XnaContent xmlns:level="pekalicious.xanor.XanorContentShared.content.level"> <Asset Type="level:Level"> <Enemies> <Enemy> <EnemyType>data/enemies/smallenemy</EnemyType> <SpawnTime>PT0S</SpawnTime> <NumberOfSpawns>60</NumberOfSpawns> <SpawnOffset>PT0.2S</SpawnOffset> </Enemy> <Enemy> <EnemyType>data/enemies/secondenemy</EnemyType> <SpawnTime>PT0S</SpawnTime> <NumberOfSpawns>10</NumberOfSpawns> <SpawnOffset>PT0.5S</SpawnOffset> </Enemy> <Enemy> <EnemyType>data/enemies/secondenemy</EnemyType> <SpawnTime>PT20S</SpawnTime> <NumberOfSpawns>10</NumberOfSpawns> <SpawnOffset>PT0.5S</SpawnOffset> </Enemy> <Enemy> <EnemyType>data/enemies/boss1</EnemyType> <SpawnTime>PT30S</SpawnTime> <NumberOfSpawns>1</NumberOfSpawns> <SpawnOffset>PT0S</SpawnOffset> </Enemy> </Enemies> </Asset> </XnaContent> Each Enemy element is basically a wave of specific enemy types. The type is defined in EnemyType while SpawnTime is the "level time" this wave should appear. NumberOfSpawns and SpawnOffset is the number of enemies that will show up and the time it takes between each spawn respectively. This could be a good idea or there could be better ones out there. I'm not sure. I would like to see some opinions and ideas. I have two problems with this: spawning an enemy correctly and creating a level editor. The level editor thing is an entirely different problem (which I will probably post in the future :P). As for spawning correctly, the problem lies in the fact that I have a variable update time and so I need to make sure I don't miss an enemy spawn because the spawn offset is too small, or because the update took a little more time. I kinda fixed it for the most part, but it seems to me that the problem is with how I store the level. So, any ideas? Comments? Thank you in advance.

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  • C++ property system interface for game editors (reflection system)

    - by Cristopher Ismael Sosa Abarca
    I have designed an reusable game engine for an project, and their functionality is like this: Is a completely scripted game engine instead of the usual scripting languages as Lua or Python, this uses Runtime-Compiled C++, and an modified version of Cistron (an component-based programming framework).to be compatible with Runtime-Compiled C++ and so on. Using the typical GameObject and Component classes of the Component-based design pattern, is serializable via JSON, BSON or Binary useful for selecting which objects will be loaded the next time. The main problem: We want to use our custom GameObjects and their components properties in our level editor, before used hardcoded functions to access GameObject base class virtual functions from the derived ones, if do you want to modify an property specifically from that class you need inside into the code, this situation happens too with the derived classes of Component class, in little projects there's no problem but for larger projects becomes tedious, lengthy and error-prone. I've researched a lot to find a solution without luck, i tried with the Ogitor's property system (since our engine is Ogre-based) but we find it inappropiate for the component-based design and it's limited only for the Ogre classes and can lead to performance overhead, and we tried some code we find in the Internet we tested it and worked a little but we considered the macro and lambda abuse too horrible take a look (some code omitted): IWE_IMPLEMENT_PROP_BEGIN(CBaseEntity) IWE_PROP_LEVEL_BEGIN("Editor"); IWE_PROP_INT_S("Id", "Internal id", m_nEntID, [](int n) {}, true); IWE_PROP_LEVEL_END(); IWE_PROP_LEVEL_BEGIN("Entity"); IWE_PROP_STRING_S("Mesh", "Mesh used for this entity", m_pModelName, [pInst](const std::string& sModelName) { pInst->m_stackMemUndoType.push(ENT_MEM_MESH); pInst->m_stackMemUndoStr.push(pInst->getModelName()); pInst->setModel(sModelName, false); pInst->saveState(); }, false); IWE_PROP_VECTOR3_S("Position", m_vecPosition, [pInst](float fX, float fY, float fZ) { pInst->m_stackMemUndoType.push(ENT_MEM_POSITION); pInst->m_stackMemUndoVec3.push(pInst->getPosition()); pInst->saveState(); pInst->m_vecPosition.Get()[0] = fX; pInst->m_vecPosition.Get()[1] = fY; pInst->m_vecPosition.Get()[2] = fZ; pInst->setPosition(pInst->m_vecPosition); }, false); IWE_PROP_QUATERNION_S("Orientation (Quat)", m_quatOrientation, [pInst](float fW, float fX, float fY, float fZ) { pInst->m_stackMemUndoType.push(ENT_MEM_ROTATE); pInst->m_stackMemUndoQuat.push(pInst->getOrientation()); pInst->saveState(); pInst->m_quatOrientation.Get()[0] = fW; pInst->m_quatOrientation.Get()[1] = fX; pInst->m_quatOrientation.Get()[2] = fY; pInst->m_quatOrientation.Get()[3] = fZ; pInst->setOrientation(pInst->m_quatOrientation); }, false); IWE_PROP_LEVEL_END(); IWE_IMPLEMENT_PROP_END() We are finding an simplified way to this, without leading confusing the programmers, (will be released to the public) i find ways to achieve this but they are only available for the common scripting as Lua or editors using C#. also too portable, we can write "wrappers" for different GUI toolkits as Qt or GTK, also i'm thinking to using Boost.Wave to get additional macro functionality without creating my own compiler. The properties designed to use in the editor they are removed in the game since the save file contains their data and loads it using an simple 'load' function to reduce unnecessary code bloat may will be useful if some GameObject property wants to be hidden instead. In summary, there's a way to implement an reflection(property) system for a level editor based in properties from derived classes? Also we can use C++11 and Boost (restricted only to Wave and PropertyTree)

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  • record output sound in python

    - by aaronstacy
    i want to programatically record sound coming out of my laptop in python. i found PyAudio and came up with the following program that accomplishes the task: import pyaudio, wave, sys chunk = 1024 FORMAT = pyaudio.paInt16 CHANNELS = 1 RATE = 44100 RECORD_SECONDS = 5 WAVE_OUTPUT_FILENAME = sys.argv[1] p = pyaudio.PyAudio() channel_map = (0, 1) stream_info = pyaudio.PaMacCoreStreamInfo( flags = pyaudio.PaMacCoreStreamInfo.paMacCorePlayNice, channel_map = channel_map) stream = p.open(format = FORMAT, rate = RATE, input = True, input_host_api_specific_stream_info = stream_info, channels = CHANNELS) all = [] for i in range(0, RATE / chunk * RECORD_SECONDS): data = stream.read(chunk) all.append(data) stream.close() p.terminate() data = ''.join(all) wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb') wf.setnchannels(CHANNELS) wf.setsampwidth(p.get_sample_size(FORMAT)) wf.setframerate(RATE) wf.writeframes(data) wf.close() the problem is i have to connect the headphone jack to the microphone jack. i tried replacing these lines: input = True, input_host_api_specific_stream_info = stream_info, with these: output = True, output_host_api_specific_stream_info = stream_info, but then i get this error: Traceback (most recent call last): File "./test.py", line 25, in data = stream.read(chunk) File "/Library/Python/2.5/site-packages/pyaudio.py", line 562, in read paCanNotReadFromAnOutputOnlyStream) IOError: [Errno Not input stream] -9975 is there a way to instantiate the PyAudio stream so that it inputs from the computer's output and i don't have to connect the headphone jack to the microphone? is there a better way to go about this? i'd prefer to stick w/ a python app and avoid cocoa.

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  • Why can't I access elements inside an XML file with XPath in XML::LibXML?

    - by John
    I have an XML file, part of which looks like this: <wave waveID="1"> <well wellID="1" wellName="A1"> <oneDataSet> <rawData>0.1123975676</rawData> </oneDataSet> </well> ... more wellID's and rawData continues here... I am trying to parse the file with Perl's libXML and output the wellName and the rawData using the following: use XML::LibXML; my $parser = XML::LibXML->new(); my $doc = $parser->parse_file('/Users/johncumbers/Temp/1_12-18-09-111823.orig.xml'); my $xc = XML::LibXML::XPathContext->new( $doc->documentElement() ); $xc->registerNs('ns', 'http://moleculardevices.com/microplateML'); my @n = $xc->findnodes('//ns:wave[@waveID="1"]'); #xc is xpathContent # should find a tree from the node representing everything beneath the waveID 1 foreach $nod (@n) { my @c = $nod->findnodes('//rawData'); #element inside the tree. print @c; } It is not printing out anything right now and I think I have a problem with my Xpath statements. Please can you help me fix it, or can you show me how to trouble shoot the xpath statements? Thanks.

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  • Linux: how to use Jellyfish from Jack Meterbridge?

    - by klox
    dear all, i have installed Meterbridge. But,i'm just need to use Jellyfish from this package. I changed the Meterbridge properties become: /usr/bin/meterbridge -t jf alsa_pcm:playback_1 alsa_pcm:playback_2 My problem come here, i can open the Jellyfish window but i can't show the wave from input jack. How should i do? have you ever try this? some tell me to set up the Jack Audio Connection Kit, But i don't understand how to do it because i'm new for this

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